Internet Draft
Internet Engineering Task Force SIP WG
Internet Draft Handley/Schulzrinne/Schooler/Rosenberg
draft-ietf-sip-rfc2543bis-00.txt ACIRI/Columbia U./Caltech/dynamicsoft
July 13, 2000
Expires: December 2000
SIP: Session Initiation Protocol
STATUS OF THIS MEMO
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Abstract
The Session Initiation Protocol (SIP) is an application-layer control
(signaling) protocol for creating, modifying and terminating sessions
with one or more participants. These sessions include Internet
multimedia conferences, Internet telephone calls and multimedia
distribution. Members in a session can communicate via multicast or
via a mesh of unicast relations, or a combination of these.
SIP invitations used to create sessions carry session descriptions
which allow participants to agree on a set of compatible media types.
SIP supports user mobility by proxying and redirecting requests to
the user's current location. Users can register their current
location. SIP is not tied to any particular conference control
protocol. SIP is designed to be independent of the lower-layer
transport protocol and can be extended with additional capabilities.
1 Introduction
1.1 Overview of SIP Functionality
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The Session Initiation Protocol (SIP) is an application-layer control
protocol that can establish, modify and terminate multimedia sessions
or calls. These multimedia sessions include multimedia conferences,
distance learning, Internet telephony and similar applications. SIP
can invite both persons and "robots", such as a media storage
service. SIP can invite parties to both unicast and multicast
sessions; the initiator does not necessarily have to be a member of
the session to which it is inviting. Media and participants can be
added to an existing session.
SIP can be used to initiate sessions as well as invite members to
sessions that have been advertised and established by other means.
Sessions can be advertised using multicast protocols such as SAP,
electronic mail, news groups, web pages or directories (LDAP), among
others.
SIP transparently supports name mapping and redirection services,
allowing the implementation of ISDN and Intelligent Network telephony
subscriber services. These facilities also enable personal mobility.
In the parlance of telecommunications intelligent network services,
this is defined as: "Personal mobility is the ability of end users to
originate and receive calls and access subscribed telecommunication
services on any terminal in any location, and the ability of the
network to identify end users as they move. Personal mobility is
based on the use of a unique personal identity (i.e., personal
number)." [1]. Personal mobility complements terminal mobility, i.e.,
the ability to maintain communications when moving a single end
system from one subnet to another.
SIP supports five facets of establishing and terminating multimedia
communications:
User location: determination of the end system to be used for
communication;
User capabilities: determination of the media and media
parameters to be used;
User availability: determination of the willingness of the
called party to engage in communications;
Call setup: "ringing", establishment of call parameters at both
called and calling party;
Call handling: including transfer and termination of calls.
SIP can also initiate multi-party calls using a multipoint control
unit (MCU) or fully-meshed interconnection instead of multicast.
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Internet telephony gateways that connect Public Switched Telephone
Network (PSTN) parties can also use SIP to set up calls between them.
SIP is designed as part of the overall IETF multimedia data and
control architecture currently incorporating protocols such as RSVP
(RFC 2205 [2]) for reserving network resources, the real-time
transport protocol (RTP) (RFC 1889 [3]) for transporting real-time
data and providing QOS feedback, the real-time streaming protocol
(RTSP) (RFC 2326 [4]) for controlling delivery of streaming media,
the session announcement protocol (SAP) [5] for advertising
multimedia sessions via multicast and the session description
protocol (SDP) (RFC 2327 [6]) for describing multimedia sessions.
However, the functionality and operation of SIP does not depend on
any of these protocols.
SIP can also be used in conjunction with other call setup and
signaling protocols. In that mode, an end system uses SIP exchanges
to determine the appropriate end system address and protocol from a
given address that is protocol-independent. For example, SIP could be
used to determine that the party can be reached via H.323 [7], obtain
the H.245 [8] gateway and user address and then use H.225.0 [9] to
establish the call.
In another example, SIP might be used to determine that the callee is
reachable via the PSTN and indicate the phone number to be called,
possibly suggesting an Internet-to-PSTN gateway to be used.
SIP does not offer conference control services such as floor control
or voting and does not prescribe how a conference is to be managed,
but SIP can be used to introduce conference control protocols. SIP
does not allocate multicast addresses.
SIP can invite users to sessions with and without resource
reservation. SIP does not reserve resources, but can convey to the
invited system the information necessary to do this.
1.2 Terminology
In this document, the key words "MUST", "MUST NOT", "REQUIRED",
"SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY",
and "OPTIONAL" are to be interpreted as described in RFC 2119 [10]
and indicate requirement levels for compliant SIP implementations.
1.3 Definitions
This specification uses a number of terms to refer to the roles
played by participants in SIP communications. The definitions of
client, server and proxy are similar to those used by the Hypertext
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Transport Protocol (HTTP) (RFC 2616 [11]). The terms and generic
syntax of URI and URL are defined in RFC 2396 [12]. The following
terms have special significance for SIP.
Call: A call consists of all participants in a conference
invited by a common source. A SIP call is identified by a
globally unique call-id (Section 6.13). Thus, if a user is,
for example, invited to the same multicast session by
several people, each of these invitations will be a unique
call. A point-to-point Internet telephony conversation maps
into a single SIP call. In a multiparty conference unit
(MCU) based call-in conference, each participant uses a
separate call to invite himself to the MCU.
Call leg: A call leg is identified by the combination of the
Call-ID header field and the addr-spec and tag of the To
and From header fields. Only the user and hostport parts of
the addr-spec are significant. Within the same Call-ID,
requests with From A and To value B belong to the same call
leg as the requests in the opposite direction, i.e., From B
and To A.
Client: An application program that sends SIP requests. Clients
may or may not interact directly with a human user. User
agents and proxies contain clients (and servers).
Conference: A multimedia session (see below), identified by a
common session description. A conference can have zero or
more members and includes the cases of a multicast
conference, a full-mesh conference and a two-party
"telephone call", as well as combinations of these. Any
number of calls can be used to create a conference.
Downstream: Requests sent in the direction from the caller to
the callee (i.e., user agent client to user agent server).
Final response: A response that terminates a SIP transaction, as
opposed to a provisional response that does not. All 2xx,
3xx, 4xx, 5xx and 6xx responses are final.
Initiator, calling party, caller: The party initiating a
conference invitation. Note that the calling party does not
have to be the same as the one creating the conference.
Invitation: A request sent to a user (or service) requesting
participation in a session. A successful SIP invitation
consists of two transactions: an INVITE request followed by
an ACK request.
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Invitee, invited user, called party, callee: The person or
service that the calling party is trying to invite to a
conference.
Isomorphic request or response: Two requests or responses are
defined to be isomorphic for the purposes of this document
if they have the same values for the Call-ID, To, From and
CSeq header fields. In addition, isomorphic requests have
to have the same Request-URI.
Location server: See location service.
Location service: A location service is used by a SIP redirect
or proxy server to obtain information about a callee's
possible location(s). Examples of sources of location
information include SIP registrars, databases or mobility
registration protocols. Location services are offered by
location servers. Location servers MAY be part of a SIP
server, but the manner in which a SIP server requests
location services is beyond the scope of this document.
Outbound proxy: A proxy that is located near the originator of
requests. It receives all outgoing requests from a
particular UAC, including those requests whose Request-URLs
identify a host other than the outbound proxy. The outbound
proxy sends these requests, after any local processing, to
the address indicated in the request-URI. (All other proxy
servers are simply referred as proxies, not inbound
proxies.)
Parallel search: In a parallel search, a proxy issues several
requests to possible user locations upon receiving an
incoming request. Rather than issuing one request and then
waiting for the final response before issuing the next
request as in a sequential search , a parallel search
issues requests without waiting for the result of previous
requests.
Provisional response: A response used by the server to indicate
progress, but that does not terminate a SIP transaction.
1xx responses are provisional, other responses are
considered final.
Proxy, proxy server: An intermediary program that acts as both a
server and a client for the purpose of making requests on
behalf of other clients. Requests are serviced internally
or by passing them on, possibly after translation, to other
servers. A proxy interprets, and, if necessary, rewrites a
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request message before forwarding it.
Redirect server: A redirect server is a server that accepts a
SIP request, maps the address into zero or more new
addresses and returns these addresses to the client. Unlike
a proxy server , it does not initiate its own SIP request.
Unlike a user agent server , it does not accept calls.
Registrar: A registrar is a server that accepts REGISTER
requests. A registrar is typically co-located with a proxy
or redirect server and MAY make its information available
through the location server.
Ringback: Ringback is the signaling tone produced by the calling
client's application indicating that a called party is
being alerted (ringing).
Server: A server is an application program that accepts requests
in order to service requests and sends back responses to
those requests. Servers are either proxy, redirect or user
agent servers or registrars.
Session: From the SDP specification: "A multimedia session is a
set of multimedia senders and receivers and the data
streams flowing from senders to receivers. A multimedia
conference is an example of a multimedia session." (RFC
2327 [6]) (A session as defined for SDP can comprise one or
more RTP sessions.) As defined, a callee can be invited
several times, by different calls, to the same session. If
SDP is used, a session is defined by the concatenation of
the user name , session id , network type , address type
and address elements in the origin field.
(SIP) transaction: A SIP transaction occurs between a client and
a server and comprises all messages from the first request
sent from the client to the server up to a final (non-1xx)
response sent from the server to the client. A transaction
is identified by the CSeq sequence number (Section 6.21)
within a single call leg. The ACK request has the same
CSeq number as the corresponding INVITE request, but
comprises a transaction of its own.
Upstream: Responses sent in the direction from the user agent
server to the user agent client.
URL-encoded: A character string encoded according to RFC 1738,
Section 2.2 [13].
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User agent client (UAC): A user agent client is a client
application that initiates a SIP request.
User agent server (UAS): A user agent server is a server
application that contacts the user when a SIP request is
received and that returns a response on behalf of the user.
The response accepts, rejects or redirects the request.
User agent (UA): An application which can act both as a user
agent client and user agent server.
An application program MAY be capable of acting both as a client and
a server. For example, a typical multimedia conference control
application would act as a user agent client to initiate calls or to
invite others to conferences and as a user agent server to accept
invitations. The role of UAC and UAS as well as proxy and redirect
servers are defined on a request-by-request basis. For example, the
user agent initiating a call acts as a UAC when sending the initial
INVITE request and as a UAS when receiving a BYE request from the
callee. Similarly, the same software can act as a proxy server for
one request and as a redirect server for the next request.
Proxy, redirect, location and registrar servers defined above are
logical entities; implementations MAY combine them into a single
application program. The properties of the different SIP server
types are summarized in Table 1.
property redirect proxy user agent registrar
server server server
__________________________________________________________________
also acts as a SIP client no yes no no
returns 1xx status yes yes yes yes
returns 2xx status no yes yes yes
returns 3xx status yes yes yes yes
returns 4xx status yes yes yes yes
returns 5xx status yes yes yes yes
returns 6xx status no yes yes yes
inserts Via header no yes no no
accepts ACK yes yes yes no
Table 1: Properties of the different SIP server types
1.4 Overview of SIP Operation
This section explains the basic protocol functionality and operation.
Callers and callees are identified by SIP addresses, described in
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Section 1.4.1. When making a SIP call, a caller first locates the
appropriate server (Section 1.4.2) and then sends a SIP request
(Section 1.4.3). The most common SIP operation is the invitation
(Section 1.4.4). Instead of directly reaching the intended callee, a
SIP request may be redirected or may trigger a chain of new SIP
requests by proxies (Section 1.4.5). Users can register their
location(s) with SIP servers (Section 4.2.6).
1.4.1 SIP Addressing
The "objects" addressed by SIP are users at hosts, identified by a
SIP URL. The SIP URL takes a form similar to a mailto or telnet URL,
i.e., user@host. The user part is a user name or a telephone number.
The host part is either a domain name or a numeric network address.
See section 2 for a detailed discussion of SIP URL's.
A user's SIP address can be obtained out-of-band, can be learned via
existing media agents, can be included in some mailers' message
headers, or can be recorded during previous invitation interactions.
In many cases, a user's SIP URL can be guessed from their email
address.
A SIP URL address can designate an individual (possibly located at
one of several end systems), the first available person from a group
of individuals or a whole group. The form of the address, for
example, sip:sales@example.com , is not sufficient, in general, to
determine the intent of the caller.
If a user or service chooses to be reachable at an address that is
guessable from the person's name and organizational affiliation, the
traditional method of ensuring privacy by having an unlisted "phone"
number is compromised. However, unlike traditional telephony, SIP
offers authentication and access control mechanisms and can avail
itself of lower-layer security mechanisms, so that client software
can reject unauthorized or undesired call attempts.
1.4.2 Locating a SIP Server
When a client wishes to send a request, the client either sends it to
a locally configured SIP proxy server (as in HTTP), independent of
the Request-URI, or sends it to the IP address and port corresponding
to the Request-URI.
For the latter case, the client must determine the protocol, port and
IP address of a server to which to send the request. A client SHOULD
follow the steps below to obtain this information,
At each step, unless stated otherwise, the client SHOULD try to
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contact a server at the port number listed in the Request-URI. If no
port number is present in the Request-URI, the client uses port 5060.
If the Request-URI specifies a protocol (TCP or UDP), the client
contacts the server using that protocol. If no protocol is specified,
the client tries UDP (if UDP is supported). If the attempt fails, or
if the client doesn't support UDP but supports TCP, it then tries
TCP.
A client SHOULD be able to interpret explicit network notifications
(such as ICMP messages) which indicate that a server is not
reachable, rather than relying solely on timeouts. (For socket-based
programs: For TCP, connect() returns ECONNREFUSED if the client
could not connect to a server at that address. For UDP, the socket
needs to be bound to the destination address using connect() rather
than sendto() or similar so that a second write() or send() fails
with ECONNREFUSED if there is no server listening) If the client
finds the server is not reachable at a particular address, it SHOULD
behave as if it had received a 400-class error response to that
request.
The client tries to find one or more addresses for the SIP server by
querying DNS. If a step elicits no addresses, the client continues to
the next step. However if a step elicits one or more addresses, but
no SIP server at any of those addresses responds, then the client
concludes the server is down and doesn't continue on to the next
step.
The service identifier for DNS SRV records [14] is "_sip". SRV
records contain port numbers for servers, in addition to IP
addresses; the client always uses this port number when contacting
the SIP server. Otherwise, the port number in the SIP URI is used, if
present. If there is no port number in the URI, the default port,
5060, is used.
The procedure is described below.
1. If the maddr parameter of the Request-URI is an IP address,
the client contacts the server at the given address and
ignores the remaining steps.
2. If the maddr parameter of the Request-URI is a host name,
the client proceeds to the last step.
3. If the host portion of the Request-URI is an IP address,
the client contacts the server at the given address.
Otherwise, the client proceeds to the next step.
4. The Request-URI is examined. If it contains an explicit
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port number other than 5060, the next two steps are
skipped.
5. The Request-URI is examined. If it does not specify a
protocol (TCP or UDP), the client queries the name server
for SRV records for both UDP (if supported by the client)
and TCP (if supported by the client) SIP servers. The
format of these queries is defined in RFC 2782 [14]. The
results of the query or queries are merged together and
ordered based on priority. Then, the searching technique
outlined in RFC 2782 [14] is used to select servers in
order. If DNS does not return any records, the user goes
to the last step. Otherwise, the user attempts to contact
each server in the order listed. If none of the server can
be contacted, the user gives up.
6. If the Request-URI specifies a protocol (TCP or UDP) that
is supported by the client, the client queries the name
server for SRV records for SIP servers of that protocol
type only. If the client does not support the protocol
specified in the Request-URI, it gives up. The searching
technique outlined in RFC 2782 [14] is used to select
servers from the DNS response in order. If DNS does not
return any records, the user goes to the last step.
Otherwise, the user attempts to contact each server in the
order listed. If no server is contacted, the user gives up.
7. The client queries the DNS server for address records for
the host portion of the maddr parameter, if preent, or the
Request-URI. If the DNS server returns no address records,
the client stops, as it has been unable to locate a server.
Address records include A RR's, AAAA RR's, or other similar
records, chosen according to the client's network protocol
capabilities.
There are no mandatory rules on how to select a host name
for a SIP server. Users are encouraged to name their SIP
servers using the sip.domainname (e.g., sip.example.com )
convention, as specified in RFC 2219 [15]. Users may only
know an email address instead of a full SIP URL for a
callee, however. In that case, implementations may be able
to increase the likelihood of reaching a SIP server for
that domain by constructing a SIP URL from that email
address by prefixing the host name with "sip.". In the
future, this mechanism is likely to become unnecessary as
SRV records, described above, become widely available.
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A client MAY cache the list of DNS query results if one of the
addresses was contacted successfully. Request for the same
transaction SHOULD be sent to the same network address. Other
requests from the same client select a server from the list of
addresses cached, using the SRV load-balancing mechanism if
applicable. The client must invalidate this list and retry the DNS
query according to the rules in RFC1035 [16].
A client MAY omit attempting to reach a server which it had failed to
reach for a previous request.
If the DNS time-to-live value exceeds a few minutes,
servers generating a large number of requests are probably
well advised to retry failed servers every few minutes.
1.4.3 SIP Transaction
Once the host part has been resolved to a SIP server, the client
sends one or more SIP requests to that server and receives one or
more responses from the server. A request (and its retransmissions)
together with the responses triggered by that request make up a SIP
transaction. All responses to a request contain the same values in
the Call-ID, CSeq, To, and From fields (with the possible addition of
a tag in the To field (section 6.44)). This allows responses to be
matched with requests. The ACK request following an INVITE is not
part of the transaction since it may traverse a different set of
hosts.
If TCP is used, request and responses within a single SIP transaction
are carried over the same TCP connection (see Section 10). Several
SIP requests from the same client to the same server MAY use the same
TCP connection or MAY use a new connection for each request.
If the client sent the request via unicast UDP, the response is sent
to the address contained in the next Via header field (Section 6.47)
of the response. If the request is sent via multicast UDP, the
response is directed to the same multicast address and destination
port. For UDP, reliability is achieved using retransmission (Section
10).
The SIP message format and operation is independent of the transport
protocol.
1.4.4 SIP Invitation
A successful SIP invitation consists of two requests, INVITE followed
by ACK. The INVITE (Section 4.2.1) request asks the callee to join a
particular conference or establish a two-party conversation. After
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the callee has agreed to participate in the call, the caller confirms
that it has received that response by sending an ACK (Section 4.2.2)
request. If the caller no longer wants to participate in the call, it
sends a BYE request instead of an ACK.
The INVITE request typically contains a session description, for
example written in SDP (RFC 2327 [6]) format, that provides the
called party with enough information to join the session. For
multicast sessions, the session description enumerates the media
types and formats that are allowed to be distributed to that session.
For a unicast session, the session description enumerates the media
types and formats that the caller is willing to use and where it
wishes the media data to be sent. In either case, if the callee
wishes to accept the call, it responds to the invitation by returning
a similar description listing the media it wishes to use. For a
multicast session, the callee SHOULD only return a session
description if it is unable to receive the media indicated in the
caller's description or wants to receive data via unicast.
The protocol exchanges for the INVITE method are shown in Fig. 1 for
a proxy server and in Fig. 2 for a redirect server. (Note that the
messages shown in the figures have been abbreviated slightly.) In
Fig. 1, the proxy server accepts the INVITE request (step 1),
contacts the location service with all or parts of the address (step
2) and obtains a more precise location (step 3). The proxy server
then issues a SIP INVITE request to the address(es) returned by the
location service (step 4). The user agent server alerts the user
(step 5) and returns a success indication to the proxy server (step
6). The proxy server then returns the success result to the original
caller (step 7). The receipt of this message is confirmed by the
caller using an ACK request, which is forwarded to the callee (steps
8 and 9). Note that an ACK can also be sent directly to the callee,
bypassing the proxy. All requests and responses have the same Call-
ID.
The redirect server shown in Fig. 2 accepts the INVITE request (step
1), contacts the location service as before (steps 2 and 3) and,
instead of contacting the newly found address itself, returns the
address to the caller (step 4), which is then acknowledged via an ACK
request (step 5). The caller issues a new request, with the same
call-ID but a higher CSeq, to the address returned by the first
server (step 6). In the example, the call succeeds (step 7). The
caller and callee complete the handshake with an ACK (step 8).
The next section discusses what happens if the location service
returns more than one possible alternative.
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+....... cs.columbia.edu .......+
: :
: (~~~~~~~~~~) :
: ( location ) :
: ( service ) :
: (~~~~~~~~~~) :
: ^ | :
: | hgs@lab :
: 2| 3| :
: | | :
: henning | :
+.. cs.tu-berlin.de ..+ 1: INVITE : | | :
: : henning@cs.col: | \/ 4: INVITE 5: ring :
: cz@cs.tu-berlin.de ========================>(~~~~~~)=========>(~~~~~~) :
: <........................( )<.........( ) :
: : 7: 200 OK : ( )6: 200 OK ( ) :
: : : ( work ) ( lab ) :
: : 8: ACK : ( )9: ACK ( ) :
: ========================>(~~~~~~)=========>(~~~~~~) :
+.....................+ +...............................+
====> SIP request
....> SIP response
^
| non-SIP protocols
|
Figure 1: Example of SIP proxy server
1.4.5 Locating a User
A callee may move between a number of different end systems over
time. These locations can be dynamically registered with the SIP
server (Sections 1.4.7, 4.2.6). A location server MAY also use one or
more other protocols, such as finger (RFC 1288 [17]), rwhois (RFC
2167 [18]), LDAP (RFC 1777 [19]), multicast-based protocols [20] or
operating-system dependent mechanisms to actively determine the end
system where a user might be reachable. A location server MAY return
several locations because the user is logged in at several hosts
simultaneously or because the location server has (temporarily)
inaccurate information. The SIP server combines the results to yield
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+....... cs.columbia.edu .......+
: :
: (~~~~~~~~~~) :
: ( location ) :
: ( service ) :
: (~~~~~~~~~~) :
: ^ | :
: | hgs@lab :
: 2| 3| :
: | | :
: henning| :
+.. cs.tu-berlin.de ..+ 1: INVITE : | | :
: : henning@cs.col: | \/ :
: cz@cs.tu-berlin.de =======================>(~~~~~~) :
: | ^ | <.......................( ) :
: | . | : 4: 302 Moved : ( ) :
: | . | : hgs@lab : ( work ) :
: | . | : : ( ) :
: | . | : 5: ACK : ( ) :
: | . | =======================>(~~~~~~) :
: | . | : : :
+.......|...|.........+ : :
| . | : :
| . | : :
| . | : :
| . | : :
| . | 6: INVITE hgs@lab.cs.columbia.edu (~~~~~~) :
| . ==================================================> ( ) :
| ..................................................... ( ) :
| 7: 200 OK : ( lab ) :
| : ( ) :
| 8: ACK : ( ) :
======================================================> (~~~~~~) :
+...............................+
====> SIP request
....> SIP response
^
| non-SIP protocols
|
Figure 2: Example of SIP redirect server
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a list of a zero or more locations.
The action taken on receiving a list of locations varies with the
type of SIP server. A SIP redirect server returns the list to the
client as Contact headers (Section 6.15). A SIP proxy server can
sequentially or in parallel try the addresses until the call is
successful (2xx response) or the callee has declined the call (6xx
response). With sequential attempts, a proxy server can implement an
"anycast" service.
If a proxy server forwards a SIP request, it MUST add itself to the
beginning of the list of forwarders noted in the Via (Section 6.47)
headers. The Via trace ensures that replies can take the same path
back, ensuring correct operation through compliant firewalls and
avoiding request loops. On the response path, each host MUST remove
its Via, so that routing internal information is hidden from the
callee and outside networks. A proxy server MUST check that it does
not generate a request to a host listed in the Via sent-by, via-
received or via-maddr parameters (Section 6.47). (Note: If a host has
several names or network addresses, this does not always work. Thus,
each host also checks if it is part of the Via list.)
A SIP invitation may traverse more than one SIP proxy server. If one
of these "forks" the request, i.e., issues more than one request in
response to receiving the invitation request, it is possible that a
client is reached, independently, by more than one copy of the
invitation request. Each of these copies bears the same Call-ID. The
user agent MUST return the same status response returned in the first
response. Duplicate requests are not an error.
1.4.6 Changing an Existing Session
In some circumstances, it is desirable to change the parameters of an
existing session. This is done by re-issuing the INVITE, using the
same Call-ID, but a new or different body or header fields to convey
the new information. This re INVITE MUST have a higher CSeq than any
previous request from the client to the server.
For example, two parties may have been conversing and then want to
add a third party, switching to multicast for efficiency. One of the
participants invites the third party with the new multicast address
and simultaneously sends an INVITE to the second party, with the new
multicast session description, but with the old call identifier.
1.4.7 Registration Services
The REGISTER request allows a client to let a proxy or redirect
server know at which address(es) it can be reached. A client MAY also
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use it to install call handling features at the server.
1.5 Protocol Properties
1.5.1 Minimal State
A single conference session or call involves one or more SIP
request-response transactions. Proxy servers do not have to keep
state for a particular call, however, they MAY maintain state for a
single SIP transaction, as discussed in Section 12. For efficiency, a
server MAY cache the results of location service requests.
1.5.2 Lower-Layer-Protocol Neutral
SIP makes minimal assumptions about the underlying transport and
network-layer protocols. The lower-layer can provide either a packet
or a byte stream service, with reliable or unreliable service.
In an Internet context, SIP is able to utilize both UDP and TCP as
transport protocols, among others. UDP allows the application to more
carefully control the timing of messages and their retransmission, to
perform parallel searches without requiring TCP connection state for
each outstanding request, and to use multicast. Routers can more
readily snoop SIP UDP packets. TCP allows easier passage through
existing firewalls.
When TCP is used, SIP can use one or more connections to attempt to
contact a user or to modify parameters of an existing conference.
Different SIP requests for the same SIP call MAY use different TCP
connections or a single persistent connection, as appropriate.
For concreteness, this document will only refer to Internet
protocols. However, SIP MAY also be used directly with protocols
such as ATM AAL5, IPX, frame relay or X.25. The necessary naming
conventions are beyond the scope of this document. User agents SHOULD
implement both UDP and TCP transport. Proxy, registrar, and redirect
servers MUST implement both UDP and TCP transport.
1.5.3 Text-Based
SIP is text-based, using ISO 10646 in UTF-8 encoding throughout. This
allows easy implementation in languages such as Java, Tcl and Perl,
allows easy debugging, and most importantly, makes SIP flexible and
extensible. As SIP is used for initiating multimedia conferences
rather than delivering media data, it is believed that the additional
overhead of using a text-based protocol is not significant.
2 SIP Uniform Resource Locators
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SIP URLs are used within SIP messages to indicate the originator
(From), current destination (Request-URI) and final recipient (To) of
a SIP request, and to specify redirection addresses (Contact). A SIP
URL can also be embedded in web pages or other hyperlinks to indicate
that a particular user or service can be called via SIP. When used as
a hyperlink, the SIP URL indicates the use of the INVITE method.
The SIP URL scheme is defined to allow setting SIP request-header
fields and the SIP message-body.
This corresponds to the use of mailto: URLs. It makes it
possible, for example, to specify the subject, urgency or
media types of calls initiated through a web page or as
part of an email message.
A SIP URL follows the guidelines of RFC 2396 [12] and has the syntax
shown in Fig. 3. The syntax is described using Augmented Backus-Naur
Form (see Section C). Note that reserved characters have to be
escaped and that the "set of characters reserved within any given URI
component is defined by that component. In general, a character is
reserved if the semantics of the URI changes if the character is
replaced with its escaped US-ASCII encoding" [12]. Excluded US-ASCII
characters [12], such as space and control characters and characters
used as URL delimiters, also MUST be escaped.
The URI character classes referenced above are described in Appendix
C.
The components of the SIP URI have the following meanings.
user: The name of the user addressed. Note that this field MAY
be empty where the destination host does not have a notion
of users, e.g., for embedded devices.
telephone-subscriber: If the host is an Internet telephony
gateway, a telephone-subscriber field MAY be used instead
of a user field. The telephone-subscriber field uses the
notation of RFC 2806 [21]. Any characters of the un-escaped
"telephone-subscriber" that are not either in the set
"unreserved" or "user-unreserved" MUST be escaped. The set
of characters not reserved in the RFC 2806 description of
telephone-subscriber contains a number of characters in
various syntax elements that need to be escaped when used
in SIP URLs, for example quotation marks (%x22), hash
(%x23), colon (%x3a), at-sign (%x40) and the "unwise"
characters, i.e., punctuation of %x5b and above.
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The telephone number is a special case of a user name and
cannot be distinguished by a BNF. Thus, a URL parameter,
user, is added to distinguish telephone numbers from user
names.
The user parameter value "phone" indicates that the user
part contains a telephone number. Even without this
parameter, recipients of SIP URLs MAY interpret the pre-@
part as a telephone number if local restrictions on the
name space for user name allow it.
The user parameter value "np-queried" indicates that the
user part contains a telephone number and that the number
reflects the result of a query to the local number
portability database.
password: The SIP scheme MAY use the format "user:password" in
the userinfo field. The use of passwords in the userinfo is
NOT RECOMMENDED, because the passing of authentication
information in clear text (such as URIs) has proven to be a
security risk in almost every case where it has been used.
host: The host part SHOULD be a fully-qualified domain name or
numeric IP address.
The mailto: URL and RFC 822 email addresses require that
numeric host addresses ("host numbers") are enclosed in
square brackets (presumably, since host names might be
numeric), while host numbers without brackets are used for
all other URLs. The SIP URL requires the latter form,
without brackets.
port: The port number to send a request to. If not present, the
procedures outlined in Section 1.4.2 are used to determine
the port number to send a request to.
URL parameters: SIP URLs can define specific parameters of the
request. URL parameters are added after the host component
and are separated by semi-colons. The transport parameter
determines the the transport mechanism to be used for
sending SIP requests and responses. SIP can use any network
transport protocol; parameter names are defined for UDP
[22], TCP [23], TLS [24], and SCTP. UDP is to be assumed
when no explicit transport parameter is included. The
maddr parameter provides the server address to be contacted
for this user, overriding the address supplied in the host
field. This address is typically a multicast address, but
could also be the address of a backup server. The ttl
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SIP-URL = "sip:" [ userinfo "@" ] hostport
url-parameters [ headers ]
userinfo = [ user | telephone-subscriber ] [ ":" password ]
user = *( unreserved | escaped
| "&" | "=" | "+" | "$" | "," | ";" | "?" | "/" )
password = *( unreserved | escaped
| "&" | "=" | "+" | "$" | "," )
hostport = host [ ":" port ]
host = hostname | IPv4address | IPv6reference
hostname = *( domainlabel "." ) toplabel [ "." ]
domainlabel = alphanum | alphanum *( alphanum | "-" ) alphanum
toplabel = alpha | alpha *( alphanum | "-" ) alphanum
IPv4address = 1*3DIGIT "." 1*3DIGIT "." 1*3DIGIT "." 1*3DIGIT
IPv6reference = "[" IPv6address "]"
IPv6address = hexpart [ ":" IPv4address ]
hexpart = hexseq | hexseq "::" [ hexseq ] | "::" [ hexseq ]
hexseq = hex4 *( ":" hex4)
hex4 = 1*4HEX
port = 1*DIGIT
url-parameters = *( ";" url-parameter )
url-parameter = transport-param | user-param | method-param
| ttl-param | maddr-param | other-param
transport-param = "transport="
( "udp" | "tcp" | "sctp" | "tls" | other-transport )
other-transport = token
ttl-param = "ttl=" ttl
ttl = 1*3DIGIT ; 0 to 255
maddr-param = "maddr=" host
user-param = "user=" ( "phone" | "ip" | other-user )
other-user = token
method-param = "method=" Method
tag-param = "tag=" UUID
UUID = 1*( HEX | "-" )
other-param = pname [ "=" pvalue ]
pname = 1*paramchar
pvalue = 1*paramchar
paramchar = param-reserved | unreserved | escaped
param-reserved = "[" | "]" | "/" | "?" | ":" | "@" | "&" | "+" | "$"
headers = "?" header *( "&" header )
header = hname "=" hvalue
hname = 1*( hnv-unreserved | unreserved | escaped )
hvalue = *( hnv-unreserved | unreserved | escaped )
hnv-unreserved = "[" | "]" | "/" | "?" | ":" | "@" | "+" | "$"
Figure 3: SIP URL syntax
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multicast address and the transport protocol is UDP. The
user parameter was described above. For example, to specify
to call j.doe@big.com using multicast to 239.255.255.1 with
a ttl of 15, the following URL would be used:
sip:j.doe@big.com;maddr=239.255.255.1;ttl=15
The transport, maddr, and ttl parameters MUST NOT be used
in the From and To header fields and the Request-URI; they
are ignored if present.
Headers: Headers of the SIP request can be defined with the "?"
mechanism within a SIP URL. The special hname "body"
indicates that the associated hvalue is the message-body of
the SIP INVITE request. Headers MUST NOT be used in the
From and To header fields and the Request-URI; they are
ignored if present. hname and hvalue are encodings of a
SIP header name and value, respectively. All URL reserved
characters in the header names and values MUST be escaped.
Method: The method of the SIP request can be specified with the
method parameter. This parameter MUST NOT be used in the
From and To header fields and the Request-URI; they are
ignored if present.
Table 2 summarizes where the components of the SIP URL can be used.
Entries marked "m" are mandatory, those marked "o" are optional, and
those marked "-" are not allowed. For optional elements, the second
column indicates the default value if the element is not present.
Examples of SIP URLs are:
sip:j.doe@big.com
sip:j.doe:secret@big.com;transport=tcp
sip:j.doe@big.com?subject=project
sip:+1-212-555-1212:1234@gateway.com;user=phone
sip:1212@gateway.com
sip:alice@10.1.2.3
sip:alice@example.com
sip:alice
sip:alice@registrar.com;method=REGISTER
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default Req.-URI To From Contact Rec.-Route external
user -- o o o o o o
password -- o o - o o o
host mandatory m m m m m m
port 5060 o o o o o o
user-param ip o o o o o o
method INVITE - - - o - o
maddr-param -- o - - o m o
ttl-param 1 o - - o - o
transp.-param udp o - - o - o
other-param o o o o o o o
headers -- - - - o - o
Table 2: Use and default values of URL components for SIP headers,
Request-URI and references
Within a SIP message, URLs are used to indicate the source and
intended destination of a request, redirection addresses and the
current destination of a request. Normally all these fields will
contain SIP URLs.
2.1 SIP URL Comparison
SIP URLs are compared for equality according to the following rules:
o Comparisons of scheme name ("sip"), domain names, parameter
names and header names are case-insensitive, all other
comparisons are case-sensitive.
o The ordering of parameters and headers is not significant in
comparing SIP URLs.
o user, password, host, port and any url-parameter parameters of
the URI must match. If a component is omitted, it matches
based on its default value. (For example, otherwise equivalent
URLs without a port specification and with port 5060 match.)
Note that the two URLs example.com and example.com:5060 ,
while considered equal, may not lead to the same server, as
the former causes a DNS SRV lookup, while the latter only uses
the A record.
o Characters other than those in the "reserved" and "unsafe"
sets (see RFC 2396 [12]) are equivalent to their ""%" HEX HEX"
encoding.
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o An IP address that is the result of a DNS lookup of a host
name does not match that host name.
Thus, the following URLs are equivalent:
sip:juser@
sip:juser@ExAmPlE.CoM;Transport=udp
while
SIP:JUSER@ExAmPlE.CoM;Transport=udp
sip:juser@ExAmPlE.CoM;Transport=UDP
are not.
2.2 Non-SIP URLs
SIP header fields and the Request-URI MAY contain non-SIP URLs, with
the exceptions noted below. As an example, if a call from a telephone
is relayed to the Internet via SIP, the SIP From header field might
contain a tel: URL [21].
In the following locations, only SIP URLs are allowed:
o Request-URI in a REGISTER request;
o Contact header field in INVITE, OPTIONS and BYE and 2xx
responses.
Implementations MAY compare non-SIP URLs by treating them as generic
URIs [12] or, alternatively, compare them byte-by-byte.
3 SIP Message Overview
SIP is a text-based protocol and uses the ISO 10646 character set in
UTF-8 encoding (RFC 2279 [25]). Senders MUST terminate lines with a
CRLF, but receivers MUST also interpret CR and LF by themselves as
line terminators. Only the combinations CR CR, LF LF and CRLF CRLF
terminate the message header. Implementations MUST only send CRLF
CRLF.
CR and LF instead of CRLF is for backwards-compatibility;
their use is deprecated.
Except for the above difference in character sets and line
termination, much of the message syntax is and header fields are
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identical to HTTP/1.1; rather than repeating the syntax and semantics
here we use [HX.Y] to refer to Section X.Y of the current HTTP/1.1
specification (RFC 2616 [11]). In addition, we describe SIP in both
prose and an augmented Backus-Naur form (ABNF). See section C for an
overview of ABNF.
Note, however, that SIP is not an extension of HTTP.
Unlike HTTP, SIP MAY use UDP. When sent over TCP or UDP, multiple SIP
transactions can be carried in a single TCP connection or UDP
datagram. UDP datagrams, including all headers, SHOULD NOT be larger
than the path maximum transmission unit (MTU) if the MTU is known, or
1500 bytes if the MTU is unknown.
The MTU of 1500 bytes accommodates encapsulation within the
"typical" ethernet MTU without IP fragmentation. Recent
studies [26] indicate that an MTU of 1500 bytes is a
reasonable assumption. The next lower common MTU values are
1006 bytes for SLIP and 296 for low-delay PPP (RFC 1191
[27]). Thus, another reasonable value would be a message
size of 950 bytes, to accommodate packet headers within the
SLIP MTU without fragmentation.
A SIP message is either a request from a client to a server, or a
response from a server to a client.
SIP-message = Request | Response
Both Request (section 4) and Response (section 5) messages use the
generic-message format of RFC 822 [28] for transferring entities (the
body of the message). Both types of messages consist of a start-line,
one or more header fields (also known as "headers"), an empty line
(i.e., a line with nothing preceding the carriage-return line-feed
(CRLF)) indicating the end of the header fields, and an optional
message-body. To avoid confusion with similar-named headers in HTTP,
we refer to the headers describing the message body as entity
headers. These components are described in detail in the upcoming
sections.
generic-message = start-line
*message-header
CRLF
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[ message-body ]
start-line = Request-Line | ;Section 4.1
Status-Line ;Section 5.1
message-header = ( general-header
| request-header
| response-header
| entity-header )
In the interest of robustness, any leading empty line(s) MUST be
ignored. In other words, if the Request or Response message begins
with one or more CRLF, CR, or LFs, these characters MUST be ignored.
4 Request
The Request message format is shown below:
Request = Request-Line ; Section 4.1
*( general-header
| request-header
| entity-header )
CRLF
[ message-body ] ; Section 8
4.1 Request-Line
The Request-Line begins with a method token, followed by the
Request-URI and the protocol version, and ending with CRLF. The
elements are separated by SP characters. No CR or LF are allowed
except in the final CRLF sequence.
Request-Line = Method SP Request-URI SP SIP-Version CRLF
Request-URI = SIP-URL | absoluteURI
4.2 Methods
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general-header = Accept ; Section 6.7
| Accept-Encoding ; Section 6.8
| Accept-Language ; Section 6.9
| Call-ID ; Section 6.13
| Contact ; Section 6.15
| CSeq ; Section 6.21
| Date ; Section 6.22
| Encryption ; Section 6.23
| From ; Section 6.25
| Organization ; Section 6.30
| Record-Route ; Section 6.35
| Require ; Section 6.36
| Supported ; Section 6.42
| Timestamp ; Section 6.43
| To ; Section 6.44
| User-Agent ; Section 6.46
| Via ; Section 6.47
entity-header = Allow ; Section 6.11
| Content-Disposition ; Section 6.16
| Content-Encoding ; Section 6.17
| Content-Language ; Section 6.18
| Content-Length ; Section 6.19
| Content-Type ; Section 6.20
| Expires ; Section 6.24
request-header = Authorization ; Section 6.12
| Hide ; Section 6.26
| In-Reply-To ; Section 6.27
| Max-Forwards ; Section 6.28
| Priority ; Section 6.31
| Proxy-Authorization ; Section 6.33
| Proxy-Require ; Section 6.34
| Route ; Section 6.39
| Response-Key ; Section 6.37
| Subject ; Section 6.41
response-header = Proxy-Authenticate ; Section 6.32
| Retry-After ; Section 6.38
| Server ; Section 6.40
| Unsupported ; Section 6.45
| Warning ; Section 6.48
| WWW-Authenticate ; Section 6.49
Table 3: SIP headers
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The methods are defined below. Methods that are not supported by a
proxy or redirect server are treated by that server as if they were
an OPTIONS method and forwarded accordingly. Methods that are not
supported by a user agent server or registrar cause a 501 (Not
Implemented) response to be returned (Section 7). As in HTTP, the
Method token is case-sensitive.
Method = "INVITE" | "ACK" | "OPTIONS" | "BYE"
| "CANCEL" | "REGISTER" | extension-method
extension-method = token
4.2.1 INVITE
The INVITE method indicates that the user or service is being invited
to participate in a session. The message body MAY contain a
description of the session to which the callee is being invited. For
two-party calls, the caller indicates the type of media it is able to
receive and possibly the media it is willing to send as well as their
parameters such as network destination. A success response MUST
indicate in its message body which media the callee wishes to receive
and MAY indicate the media the callee is going to send.
Not all session description formats have the ability to
indicate sending media.
The caller MAY choose to omit the request body (i.e., not send a
session description) or send a session description that does not list
any media types. This indicates that the caller does not know its
desired media characteristics until the call has been accepted. In
this case, the UAS SHOULD still return a session description in its
informational (1xx) or success (2xx) response, containing those media
streams and codecs it supports.
If the INVITE request did not contain a complete session description,
the caller MUST include one in the ACK request. A UAC MUST NOT send
an updated session description in an ACK request if it had already
sent a complete session description in the INVITE request. If the
UAC wishes to modify the session after the call setup has begun, it
MUST use another INVITE request instead.
Delaying the session description until the ACK request is
useful for gateways from H.323v1 to SIP, where the H.323
media characteristics are not known until the call is
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established.
A server MAY automatically respond to an invitation for a conference
the user is already participating in, identified either by the SIP
Call-ID or a globally unique identifier within the session
description, with a 200 (OK) response.
The behavior of UAS depend on whether they are Internet telephony
gateways to the PSTN. A UAS not acting as a gateway which receives an
INVITE with a Request-URI that does not correspond to one of its
configured addresses, MUST respond with 404 (Not Found).
A UAS acting as a gateway translates the INVITE request into a
telephony signaling message. If the INVITE has a Call-ID value that
matches a recent call, the UAS compares the Request-URI with the
Request-URI of the previous INVITE request for the same Call-ID. If
the Request-URI contains additional digits in the "user" part, the
UAS treats the INVITE as adding additional digits to the original
dialed string. This is known as overlap dialing.
If the gateway knows that the telephone number is incomplete, it
returns a 484 (Address Incomplete) status response.
If a user agent receives an INVITE request for an existing call leg
with a higher CSeq sequence number than any previous INVITE for the
same Call-ID, it MUST check any version identifiers in the session
description or, if there are no version identifiers, the content of
the session description to see if it has changed. It MUST also
inspect any other header fields for changes. If there is a change,
the user agent MUST update any internal state or information
generated as a result of that header. If the session description has
changed, the user agent server MUST adjust the session parameters
accordingly, possibly after asking the user for confirmation.
(Versioning of the session description can be used to accommodate the
capabilities of new arrivals to a conference, add or delete media or
change from a unicast to a multicast conference.)
If an INVITE request for an existing session fails, the session
description agreed upon in the last successful INVITE transaction
remains in force.
A UAC MUST NOT issue another INVITE request for the same call leg
before the previous transaction has completed. A UAS that receives an
INVITE before it sent the final response to an INVITE with a lower
CSeq number MUST return a 400 (Bad Request) response and MUST include
a Retry-After header field with a randomly chosen value of between 0
and 10 seconds.
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If a UA A sends an INVITE request to B and receives an INVITE request
from B before it has received the response to its request from B, A
MAY return a 500 (Internal Server Error), which SHOULD include a
Retry-After header field specifying when the request should be
resubmitted.
In most cases, a UA can assume that the order of messages
received corresponds to the order they were sent. In rare
circumstances, the response from B and the request from B
may be reordered on the wire.
In addition, if A or B change multicast addresses, strict transaction
ordering is necessary so that both sides agree on the final result.
A UAC MUST be prepared to receive media data according to the session
description as soon as it sends an INVITE (or re-INVITE) and can
start sending media data when it receives a provisional or final
response containing a session description.
The initial INVITE from the UAC SHOULD contain the Allow and
Supported header fields, and MAY contain the Accept header field. A
200 (OK) response to the initial INVITE for a call SHOULD contain the
Allow and Supported header fields, and MAY contain the Accept header
field.
Including these header fields allows the UAC to determine
the features and extensions supported by the UAS for the
duration of the call, without probing.
This method MUST be supported by SIP proxy, redirect and user agent
servers as well as clients.
4.2.2 ACK
The ACK request confirms that the client has received a final
response to an INVITE request. (ACK is used only with INVITE
requests.) 2xx responses are acknowledged by client user agents, all
other final responses by the first proxy or client user agent to
receive the response. The Via is always initialized to the host that
originates the ACK request, i.e., the client user agent after a 2xx
response or the first proxy to receive a non-2xx final response. The
ACK request is forwarded as the corresponding INVITE request, based
on its Request-URI. See Section 10 for details.
The ACK request MAY contain a message body with the final session
description to be used by the callee. If the ACK message body is
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empty, the callee uses the session description in the INVITE request.
See Section 4.2.1 for further details on the relationship between
session descriptions in INVITE and ACK requests.
A proxy server receiving an ACK request after having sent a 3xx, 4xx,
5xx, or 6xx response must make a determination about whether the ACK
is for it, or for some user agent or proxy server further downstream.
This determination is made by examining the tag in the To field. If
the tag in the ACK To header field matches the tag in the To header
field of the response, and the From, CSeq and Call-ID header fields
in the response match those in the ACK, the ACK is meant for the
proxy server. Otherwise, the ACK SHOULD be proxied downstream as any
other request.
It is possible for a user agent client or proxy server to
receive multiple 3xx, 4xx, 5xx, and 6xx responses to a
request along a single branch. This can happen under
various error conditions, typically when a forking proxy
transitions from stateful to stateless before receiving all
responses. The various responses will all be identical,
except for the tag in the To field, which is different for
each one. It can therefore be used as a means to
disambiguate them.
This method MUST be supported by SIP proxy, redirect and user agent
servers as well as clients.
4.2.3 OPTIONS
The server is being queried as to its capabilities. A server that
believes it can contact the user, such as a user agent where the user
is logged in and has been recently active, MAY respond to this
request with a capability set. A called user agent MAY return a
status reflecting how it would have responded to an invitation, e.g.,
600 (Busy). A server SHOULD return Allow, Accept, Accept-Encoding,
Accept-Language and Supported header fields. The response MAY contain
a message body indicating the capabilities of the end system (rather
than properties of any existing call).
The use of the Call-ID header field is discussed in Section 6.13. An
OPTIONS requests for an existing call-id has no impact on that call.
Proxy and redirect servers simply forward the request without
indicating their capabilities. This method MUST be supported by SIP
proxy, redirect and user agent servers, registrars and clients.
4.2.4 BYE
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The user agent client uses BYE to indicate to the server that it
wishes to release the call. A BYE request is forwarded like an INVITE
request and MAY be issued by either caller or callee. A party to a
call SHOULD issue a BYE request before releasing a call ("hanging
up"). A party receiving a BYE request MUST cease transmitting media
streams specifically directed at the party issuing the BYE request.
A BYE request from either called or calling party terminates any
pending INVITE at a UA, but the INVITE request transaction MUST be
completed with a final response and ACK.
If the INVITE request contained a Contact header, the callee SHOULD
send a BYE request to that address rather than the From address.
This method MUST be supported by proxy servers and SHOULD be
supported by redirect and user agent SIP servers.
4.2.5 CANCEL
The CANCEL request cancels a pending request with the same Call-ID,
To, From and CSeq (sequence number only) header field values, but
does not affect a completed request or existing calls. (A request is
considered completed if the server has returned a final status
response.)
A user agent client or proxy client MAY issue a CANCEL request at any
time. A proxy, in particular, MAY choose to send a CANCEL to
destinations that have not yet returned a final response after it has
received a 2xx or 6xx response for one or more of the parallel-search
requests. A proxy that receives a CANCEL request forwards the request
to all destinations with pending requests.
The Call-ID, To, the numeric part of CSeq and From header fields in
the CANCEL request are identical to those in the original request.
This allows a CANCEL request to be matched with the request it
cancels. However, to allow the client to distinguish responses to the
CANCEL from those to the original request, the CSeq Method component
is set to CANCEL. The Via header field is initialized to the proxy
issuing the CANCEL request. (Thus, responses to this CANCEL request
only reach the issuing proxy.)
The behavior of the user agent or redirect server on receiving a
CANCEL request depends on whether the server has already sent a final
response for the original request. If it has, the CANCEL request has
no effect on the original request, any call state and on the
responses generated for the original request. If the server has not
issued a final response for the original request, it immediately
sends a 487 (Request Terminated) for the original request. The UAC
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confirms receipt of any final response for the original request as
normal with an ACK request. CANCEL request itself is answered with a
200 (OK) response in either case.
The BYE request cannot be used to cancel branches of a
parallel search, since several branches may, through
intermediate proxies, find the same user agent server and
then terminate the call. To terminate a call instead of
just pending searches, the UAC must use BYE instead of or
in addition to CANCEL. While CANCEL can terminate any
pending request other than ACK or CANCEL, it is typically
useful only for INVITE. 200 responses to INVITE and 200
responses to CANCEL can be distinguished by the method in
the Cseq header field.
This method MUST be supported by proxy servers and SHOULD be
supported by all other SIP server types.
4.2.6 REGISTER
A client uses the REGISTER method to register the address listed in
the To header field with a SIP server.
A user agent SHOULD register with a local server on startup and
periodically thereafter by sending a REGISTER request. The period is
given by the expiration time indicated in the registration response.
It is RECOMMENDED that the UA registers via multicast and send a
registration to its "home" address, i.e., the server for the domain
that it uses as its From address in outgoing requests.
Multicast registrations are addressed to the well-known "all SIP
servers" multicast address "sip.mcast.net" (224.0.1.75). This request
SHOULD be scoped to ensure it is not forwarded beyond the boundaries
of the administrative system. This MAY be done with either TTL or
administrative scopes [29], depending on what is implemented in the
network. SIP user agents MAY listen to that address and use it to
become aware of the location of other local users [20]; however, they
do not respond to the request.
Since registrations expire, clients need to periodically refresh
their registrations. Such refreshes SHOULD be sent to the same
address as the original registration, unless redirected.
Registration requests may be redirected by a 3xx response, or with a
Contact header in a non-200 response (typically, 401) to a REGISTER
request. Refreshes for the same address of record SHOULD be directed
to this new address for all subsequent registrations. A client MAY
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revert to the original address upon reboot or upon an
administratively configured lifetime. This implies that servers MUST
be prepared to accept registrations on the original configured
address and the redirected address.
It is RECOMMENDED that clients ignore redirection responses from
untrusted hosts or require that such redirection responses are
cryptographically signed.
Multicast registration may be inappropriate in some
environments, for example, if multiple businesses share the
same local area network.
Requests are processed in the order received. Clients SHOULD avoid
sending a new registration (as opposed to a retransmission) until
they have received the response from the server for the previous one.
Clients may register from different locations, by necessity
using different Call-ID values. Thus, the CSeq value cannot
be used to enforce ordering. Since registrations are
additive, ordering is less of a problem than if each
REGISTER request completely replaced all earlier ones.
The meaning of the REGISTER request-header fields is defined as
follows. We define "address-of-record" as the SIP address that the
registry knows the registrand, typically of the form "user@domain"
rather than "user@host". In third-party registration, the entity
issuing the request is different from the entity being registered.
To: The To header field contains the address-of-record whose
registration is to be created or updated.
From: The From header field contains the address-of-record of
the person responsible for the registration. For first-
party registration, it is identical to the To header field
value.
Request-URI: The Request-URI names the destination of the
registration request, i.e., the domain of the registrar.
The user name MUST be empty. Generally, the domains in the
Request-URI and the To header field have the same value;
however, it is possible to register as a "visitor", while
maintaining one's name. For example, a traveler
sip:alice@acme.com (To) might register under the Request-
URI sip:atlanta.hiayh.org , with the former as the To
header field and the latter as the Request-URI. The
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REGISTER request is no longer forwarded once it has reached
the server whose authoritative domain is the one listed in
the Request-URI.
Call-ID: All registrations from a client SHOULD use the same
Call-ID header value, at least within the same reboot
cycle.
Cseq: Registrations with the same Call-ID MUST have increasing
CSeq header values. However, the server does not reject
out-of-order requests.
Contact: The request MAY contain a Contact header field. Future
non-REGISTER requests for the URI given in the To header
field SHOULD be directed to the address(es) given in the
Contact header.
If the request does not contain a Contact header, the
registration remains unchanged.
This is useful to obtain the current list of
registrations in the response. If a SIP URI in a
registration Contact header field differs from
existing registrations according to the rules in
Section 2.1, it is added to the list of registration.
If it is equivalent, according to these rules, to an
existing registration, all Contact header field
parameters for this entry are updated accordingly.
URIs other than SIP URIs are compared according to the
standard URI equivalency rules for the URI schema.
All current registrations MUST share the same action value.
Registrations that have a different action than current
registrations for the same user MUST be rejected with
status of 409 (Conflict).
A proxy server ignores the q parameter when processing
non-REGISTER requests, while a redirect server simply
returns that parameter in its Contact response header
field.
Having the proxy server interpret the q parameter is
not sufficient to guide proxy behavior, as it is not
clear, for example, how long it is supposed to wait
between trying addresses.
If the registration is changed while a user agent or proxy
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server processes an invitation, the new information SHOULD
be used.
This allows a service known as "directed pick-up". In
the telephone network, directed pickup permits a user
at a remote station who hears his own phone ringing to
pick up at that station, dial an access code, and be
connected to the calling user as if he had answered
his own phone.
Responses MAY contain a Contact header field. Its
interpretation differs depending on the response status. If
contained within a 2xx response, the header field indicates
the list of current registrations. In 3xx and 4xx
responses, it indicates where future REGISTER requests
should be directed.
A server MAY choose any duration for the registration lifetime.
Registrations not refreshed after this amount of time SHOULD be
silently discarded. Responses to a registration SHOULD include an
Expires header (Section 6.24) or expires Contact parameters (Section
6.15), indicating the time at which the server will drop the
registration. If none is present, one hour is assumed. Clients MAY
request a registration lifetime by indicating the time in an Expires
header in the request. A server SHOULD NOT use a higher lifetime than
the one requested, but MAY use a lower one. A single address (if
host-independent) MAY be registered from several different clients.
A client cancels an existing registration by sending a REGISTER
request with an expiration time (Expires) of zero seconds, either for
a particular Contact or the wildcard Contact designated by a "*" if
it wants to cancel all registrations. Registrations are matched based
on the user, host, port and maddr parameters.
The server SHOULD return the current list of registrations in the 200
response as Contact header fields.
It is particularly important that REGISTER requests are authenticated
since they allow to redirect future requests (see Section 13.2).
Beyond its use as a simple location service, this method is
needed if there are several SIP servers on a single host.
In that case, only one of the servers can use the default
port number.
Support of this method is RECOMMENDED.
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4.3 Request-URI
The Request-URI is a SIP URL as described in Section 2 or a general
URI (RFC 2396 [12]). In particular, it MUST NOT contain unescaped
spaces or control characters. It indicates the user or service to
which this request is being addressed. Unlike the To field, the
Request-URI MAY be re-written by proxies.
As shown in Table 2, the Request-URI MAY contain the user-param
parameter as well as transport-related parameters. A server that
receives a SIP-URL with illegal elements removes them before further
processing.
Transport-related parameters are needed when a UAC proxies
all requests to a default proxy, which would then need this
information to generate the appropriate request.
Typically, the UAC sets the Request-URI and To to the same
SIP URL, presumed to remain unchanged over long time
periods. However, if the UAC has cached a more direct path
to the callee, e.g., from the Contact header field of a
response to a previous request, the To would still contain
the long-term, "public" address, while the Request-URI
would be set to the cached address.
Proxy and redirect servers MAY use the information in the Request-URI
and request header fields to handle the request and possibly rewrite
the Request-URI. For example, a request addressed to the generic
address sip:sales@acme.com is proxied to the particular person, e.g.,
sip:bob@ny.acme.com , with the To field remaining as
sip:sales@acme.com. At ny.acme.com , Bob then designates Alice as
the temporary substitute.
The host part of the Request-URI typically agrees with one of the
host names of the receiving server. If it does not, the server SHOULD
proxy the request to the address indicated or return a 404 (Not
Found) response if it is unwilling or unable to do so. For example,
the Request-URI and server host name can disagree in the case of a
firewall proxy that handles outgoing calls. This mode of operation is
similar to that of HTTP proxies.
SIP servers MAY support Request-URIs with schemes other than "sip",
for example the "tel" URI scheme [21]. It MAY translate non-SIP URIs
using any mechanism at its disposal, resulting in either a SIP URI or
some other scheme.
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If a SIP server receives a request with a URI indicating a scheme the
server does not understand, the server MUST return a 400 (Bad
Request) response. It MUST do this even if the To header field
contains a scheme it does understand, since proxies are responsible
for processing the Request-URI. (The To field is only of interest to
the UAS.)
4.3.1 SIP Version
Both request and response messages include the version of SIP in use,
and follow [H3.1] (with HTTP replaced by SIP, and HTTP/1.1 replaced
by SIP/2.0) regarding version ordering, compliance requirements, and
upgrading of version numbers. To be compliant with this
specification, applications sending SIP messages MUST include a SIP-
Version of "SIP/2.0".
4.4 Option Tags
Option tags are unique identifiers used to designate new options in
SIP. These tags are used in Require (Section 6.36), Supported
(Section 6.42) and Unsupported (Section 6.45) header fields.
Syntax:
option-tag = token
See Section C for the definition of token. The creator of a new SIP
option MUST either prefix the option with their reverse domain name
or register the new option with the Internet Assigned Numbers
Authority (IANA).
An example of a reverse-domain-name option is "com.foo.mynewfeature",
whose inventor can be reached at "foo.com". For these features,
individual organizations are responsible for ensuring that option
names do not collide within the same domain. The host name part of
the option MUST use lower-case; the option name is case-sensitive.
Options registered with IANA do not contain periods and are globally
unique. IANA option tags are case-sensitive.
4.4.1 Registering New Option Tags with IANA
When registering a new SIP option, the following information MUST be
provided:
o Name and description of option. The name MAY be of any length,
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but SHOULD be no more than twenty characters long. The name
MUST consist of alphanum (See Figure 3) characters only;
o A listing of any new SIP header fields, header parameter
fields or parameter values defined by this option. A SIP
option MUST NOT redefine header fields or parameters defined
in either RFC 2543, any standards-track extensions to RFC
2543, or other extensions registered through IANA.
o Indication of who has change control over the option (for
example, IETF, ISO, ITU-T, other international standardization
bodies, a consortium or a particular company or group of
companies);
o A reference to a further description, if available, for
example (in order of preference) an RFC, a published paper, a
patent filing, a technical report, documented source code or a
computer manual;
o Contact information (postal and email address).
Registrations should be sent to iana@iana.org
This procedure has been borrowed from RTSP [4] and the RTP
AVP [30].
5 Response
After receiving and interpreting a request message, the recipient
responds with a SIP response message. The response message format is
shown below:
Response = Status-Line ; Section 5.1
*( general-header
| response-header
| entity-header )
CRLF
[ message-body ] ; Section 8
SIP's structure of responses is similar to [H6], but is defined
explicitly here.
5.1 Status-Line
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The first line of a Response message is the Status-Line, consisting
of the protocol version (Section 4.3.1) followed by a numeric
Status-Code and its associated textual phrase, with each element
separated by SP characters. No CR or LF is allowed except in the
final CRLF sequence.
Status-Line = SIP-version SP Status-Code SP Reason-Phrase CRLF
5.1.1 Status Codes and Reason Phrases
The Status-Code is a 3-digit integer result code that indicates the
outcome of the attempt to understand and satisfy the request. The
Reason-Phrase is intended to give a short textual description of the
Status-Code. The Status-Code is intended for use by automata, whereas
the Reason-Phrase is intended for the human user. The client is not
required to examine or display the Reason-Phrase.
Status-Code = Informational ;Fig. 4
| Success ;Fig. 4
| Redirection ;Fig. 5
| Client-Error ;Fig. 6
| Server-Error ;Fig. 7
| Global-Failure ;Fig. 8
| extension-code
extension-code = 3DIGIT
Reason-Phrase = *
We provide an overview of the Status-Code below, and provide full
definitions in Section 7. The first digit of the Status-Code defines
the class of response. The last two digits do not have any
categorization role. SIP/2.0 allows 6 values for the first digit:
1xx: Informational -- request received, continuing to process
the request;
2xx: Success -- the action was successfully received,
understood, and accepted;
3xx: Redirection -- further action needs to be taken in order to
complete the request;
4xx: Client Error -- the request contains bad syntax or cannot
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be fulfilled at this server;
5xx: Server Error -- the server failed to fulfill an apparently
valid request;
6xx: Global Failure -- the request cannot be fulfilled at any
server.
Figures 4 through 8 present the individual values of the numeric
response codes, and an example set of corresponding reason phrases
for SIP/2.0. These reason phrases are only recommended; they may be
replaced by local equivalents without affecting the protocol. Note
that SIP adopts many HTTP/1.1 response codes. SIP/2.0 adds response
codes in the range starting at x80 to avoid conflicts with newly
defined HTTP response codes, and adds a new class, 6xx, of response
codes.
SIP response codes are extensible. SIP applications are not required
to understand the meaning of all registered response codes, though
such understanding is obviously desirable. However, applications MUST
understand the class of any response code, as indicated by the first
digit, and treat any unrecognized response as being equivalent to the
x00 response code of that class, with the exception that an
unrecognized response MUST NOT be cached. For example, if a client
receives an unrecognized response code of 431, it can safely assume
that there was something wrong with its request and treat the
response as if it had received a 400 (Bad Request) response code. In
such cases, user agents SHOULD present to the user the message body
returned with the response, since that message body is likely to
include human-readable information which will explain the unusual
status.
Informational = "100" ; Trying
| "180" ; Ringing
| "181" ; Call Is Being Forwarded
| "182" ; Queued
| "183" ; Session Progress
Success = "200" ; OK
Figure 4: Informational and success status codes
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Redirection = "300" ; Multiple Choices
| "301" ; Moved Permanently
| "302" ; Moved Temporarily
| "305" ; Use Proxy
| "380" ; Alternative Service
Figure 5: Redirection status codes
Client-Error = "400" ; Bad Request
| "401" ; Unauthorized
| "402" ; Payment Required
| "403" ; Forbidden
| "404" ; Not Found
| "405" ; Method Not Allowed
| "406" ; Not Acceptable
| "407" ; Proxy Authentication Required
| "408" ; Request Timeout
| "409" ; Conflict
| "410" ; Gone
| "411" ; Length Required
| "413" ; Request Entity Too Large
| "414" ; Request-URI Too Large
| "415" ; Unsupported Media Type
| "420" ; Bad Extension
| "480" ; Temporarily not available
| "481" ; Call Leg/Transaction Does Not Exist
| "482" ; Loop Detected
| "483" ; Too Many Hops
| "484" ; Address Incomplete
| "485" ; Ambiguous
| "486" ; Busy Here
| "487" ; Request Cancelled
| "488" ; Not Acceptable Here
Figure 6: Client error status codes
6 Header Field Definitions
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Server-Error = "500" ; Internal Server Error
| "501" ; Not Implemented
| "502" ; Bad Gateway
| "503" ; Service Unavailable
| "504" ; Gateway Time-out
| "505" ; SIP Version not supported
Figure 7: Server error status codes
Global-Failure = "600" ; Busy Everywhere
| "603" ; Decline
| "604" ; Does not exist anywhere
| "606" ; Not Acceptable
Figure 8: Global failure status codes
SIP header fields are similar to HTTP header fields in both syntax
and semantics. In particular, SIP header fields follow the syntax for
message-header as described in [H4.2]. The rules for extending header
fields over multiple lines, and use of multiple message-header fields
with the same field-name, described in [H4.2] also apply to SIP. The
rules in [H4.2] regarding ordering of header fields apply to SIP,
with the exception of Via fields, see below, whose order matters.
Additionally, if authentication is used, the rules in Section 13.2
apply. Proxies SHOULD NOT reorder header fields. Proxies add Via
header fields and MAY add other hop-by-hop header fields. They can
modify certain header fields, such as Max-Forwards (Section 6.28) and
"fix up" the Via header fields with "received" parameters as
described in Section 6.47.1. Proxies MUST NOT alter any fields that
are authenticated (see Section 13.2).
The header fields required, optional and not applicable for each
method are listed in Table 4 and Table 5. The table uses "o" to
indicate optional, "m" mandatory and "-" for not applicable. A "*"
indicates that the header fields are needed only if message body is
not empty. See sections 6.19, 6.20 and 8 for details.
The "where" column describes the request and response types with
which the header field can be used. "R" refers to header fields that
can be used in requests (that is, request and general header fields).
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"r" designates a response or general-header field as applicable to
all responses, while a list of numeric values indicates the status
codes with which the header field can be used. "g" and "e" designate
general (Section 6.1) and entity header (Section 6.2) fields,
respectively. If a header field is marked "c", it is copied from the
request to the response.
The "enc." column describes whether this message header field MAY be
encrypted end-to-end. A "n" designates fields that MUST NOT be
encrypted, while "e" designates fields that SHOULD be encrypted if
encryption is used.
The "e-e" column has a value of "e" for end-to-end and a value of "h"
for hop-by-hop header fields.
Other header fields can be added as required; a server MUST ignore
header fields not defined in this specification that it does not
understand. A proxy MUST NOT remove or modify header fields not
defined in this specification that it does not understand. A compact
form of these header fields is also defined in Section 9 for use over
UDP when the request has to fit into a single packet and size is an
issue.
Table 6 in Appendix A lists those header fields that different client
and server types MUST be able to parse.
6.1 General Header Fields
General header fields apply to both request and response messages.
The "general-header" field names can be extended reliably only in
combination with a change in the protocol version. However, new or
experimental header fields MAY be given the semantics of general
header fields if all parties in the communication recognize them to
be "general-header" fields. Unrecognized header fields are treated as
"entity-header" fields.
6.2 Entity Header Fields
The "entity-header" fields define meta-information about the
message-body or, if no body is present, about the resource identified
by the request. The term "entity header" is an HTTP 1.1 term where
the response body can contain a transformed version of the message
body. The original message body is referred to as the "entity". We
retain the same terminology for header fields but usually refer to
the "message body" rather then the entity as the two are the same in
SIP.
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where enc. e-e ACK BYE CAN INV OPT REG
_____________________________________________________________
Accept R e - o o o o o
Accept 415 e - o o o o o
Accept r e - - - o o o
Accept-Encoding R e - o o o o o
Accept-Encoding 415 e - o o o o o
Accept-Language R e - o o o o o
Accept-Language 415 e - o o o o o
Alert-Info R e e - - - o - -
Allow 200 e - - - o o o
Allow 405 e m m m m m m
Authorization R e o o o o o o
Authorization r e o o o o o o
Call-ID gc n e m m m m m m
Call-Info g e e - - - o o o
Contact R e o - - o o o
Contact 1xx e - - - o o -
Contact 2xx e - - - o o o
Contact 3xx e - o - o o o
Contact 485 e - o - o o o
Content-Disposition e e o o - o o o
Content-Encoding e e o o - o o o
Content-Language e e m m m m m m
Content-Length e n e m m m m m m
Content-Type e e * * - * * *
CSeq gc n e m m m m m m
Date g e o o o o o o
Encryption g n e o o o o o o
Expires g e - - - o - o
From gc n e m m m m m m
Hide R n h o o o o o o
In-Reply-To R e e - - - o - -
Max-Forwards R n e o o o o o o
MIME-Version g n e o o o o o o
Organization g e h - - - o o o
Table 4: Summary of header fields, A--O
6.3 Request Header Fields
The "request-header" fields allow the client to pass additional
information about the request, and about the client itself, to the
server. These fields act as request modifiers, with semantics
equivalent to the parameters of a programming language method
invocation.
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where enc. e-e ACK BYE CAN INV OPT REG
_______________________________________________________________________
Priority R c e - - - o - -
Proxy-Authenticate 401,407 n h o o o o o o
Proxy-Authorization R n h o o o o o o
Proxy-Require R n h o o o o o o
Record-Route R h o o o o o o
Record-Route 2xx,401,484 h o o o o o o
Require g e o o o o o o
Response-Key R c e - o o o o o
Retry-After R c e - - - - - o
Retry-After 404,413,480,486 c e o o o o o o
500,503 c e o o o o o o
600,603 c e o o o o o o
Route R h o o o o o o
Server r c e o o o o o o
Subject R c e - - - o - -
Supported g c e - o o o o o
Timestamp g e o o o o o o
To gc(1) n e m m m m m m
Unsupported R e o o o o o o
Unsupported 420 e o o o o o o
User-Agent g c e o o o o o o
Via gc(2) n e m m m m m m
Warning r e o o o o o o
WWW-Authenticate R c e o o o o o o
WWW-Authenticate 401 c e o o o o o o
Table 5: Summary of header fields, P--Z; (1): copied with possible
addition of tag; (2): UAS removes first Via header field
The "request-header" field names can be extended reliably only in
combination with a change in the protocol version. However, new or
experimental header fields MAY be given the semantics of "request-
header" fields if all parties in the communication recognize them to
be request-header fields. Unrecognized header fields are treated as
"entity-header" fields.
6.4 Response Header Fields
The "response-header" fields allow the server to pass additional
information about the response which cannot be placed in the Status-
Line. These header fields give information about the server and about
further access to the resource identified by the Request-URI.
Response-header field names can be extended reliably only in
combination with a change in the protocol version. However, new or
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experimental header fields MAY be given the semantics of "response-
header" fields if all parties in the communication recognize them to
be "response-header" fields. Unrecognized header fields are treated
as "entity-header" fields.
6.5 End-to-end and Hop-by-hop Headers
End-to-end headers MUST be transmitted unmodified across all proxies,
while hop-by-hop headers MAY be modified or added by proxies.
6.6 Header Field Format
Header fields ("general-header", "request-header", "response-header",
and "entity-header") follow the same generic header format as that
given in Section 3.1 of RFC 822 [28]. Each header field consists of a
name followed by a colon (":") and the field value. Field names are
case-insensitive. The field value MAY be preceded by any amount of
leading white space (LWS), though a single space (SP) is preferred.
Header fields can be extended over multiple lines by preceding each
extra line with at least one SP or horizontal tab (HT). Applications
MUST follow HTTP "common form" when generating these constructs,
since there might exist some implementations that fail to accept
anything beyond the common forms.
message-header = field-name ":" [ field-value ] CRLF
field-name = token
field-value = *( field-content | LWS )
field-content = < the OCTETs making up the field-value
and consisting of either *TEXT-UTF8
or combinations of token,
separators, and quoted-string>
The relative order of header fields with different field names is not
significant. Multiple header fields with the same field-name may be
present in a message if and only if the entire field-value for that
header field is defined as a comma-separated list (i.e., #(values)).
It MUST be possible to combine the multiple header fields into one
"field-name: field-value" pair, without changing the semantics of the
message, by appending each subsequent field-value to the first, each
separated by a comma. The order in which header fields with the same
field-name are received is therefore significant to the
interpretation of the combined field value, and thus a proxy MUST NOT
change the order of these field values when a message is forwarded.
Unless otherwise noted, comparisons are case-sensitive. However, any
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parameters preceded by semicolons are case-insensitive, while any
quoted string literals are compared taken case into consideration.
The Contact, From and To header fields contain a URL. If the URL
contains a comma, question mark or semicolon, the URL MUST be
enclosed in angle brackets (< and >). Any URL parameters are
contained within these brackets. If the URL is not enclosed in angle
brackets, any semicolon-delimited parameters are header-parameters,
not URL parameters.
6.7 Accept
The Accept header follows the syntax defined in [H14.1]. The
semantics are also identical, with the exception that if no Accept
header is present, the server SHOULD assume a default value of
application/sdp
As a request-header field, it is used only with those methods that
take message bodies. In a 415 (Unsupported Media Type) response, it
indicates which content types are acceptable in requests. In 200 (OK)
responses for INVITE, it lists the content types acceptable for
future requests in this call.
Example:
Accept: application/sdp;level=1, application/x-private, text/html
6.8 Accept-Encoding
The Accept-Encoding request-header field is similar to Accept, but
restricts the content-codings [H3.4.1] that are acceptable in the
response. See [H14.3]. The syntax of this header is defined in
[H14.3]. The semantics in SIP are identical to those defined in
[H14.3].
Note: An empty Accept-Encoding header field is permissible,
even though the syntax in [H14.3] does not provide for it.
It is equivalent to Accept-Encoding: identity, i.e., only
the identity encoding, meaning no encoding, is permissible.
If no Accept-Encoding header field is present in a request, the
server MUST use the "identity" encoding.
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HTTP/1.1 [H14.3] states that the server SHOULD use the
"identity" encoding unless it has additional information
about the capabilities of the client. This is needed for
backwards-compatibility with old HTTP clients and does not
affect SIP.
6.9 Accept-Language
The Accept-Language header follows the syntax defined in [H14.4]. The
rules for ordering the languages based on the q parameter apply to
SIP as well. When used in SIP, the Accept-Language request-header
field can be used to allow the client to indicate to the server in
which language it would prefer to receive reason phrases, session
descriptions or status responses carried as message bodies. A proxy
MAY use this field to help select the destination for the call, for
example, a human operator conversant in a language spoken by the
caller.
Example:
Accept-Language: da, en-gb;q=0.8, en;q=0.7
6.10 Alert-Info
The Alert-Info header field indicates that the content indicated in
the URLs should be rendered instead of ring tone. A user SHOULD be
able to disable this feature selectively to prevent unauthorized
disruptions.
Alert-Info = "Alert-Info" ":" # ( "<" URI ">" *(";" generic-param ))
Example:
Alert-Info:
6.11 Allow
The Allow entity-header field lists the set of methods supported by
the resource identified by the Request-URI. The purpose of this field
is strictly to inform the recipient of valid methods associated with
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the resource. An Allow header field MUST be present in a 405 (Method
Not Allowed) response, SHOULD be present in an OPTIONS response
SHOULD be present in the 200 (OK) response to the initial INVITE for
a call and MAY be present in final responses for other methods.
Supplying an Allow header in responses to methods other
than OPTIONS cuts down on the number of messages needed.
Allow = "Allow" ":" 1#Method
6.12 Authorization
A user agent that wishes to authenticate itself with a UAS or
registrar -- usually, but not necessarily, after receiving a 401
response -- MAY do so by including an Authorization request-header
field with the request. The Authorization field value consists of
credentials containing the authentication information of the user
agent for the realm of the resource being requested.
Section 13.2 overviews the use of the Authorization header, and
section 15 describes the syntax and semantics when used with PGP
based authentication.
6.13 Call-ID
The Call-ID general-header field uniquely identifies a particular
invitation or all registrations of a particular client. Note that a
single multimedia conference can give rise to several calls with
different Call-IDs, e.g., if a user invites a single individual
several times to the same (long-running) conference.
For an INVITE request, a callee user agent server SHOULD NOT alert
the user if the user has responded previously to the Call-ID in the
INVITE request. If the user is already a member of the conference and
the conference parameters contained in the session description have
not changed, a callee user agent server MAY silently accept the call,
regardless of the Call-ID. An invitation for an existing Call-ID or
session can change the parameters of the conference. A client
application MAY decide to simply indicate to the user that the
conference parameters have been changed and accept the invitation
automatically or it MAY require user confirmation.
A user may be invited to the same conference or call using several
different Call-IDs. If desired, the client MAY use identifiers within
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the session description to detect this duplication. For example, SDP
contains a session id and version number in the origin (o) field.
The REGISTER and OPTIONS methods use the Call-ID value (in addition
to the CSeq value) to unambiguously match requests and responses. All
REGISTER requests issued by a single client SHOULD use the same
Call-ID, at least within the same boot cycle. For these requests, it
makes no difference whether the Call-ID value matches an existing
call or not.
Since the Call-ID is generated by and for SIP, there is no
reason to deal with the complexity of URL-encoding and
case-ignoring string comparison.
callid = token [ "@" token ]
Call-ID = ( "Call-ID" | "i" ) ":" callid
The callid MUST be a globally unique identifier and MUST NOT be
reused for later calls. Use of cryptographically random identifiers
[31] is RECOMMENDED. Implementations MAY use the form "localid@host".
Call-IDs are case-sensitive and are simply compared byte-by-byte.
Using cryptographically random identifiers provides some
protection against session hijacking. Call-ID, To and From
are needed to identify a call leg. The distinction between
call and call leg matters in calls with third-party
control.
For systems which have tight bandwidth constraints, many of the
mandatory SIP headers have a compact form, as discussed in Section 9.
These are alternate names for the headers which occupy less space in
the message. In the case of Call-ID, the compact form is i.
For example, both of the following are valid:
Call-ID: f81d4fae-7dec-11d0-a765-00a0c91e6bf6@foo.bar.com
or
i:f81d4fae-7dec-11d0-a765-00a0c91e6bf6@foo.bar.com
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6.14 Call-Info
The Call-Info general header field provides additional information
about the caller or callee, depending on whether it is found in a
request or response. The purpose of the URI is described by the
"purpose" parameter. "icon" designates an image suitable as an iconic
representation of the caller or callee; "info" describes the caller
or callee in general, e.g., through a web page; "card" provides a
business card (e.g., in vCard [32] or LDIF [33] formats).
Call-Info = "Call-Info" ":" # ( "<" URI ">"
[ ";" purpose "=" ( "icon" | "info" | "card" | token) ]
*(";" generic-param ))
Example:
Call-Info: ;purpose=icon,
;purpose=info
6.15 Contact
The Contact general-header field can appear in INVITE, OPTIONS, ACK,
and REGISTER requests, and in 1xx, 2xx, 3xx, and 485 responses. In
general, it provides a URL where the user can be reached for further
communications.
In some of the cases below, the client uses information from the
Contact header field in Request-URI of future requests. In these
cases, the client copies all but the "method-param" and "header"
elements of the addr-spec part of the Contact header field into the
Request-URI of the request. It uses the "header" parameters to create
headers for the request, replacing any default headers normally used.
Unless the client is configured to use a default proxy for all
outgoing requests, it then directs the request to the address and
port specified by the "maddr" and "port" parameters, using the
transport protocol given in the "transport" parameter. If "maddr" is
a multicast address, the value of "ttl" is used as the time-to-live
value.
INVITE, OPTIONS and ACK requests: INVITE requests MUST and ACK
requests MAY contain Contact headers indicating from which
location the request is originating. The URL in the Contact
header field is then used by subsequent requests from the
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callee. For OPTIONS, Contact provides a hint where future
SIP requests can be sent or the user can be contacted via
non-SIP means.
This allows the callee to send future requests, such
as BYE, directly to the caller instead of through a
series of proxies. The Via header is not sufficient
since the desired address may be that of a proxy.
INVITE 1xx responses: A UAS sending a provisional response (1xx)
MAY insert a Contact response header. It has the same
semantics in a 1xx response as a 2xx INVITE response. Note
that CANCEL requests MUST NOT be sent to that address, but
rather follow the same path as the original request.
INVITE and OPTIONS 2xx responses: A user agent server sending a
definitive, positive response (2xx) MUST insert a Contact
response header field indicating the SIP address under
which it is reachable most directly for future SIP
requests, such as ACK, within the same Call-ID. The Contact
header field contains the address of the server itself or
that of a proxy, e.g., if the host is behind a firewall.
The value of this Contact header is copied into the
Request-URI of subsequent requests for this call if the
response did not also contain a Record-Route header. If the
response also contains a Record-Route header field, the
address in the Contact header field is added as the last
item in the Route header field. See Section 6.35 for
details.
If a UA supports both UDP and TCP, it SHOULD NOT indicate a
transport parameter in the URI.
The Contact value SHOULD NOT be cached across calls,
as it may not represent the most desirable location
for a particular destination address.
REGISTER requests: REGISTER requests MAY contain a Contact
header field indicating at which locations the user is
reachable. The REGISTER request defines a wildcard Contact
field, "*", which MUST only be used with Expires: 0 to
remove all registrations for a particular user. An optional
"expires" parameter indicates the desired expiration time
of the registration. If a Contact entry does not have an
"expires" parameter, the Expires header field is used as
the default value. If neither of these mechanisms is used,
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SIP URIs are assumed to expire after one hour. Other URI
schemes have no expiration times.
REGISTER 2xx responses: A REGISTER response MAY return all
locations at which the user is currently reachable. An
optional "expires" parameter indicates the expiration time
of the registration. If a Contact entry does not have an
"expires" parameter, the value of the Expires header field
indicates the expiration time. If neither mechanism is
used, the expiration time specified in the request,
explicitly or by default, is used.
3xx and 485 responses: The Contact response-header field can be
used with a 3xx or 485 (Ambiguous) response codes to
indicate one or more alternate addresses to try. It can
appear in responses to BYE, INVITE and OPTIONS methods. The
Contact header field contains URIs giving the new locations
or user names to try, or may simply specify additional
transport parameters. A 300 (Multiple Choices), 301 (Moved
Permanently), 302 (Moved Temporarily) or 485 (Ambiguous)
response SHOULD contain a Contact field containing URIs of
new addresses to be tried. A 301 or 302 response may also
give the same location and username that was being tried
but specify additional transport parameters such as a
different server or multicast address to try or a change of
SIP transport from UDP to TCP or vice versa. The client
copies information from the Contact header field into the
Request-URI as described above.
4xx, 5xx and 6xx responses: The Contact response-header field
can be used with a 4xx, 5xx or 6xx response to indicate the
location where additional information about the error can
be found.
Note that the Contact header field MAY also refer to a different
entity than the one originally called. For example, a SIP call
connected to GSTN gateway may need to deliver a special information
announcement such as "The number you have dialed has been changed."
A Contact response header field can contain any suitable URI
indicating where the called party can be reached, not limited to SIP
URLs. For example, it could contain URL's for phones, fax, or irc (if
they were defined) or a mailto: (RFC 2368, [34]) URL.
The following parameters are defined. Additional parameters may be
defined in other specifications.
q: The "qvalue" indicates the relative preference among the
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locations given. "qvalue" values are decimal numbers from 0
to 1, with higher values indicating higher preference.
action: The "action" parameter is used only when registering
with the REGISTER request. It indicates whether the client
wishes that the server proxy or redirect future requests
intended for the client. If this parameter is not specified
the action taken depends on server configuration. In its
response, the registrar SHOULD indicate the mode used. This
parameter is ignored for other requests.
expires: The "expires" parameter indicates how long the URI is
valid. The parameter is either a number indicating seconds
or a quoted string containing a SIP-date. If this parameter
is not provided, the value of the Expires header field
determines how long the URI is valid. Implementations MAY
treat values larger than 2**32-1 (4294967295 seconds or 136
years) as equivalent to 2**32-1.
Contact = ( "Contact" | "m" ) ":"
("*" | (1# (( name-addr | addr-spec )
*( ";" contact-params ) )))
name-addr = [ display-name ] "<" addr-spec ">"
addr-spec = SIP-URL | URI
display-name = *token | quoted-string
contact-params = "q" "=" qvalue
| "action" "=" "proxy" | "redirect"
| "expires" "=" delta-seconds | <"> SIP-date <">
| contact-extension
contact-extension = generic-param
Even if the "display-name" is empty, the "name-addr" form MUST be
used if the "addr-spec" contains a comma, semicolon or question mark.
Note that there may or may not be LWS between the display-name and
the "<".
The Contact header field fulfills functionality similar to
the Location header field in HTTP. However, the HTTP header
only allows one address, unquoted. Since URIs can contain
commas and semicolons as reserved characters, they can be
mistaken for header or parameter delimiters, respectively.
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The current syntax corresponds to that for the To and From
header, which also allows the use of display names.
Example:
Contact: "Mr. Watson"
;q=0.7; expires=3600,
"Mr. Watson" ;q=0.1
6.16 Content-Disposition
Content-Disposition = "Content-Disposition" ":"
disposition-type *(";" disposition-parm)
disposition-type = "render" | "session" | extension-token
disposition-parm = handling-parm | parameter
handling-parm = "handling" "="
( "optional" | "required" | other-handling )
other-handling = token
The Content-Disposition header field describes how the message body
or, in the case of multipart messages, a message body part is to be
interpreted by the UAC or UAS. The SIP header extends the MIME
Content-Type (RFC 1806 [35]).
The value "session" indicates that the body part describes a session.
The value "render" indicates that the body part should be displayed
or otherwise rendered to the user. For backward-compatibility,
message bodies in requests and 2xx responses are assumed to have the
value "session", while those in other responses have the value
"render".
The handling parameter, handling-parm, describes how the UAS should
react if it receives a message body whose content type or disposition
type it does not understand. If the parameter has the value
"optional", the UAS MUST ignore the message body; if it has the value
"required", the UAS MUST return 415 (Unsupported Media Type).
6.17 Content-Encoding
Content-Encoding = ( "Content-Encoding" | "e" ) ":"
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1#content-coding
The Content-Encoding entity-header field is used as a modifier to the
"media-type". When present, its value indicates what additional
content codings have been applied to the entity-body, and thus what
decoding mechanisms MUST be applied in order to obtain the media-type
referenced by the Content-Type header field. Content-Encoding is
primarily used to allow a body to be compressed without losing the
identity of its underlying media type.
If multiple encodings have been applied to an entity, the content
codings MUST be listed in the order in which they were applied.
All content-coding values are case-insensitive. The Internet Assigned
Numbers Authority (IANA) acts as a registry for content-coding value
tokens. See [3.5] for a definition of the syntax for content-coding.
Clients MAY apply content encodings to the body in requests. If the
server is not capable of decoding the body, or does not recognize any
of the content-coding values, it MUST send a 415 "Unsupported Media
Type" response, listing acceptable encodings in the Accept-Encoding
header. A server MAY apply content encodings to the bodies in
responses. The server MUST only use encodings listed in the Accept-
Encoding header in the request.
6.18 Content-Language
See [H14.12].
6.19 Content-Length
The Content-Length entity-header field indicates the size of the
message-body, in decimal number of octets, sent to the recipient.
Content-Length = ( "Content-Length" | "l" ) ":" 1*DIGIT
An example is
Content-Length: 3495
Applications SHOULD use this field to indicate the size of the
message-body to be transferred, regardless of the media type of the
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entity. Any Content-Length greater than or equal to zero is a valid
value. If no body is present in a message, then the Content-Length
header field MUST be set to zero. If a server receives a UDP request
without Content-Length, it MUST assume that the request encompasses
the remainder of the packet. If a server receives a UDP request with
a Content-Length, but the value is larger than the size of the body
sent in the request, the client SHOULD generate a 400 class response.
If there is additional data in the UDP packet after the last byte of
the body has been read, the server MUST treat the remaining data as a
separate message. This allows several messages to be placed in a
single UDP packet.
If a response does not contain a Content-Length, the client assumes
that it encompasses the remainder of the UDP packet or the data until
the TCP connection is closed, as applicable. Section 8 describes how
to determine the length of the message body.
The ability to omit Content-Length simplifies the creation
of cgi-like scripts that dynamically generate responses.
6.20 Content-Type
The Content-Type entity-header field indicates the media type of the
message-body sent to the recipient. The "media-type" element is
defined in [H3.7].
Content-Type = ( "Content-Type" | "c" ) ":" media-type
Examples of this header field are
Content-Type: application/sdp
Content-Type: text/html; charset=ISO-8859-4
6.21 CSeq
Clients MUST add the CSeq (command sequence) general-header field to
every request. A CSeq header field in a request contains the request
method and a single decimal sequence number chosen by the requesting
client, unique within a single call leg. The sequence number MUST be
expressible as a 32-bit unsigned integer. The initial value of the
sequence number is arbitrary, but MUST be less than 2**31.
Consecutive requests that differ in request method, headers or body,
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but have the same Call-ID MUST contain strictly monotonically
increasing and contiguous sequence numbers; sequence numbers do not
wrap around. Retransmissions of the same request carry the same
sequence number, but an INVITE with a different message body or
different header fields (a "re-invitation") acquires a new, higher
sequence number. A server MUST echo the CSeq value from the request
in its response. If the Method value is missing in the received CSeq
header field, the server fills it in appropriately.
The ACK and CANCEL requests MUST contain the same CSeq value as the
INVITE request that it refers to, while a BYE request cancelling an
invitation MUST have a higher sequence number. A BYE request with a
CSeq that is not higher should cause a 400 response to be generated.
A user agent server MUST remember the highest sequence number for any
INVITE request with the same Call-ID value. The server MUST respond
to, and then discard, any INVITE request with a lower sequence
number.
All requests spawned in a parallel search have the same CSeq value as
the request triggering the parallel search.
CSeq = "CSeq" ":" 1*DIGIT Method
Strictly speaking, CSeq header fields are needed for any
SIP request that can be cancelled by a BYE or CANCEL
request or where a client can issue several requests for
the same Call-ID in close succession. Without a sequence
number, the response to an INVITE could be mistaken for the
response to the cancellation (BYE or CANCEL). Also, if the
network duplicates packets or if an ACK is delayed until
the server has sent an additional response, the client
could interpret an old response as the response to a re-
invitation issued shortly thereafter. Using CSeq also makes
it easy for the server to distinguish different versions of
an invitation, without comparing the message body.
The Method value allows the client to distinguish the response to an
INVITE request from that of a CANCEL response. CANCEL requests can be
generated by proxies; if they were to increase the sequence number,
it might conflict with a later request issued by the user agent for
the same call.
With a length of 32 bits, a server could generate, within a single
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call, one request a second for about 136 years before needing to wrap
around. The initial value of the sequence number is chosen so that
subsequent requests within the same call will not wrap around. A
non-zero initial value allows to use a time-based initial sequence
number, if the client desires. A client could, for example, choose
the 31 most significant bits of a 32-bit second clock as an initial
sequence number.
Forked requests MUST have the same CSeq as there would be ambiguity
otherwise between these forked requests and later BYE issued by the
client user agent.
Example:
CSeq: 4711 INVITE
6.22 Date
Date is a general-header field. Its syntax is:
Date = "Date" ":" SIP-date
SIP-date = rfc1123-date
See [H14.18] for a definition of rfc1123-date. Note that unlike
HTTP/1.1, SIP only supports the most recent RFC 1123 [36] formatting
for dates. As in [H3.3], SIP restricts the timezone in SIP-date to
"GMT", while RFC 1123 allows any timezone.
The consistent use of GMT between Date, Expires and Retry-
After headers allows implementation of simple clients that
do not have a notion of absolute time. Note that rfc1123-
date is case-sensitive.
The Date header field reflects the time when the request or response
is first sent. Thus, retransmissions have the same Date header field
value as the original.
The Date header field can be used by simple end systems
without a battery-backed clock to acquire a notion of
current time. However, in its GMT-form, it requires clients
to know their offset from GMT.
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6.23 Encryption
The Encryption general-header field specifies that the content has
been encrypted. Section 13 describes the overall SIP security
architecture and algorithms. This header field is intended for end-
to-end encryption of requests and responses. Requests are encrypted
based on the public key belonging to the entity named in the To
header field. Responses are encrypted based on the public key
conveyed in the Response-Key header field. Note that the public keys
themselves may not be used for the encryption. This depends on the
particular algorithms used.
For any encrypted message, at least the message body and possibly
other message header fields are encrypted. An application receiving a
request or response containing an Encryption header field decrypts
the body and then concatenates the plaintext to the request line and
headers of the original message. Message headers in the decrypted
part completely replace those with the same field name in the
plaintext part. (Note: If only the body of the message is to be
encrypted, the body has to be prefixed with CRLF to allow proper
concatenation.) Note that the request method and Request-URI cannot
be encrypted.
Encryption only provides privacy; the recipient has no
guarantee that the request or response came from the party
listed in the From message header, only that the sender
used the recipient's public key. However, proxies will not
be able to modify the request or response.
Encryption = "Encryption" ":" encryption-scheme 1*SP
#encryption-params
encryption-scheme = token
encryption-params = generic-param
generic-param = token [ "=" ( token | quoted-string ) ]
The token indicates the form of encryption used; it is
described in section 13.
The example in Figure 9 shows a message encrypted with ASCII-armored
PGP that was generated by applying "pgp -ea" to the payload to be
encrypted.
Figure 9: PGP Encryption Example
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INVITE sip:watson@boston.bell-telephone.com SIP/2.0
Via: SIP/2.0/UDP 169.130.12.5
From:
To: T. A. Watson
Call-ID: 187602141351@worcester.bell-telephone.com
Cseq: 1 INVITE
Content-Length: 829
Encryption: PGP version=2.6.2,encoding=ascii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=bOW+
Since proxies can base their forwarding decision on any combination
of SIP header fields, there is no guarantee that an encrypted request
"hiding" header fields will reach the same destination as an
otherwise identical un-encrypted request.
6.24 Expires
The Expires entity-header field gives the date and time after which
the message content expires.
This header field is currently defined only for the REGISTER and
INVITE methods. For REGISTER, it is a request and response-header
field. In a REGISTER request, the client indicates how long it wishes
the registration to be valid. In the response, the server indicates
the earliest expiration time of all registrations. The server MAY
choose a shorter time interval than that requested by the client, but
SHOULD NOT choose a longer one. If a registration updates an
existing registration, the Expires value of the most recent
registration is used, even if it is shorter than the earlier
registration.
For INVITE requests, it is a request and response-header field. In a
request, the caller can limit the validity of an invitation, for
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example, if a client wants to limit the time duration of a search or
a conference invitation. A user interface MAY take this as a hint to
leave the invitation window on the screen even if the user is not
currently at the workstation. This also limits the duration of a
search. If the request expires before the search completes, the proxy
returns a 408 (Request Timeout) status. In a 302 (Moved Temporarily)
response, a server can advise the client of the maximal duration of
the redirection.
Note that the expiration time does not affect the duration of the
actual session that may result from the invitation. Session
description protocols may offer the ability to express time limits on
the session duration, however.
The value of this field can be either a SIP-date or an integer number
of seconds (in decimal), measured from the receipt of the request.
The latter approach is preferable for short durations, as it does not
depend on clients and servers sharing a synchronized clock.
Implementations MAY treat values larger than 2**32-1 (4294967295 or
136 years) as equivalent to 2**32-1.
Expires = "Expires" ":" ( SIP-date | delta-seconds )
Two examples of its use are
Expires: Thu, 01 Dec 1994 16:00:00 GMT
Expires: 5
6.25 From
Requests and responses MUST contain a From general-header field,
indicating the initiator of the request. (Note that this may be
different from the initiator of the call leg. Requests sent by the
callee to the caller use the callee's address in the From header
field.) The From field MAY contain the "tag" parameter. The server
copies the From header field from the request to the response. The
optional "display-name" is meant to be rendered by a human-user
interface. A system SHOULD use the display name "Anonymous" if the
identity of the client is to remain hidden.
The SIP-URL MUST NOT contain the "transport-param", "maddr-param",
"ttl-param", or "headers" elements. A server that receives a SIP-URL
with these elements ignores them.
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Even if the "display-name" is empty, the "name-addr" form MUST be
used if the "addr-spec" contains a comma, question mark, or
semicolon. Syntax issues are discussed in Section 6.6.
From = ( "From" | "f" ) ":" ( name-addr | addr-spec )
[ ";" tag-param ] *( ";" addr-extension )
tag-param = "tag=" token
addr-extension = generic-param
Examples:
From: "A. G. Bell" ;tag=a48s
From: sip:+12125551212@server.phone2net.com
From: Anonymous
The "tag" MAY appear in the From field of a request. It MUST be
present when it is possible that two instances of a user sharing a
SIP address can make call invitations with the same Call-ID.
The "tag" value MUST be globally unique and cryptographically random
with at least 32 bits of randomness. A single user agent maintains
the same tag at least within the same Call-ID, but it is RECOMMENDED
to maintain the same tag across calls and instances of the UA
application to allow restarting of a user agent.
For the purpose of identifying call legs, two From or To header
fields are equal if and only if:
o The addr-spec component is equal, according to the rules in
Section 2.1.
o Any "tag" and addr-extension parameters are equal. Parameter
names are compared as case-insensitive strings, while the
parameter value is compared as a case-sensitive string. The
tag comparison is only performed if both header fields have a
tag value.
Call-ID, To and From are needed to identify a call leg.
The distinction between call and call leg matters in calls
with multiple responses to a forked request. The format is
similar to the equivalent RFC 822 [28] header, but with a
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URI instead of just an email address.
6.26 Hide
A client uses the Hide request header field to indicate that it wants
the path comprised of the Via header fields (Section 6.47) to be
hidden from subsequent proxies and user agents. It can take two
forms: Hide: route and Hide: hop. Hide header fields are typically
added by the client user agent, but MAY be added by any proxy along
the path.
If a request contains the "Hide: route" header field, all following
proxies SHOULD hide their previous hop. If a request contains the
"Hide: hop" header field, only the next proxy SHOULD hide the
previous hop and then remove the Hide option unless it also wants to
remain anonymous.
A server hides the previous hop by encrypting the "host" (in both
sent-by and via-received), "port", "maddr" parts of the top-most Via
header field with an algorithm of its choice. Servers SHOULD add
additional "salt" to the "host" and "port" information prior to
encryption to prevent malicious downstream proxies from guessing
earlier parts of the path based on seeing identical encrypted Via
headers. Hidden Via fields are marked with the "hidden" Via option,
as described in Section 6.47.
A server that is capable of hiding Via headers MUST attempt to
decrypt all Via headers marked as "hidden" to perform loop detection.
Servers that are not capable of hiding can ignore hidden Via fields
in their loop detection algorithm.
If hidden headers were not marked, a proxy would have to
decrypt all headers to detect loops, just in case one was
encrypted, as the Hide: hop option may have been removed
along the way.
A client requesting "Hide: route" can only rely on keeping the
request path private if it sends the request to a trusted proxy.
Hiding the route of a SIP request is of limited value if the request
results in data packets being exchanged directly between two user
agents.
The use of Hide header fields is discouraged unless path privacy is
truly needed as Hide fields impose extra processing costs proxies.
The encryption of Via header fields is described in more detail in
Section 13.
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The Hide header field has the following syntax:
Hide = "Hide" ":" ( "route" | "hop" | other-hide )
other-hide = token
6.27 In-Reply-To
The In-Reply-To request header field enumerates the call-IDs that
this call references or returns.
This allows automatic call distribution systems to route
return calls to the originator of the first call and allows
callees to filter calls, so that only calls that return
calls they have originated will be accepted. This field is
not a substitute for request authentication.
In-Reply-To = "In-Reply-To" ":" 1# callid
Example:
In-Reply-To: 70710@saturn.bell-tel.com, 17320@saturn.bell-tel.com
6.28 Max-Forwards
The Max-Forwards request-header field may be used with any SIP method
to limit the number of proxies or gateways that can forward the
request to the next downstream server. This can also be useful when
the client is attempting to trace a request chain which appears to be
failing or looping in mid-chain.
Max-Forwards = "Max-Forwards" ":" 1*DIGIT
The Max-Forwards value is a decimal integer indicating the remaining
number of times this request message is allowed to be forwarded.
Each proxy or gateway recipient of a request containing a Max-
Forwards header field MUST check and update its value prior to
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forwarding the request. If the received value is zero (0), the
recipient MUST NOT forward the request and returns 483 (Too many
hops). Instead, a server MAY act as a final recipient for OPTIONS
requests. It is RECOMMENDED that the server include Supported, Server
and Allow header fields in the response.
If the received Max-Forwards value is greater than zero, then the
forwarded message MUST contain an updated Max-Forwards field with a
value decremented by one (1).
Example:
Max-Forwards: 6
6.29 MIME-Version
See [H19.4.1].
6.30 Organization
The Organization general-header field conveys the name of the
organization to which the entity issuing the request or response
belongs. It MAY also be inserted by proxies at the boundary of an
organization.
The field MAY be used by client software to filter calls.
Organization = "Organization" ":" TEXT-UTF8-TRIM
6.31 Priority
The Priority request-header field indicates the urgency of the
request as perceived by the client.
Priority = "Priority" ":" priority-value
priority-value = "emergency" | "urgent" | "normal"
| "non-urgent" | other-priority
other-priority token
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It is RECOMMENDED that the value of "emergency" only be used when
life, limb or property are in imminent danger.
Examples:
Subject: A tornado is heading our way!
Priority: emergency
Subject: Weekend plans
Priority: non-urgent
These are the values of RFC 2076 [37], with the addition of
"emergency".
6.32 Proxy-Authenticate
The Proxy-Authenticate response-header field MUST be included as part
of a 407 (Proxy Authentication Required) response. It may also occur
in a 401 (Unauthorized) response if the request was forked. The
field value consists of a challenge that indicates the authentication
scheme and parameters applicable to the proxy for this Request-URI.
Unlike its usage within HTTP, the Proxy-Authenticate header MUST be
passed upstream in the response to the UAC. In SIP, only UAC's can
authenticate themselves to proxies.
The syntax for this header is defined in [H14.33]. See 14 for further
details on its usage.
A client SHOULD cache the credentials used for a particular proxy
server and realm for the next request to that server. Credentials
are, in general, valid for a specific value of the Request-URI at a
particular proxy server. If a client contacts a proxy server that has
required authentication in the past, but the client does not have
credentials for the particular Request-URI, it MAY attempt to use the
most-recently used credential. The server responds with 401
(Unauthorized) if the client guessed wrong.
This suggested caching behavior is motivated by proxies
restricting phone calls to authenticated users. It seems
likely that in most cases, all destinations require the
same password. Note that end-to-end authentication is
likely to be destination-specific.
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6.33 Proxy-Authorization
The Proxy-Authorization request-header field allows the client to
identify itself (or its user) to a proxy which requires
authentication. The Proxy-Authorization field value consists of
credentials containing the authentication information of the user
agent for the proxy and/or realm of the resource being requested.
Unlike Authorization, the Proxy-Authorization header field applies
only to the next outbound proxy that demanded authentication using
the Proxy- Authenticate field. When multiple proxies are used in a
chain, the Proxy-Authorization header field is consumed by the first
outbound proxy that was expecting to receive credentials. A proxy MAY
relay the credentials from the client request to the next proxy if
that is the mechanism by which the proxies cooperatively authenticate
a given request.
See [H14.34] for a definition of the syntax, and section 14 for a
discussion of its usage.
6.34 Proxy-Require
The Proxy-Require header field is used to indicate proxy-sensitive
features that MUST be supported by the proxy. Any Proxy-Require
header field features that are not supported by the proxy MUST be
negatively acknowledged by the proxy to the client if not supported.
Proxy servers treat this field identically to the Require field.
See Section 6.36 for more details on the mechanics of this message
and a usage example.
6.35 Record-Route
6.35.1 Operation
The Record-Route request and response header field is added to a
request by any proxy that insists on being in the path of subsequent
requests for the same call leg. A proxy SHOULD add it to any request
for robustness, but a request route, once established, persists until
the end of the call leg, regardless of whether the Record-Route
header is present in subsequent requests.
The Record-Route header field contains a globally reachable Request-
URI that identifies the proxy server, including an address parameter
that identifies its location. Each such proxy server adds the
Request-URI of the incoming request to the beginning of the list.
Requests between both user agents involved in the call leg, in either
direction, traverse this route.
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Some proxies, such as those controlling firewalls or in an
automatic call distribution (ACD) system, need to maintain
call state and thus need to receive any BYE, re-INVITE and
ACK packets for the call. Note that proxy servers have to
add Record-Route headers to each request as long as they
want to be "visited" by the next request for the call leg.
Proxies MUST include an maddr parameter in the URI in the Record-
Route header, but MUST NOT include a transport parameter.
If it is important that all requests go to the same host,
server administrators are advised to be careful in
selecting the appropriate name or address to ensure that
name resolution does indeed resolve to the same host. For
example, a domain name having an SRV record may resolve to
a different network addresses on each attempt. Inclusion
of, say, a TCP transport parameter may prevent a UA that
supports only UDP but reached the proxy inserting the
Record-Route via another proxy from reaching this proxy.
The UAS copies the Record-Route request header field unchanged into
the response. (Record-Route is only relevant for 2xx responses and
responses where the server can expect the client to retry for the
same Call-Id, as in 401 (Unauthorized) or 484 (Address Incomplete).)
6.35.2 Construction of Route Header
Once a proxy P inserts a Record-Route header in a request from UA A
to UA B, all subsequent requests from A to B and from B to A visit P.
A UA builds the Route header field for subsequent requests from the
Record-Route header fields received in either a response or a
request.
If a UAC finds a Record-Route header in a response, it copies it into
Route header fields of all subsequent requests within the same call
leg, reversing the order of fields, so that the first entry is the
server closest to the UAC. If the response contained a Contact header
field, the user agent adds its content as the last Route header.
If a UA find a Record-Route header in a request, it copies the
Record-Route maddr parameters only, maintaining their ordering, to
the Route header field of future requests. Since the URIs contained
in the Record-Route header fields are not useful for the reverse
request path, the UA fills all other components of the Route name-
addr value with the originating name-addr value.
The originating name-addr is the name-addr value found in the Contact
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header of the request or the From header field, if there is no
Contact header field.
If the request featured a Contact header field, the Contact header
value is appended to the Route header list.
6.35.3 Request Destination
Unless this would cause a loop, any client, including the UAC, SHOULD
send the next request for this call leg to the first Request-URI in
the Route request header field. A client MAY forward the request to a
designated proxy instead, for example, if it lacks DNS resolution
capability. If a client uses the first Route entry to route the
request, it removes it.
6.35.4 Syntax
The Record-Route header field has the following syntax:
Record-Route = "Record-Route" ":" 1# name-addr [ rr-extension]
rr-extension = token [ "=" ( token | quoted-string ) ]
Proxy servers MUST include their address in a "maddr" URL parameter
to ensure that subsequent requests are guaranteed to reach exactly
the same server.
6.35.5 Example
Example for a request where the proxy servers ieee.org and bell-
telephone.com , in that order, insist on being part of subsequent
request paths:
Record-Route: ,
6.36 Require
The Require general-header field is used by clients to tell user
agent servers about options that the client expects the server to
support in order to properly process the request. If a server does
not understand the option, it MUST respond by returning status code
420 (Bad Extension) and list those options it does not understand in
the Unsupported header.
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Require = "Require" ":" 1#option-tag
Example:
C->S: INVITE sip:watson@bell-telephone.com SIP/2.0
Require: com.example.billing
Payment: sheep_skins, conch_shells
S->C: SIP/2.0 420 Bad Extension
Unsupported: com.example.billing
This is to make sure that the client-server interaction
will proceed without delay when all options are understood
by both sides, and only slow down if options are not
understood (as in the example above). For a well-matched
client-server pair, the interaction proceeds quickly,
saving a round-trip often required by negotiation
mechanisms. In addition, it also removes ambiguity when the
client requires features that the server does not
understand. Some features, such as call handling fields,
are only of interest to end systems.
Proxy and redirect servers MUST ignore features that are not
understood. If a particular extension requires that intermediate
devices support it, the extension MUST be tagged in the Proxy-Require
field as well (see Section 6.34).
6.37 Response-Key
The Response-Key request-header field can be used by a client to
request the key that the called user agent SHOULD use to encrypt the
response with. The syntax is:
Response-Key = "Response-Key" ":" key-scheme 1*SP #key-param
key-scheme = token
key-param = generic-param
The "key-scheme" gives the type of encryption to be used for the
response. Section 13 describes security schemes.
If the client insists that the server return an encrypted response,
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it includes a
Require: org.ietf.sip.encrypt-response
header field in its request. If the server cannot encrypt for
whatever reason, it MUST follow normal Require header field
procedures and return a 420 (Bad Extension) response. If this Require
header field is not present, a server SHOULD still encrypt if it can.
6.38 Retry-After
The Retry-After response-header field can be used with a 503 (Service
Unavailable) response to indicate how long the service is expected to
be unavailable to the requesting client and with a 404 (Not Found),
600 (Busy), or 603 (Decline) response to indicate when the called
party anticipates being available again. The value of this field can
be either an SIP-date or an integer number of seconds (in decimal)
after the time of the response.
A REGISTER request MAY include this header field when deleting
registrations with "Contact: * ;expires: 0". The Retry-After value
then indicates when the user might again be reachable. The registrar
MAY then include this information in responses to future calls.
An optional comment can be used to indicate additional information
about the time of callback. An optional "duration" parameter
indicates how long the called party will be reachable starting at the
initial time of availability. If no duration parameter is given, the
service is assumed to be available indefinitely.
Retry-After = "Retry-After" ":" ( SIP-date | delta-seconds )
[ comment ] [ ";" "duration" "=" delta-seconds ]
Examples of its use are
Retry-After: Mon, 21 Jul 1997 18:48:34 GMT (I'm in a meeting)
Retry-After: Mon, 01 Jan 9999 00:00:00 GMT
(Dear John: Don't call me back, ever)
Retry-After: Fri, 26 Sep 1997 21:00:00 GMT;duration=3600
Retry-After: 120
In the third example, the callee is reachable for one hour starting
at 21:00 GMT. In the last example, the delay is 2 minutes.
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6.39 Route
The Route request-header field determines the route taken by a
request. Each host removes the first entry and then proxies the
request to the host listed in that entry, also using it as the
Request-URI. The operation is described in more detail in Section
6.35.
The Route header field has the following syntax:
Route = "Route" ":" 1# name-addr [ route-extension ]
route-extension = generic-param
6.40 Server
The Server response-header field contains information about the
software used by the user agent server to handle the request. The
syntax for this field is defined in [H14.38].
6.41 Subject
This header field provides a summary or indicates the nature of the
call, allowing call filtering without having to parse the session
description. (Note that the session description does not have to use
the same subject indication as the invitation.)
Subject = ( "Subject" | "s" ) ":" TEXT-UTF8-TRIM
Example:
Subject: Tune in - they are talking about your work!
6.42 Supported
The Supported general-header field enumerates all the capabilities of
the client or server. This header field SHOULD be included in all
requests (except ACK) and in all responses.
Including the header field in all responses greatly
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simplifies the use of extensions for call control in
subsequent transactions with the same server.
Syntax:
Supported = ( "Supported" | "k" ) ":" 1#option-tag
6.43 Timestamp
The Timestamp general-header field describes when the client sent the
request to the server. The value of the timestamp is of significance
only to the client and it MAY use any timescale. The server MUST echo
the exact same value and MAY, if it has accurate information about
this, add a floating point number indicating the number of seconds
that have elapsed since it has received the request. The timestamp
is used by the client to compute the round-trip time to the server so
that it can adjust the timeout value for retransmissions.
Timestamp = "Timestamp" ":" *(DIGIT) [ "." *(DIGIT) ] [ delay ]
delay = *(DIGIT) [ "." *(DIGIT) ]
Note that there MUST NOT be any LWS between a DIGIT and the decimal
point.
6.44 To
The To general-header field specifies recipient of the request.
To = ( "To" | "t" ) ":" ( name-addr | addr-spec )
[ ";" tag-param ] *( ";" addr-extension )
Requests and responses MUST contain a To general-header field,
indicating the desired recipient of the request. The optional
"display-name" is meant to be rendered by a human-user interface. The
UAS or redirect server copies the To header field into its response,
and MUST add a "tag" parameter.
If there was more than one Via header field, the request
was handled by at least one proxy server. Since the
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receiver cannot know whether any of the proxy servers
forked the request, it is safest to assume that they might
have.
The SIP-URL MUST NOT contain the "transport-param", "maddr-param",
"ttl-param", or "headers" elements. A server that receives a SIP-URL
with these elements removes them before further processing.
The "tag" parameter serves as a general mechanism to distinguish
multiple instances of a user identified by a single SIP URL. As
proxies can fork requests, the same request can reach multiple
instances of a user (mobile and home phones, for example). As each
can respond, there needs to be a means to distinguish the responses
from each at the caller. The situation also arises with multicast
requests. The tag in the To header field serves to distinguish
responses at the UAC. It MUST be placed in the To field of the
response by user agent, registrar and redirect servers, but MUST NOT
be inserted into responses forwarded upstream by proxies. However,
responses generated locally by a proxy, and then sent upstream, MUST
contain a tag.
A UAS or redirect server MUST add a "tag" parameter for all final
responses for all transactions within a call leg. All such parameters
have the same value within the same call leg. These servers MAY add
the tag for informational responses during the initial INVITE
transaction, but MUST add a tag to informational responses for all
subsequent transactions.
See Section 6.25 for details of the "tag" parameter. The "tag"
parameter in To headers is ignored when matching responses to
requests that did not contain a "tag" in their To header.
Section 11 describes when the "tag" parameter MUST appear in
subsequent requests. Note that if a request already contained a tag,
this tag MUST be mirrored in the response; a new tag MUST NOT be
inserted.
Section 6.25 describes how To and From header fields are compared for
the purpose of matching requests to call legs.
UAS SHOULD accept requests even if they do not recognize the URI
scheme (e.g., a tel: URI) or if the To header does not address the
user. Only the Request-URI should be used to reject requests.
Even if the "display-name" is empty, the "name-addr" form MUST be
used if the "addr-spec" contains a comma, question mark, or
semicolon. Note that LWS is common, but not mandatory between the
display-name and the "<".
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The following are examples of valid To headers:
To: The Operator ;tag=287447
To: sip:+12125551212@server.phone2net.com
Call-ID, To and From are needed to identify a call leg.
The distinction between call and call leg matters in calls
with multiple responses from a forked request. The "tag" is
added to the To header field in the response to allow
forking of future requests for the same call by proxies,
while addressing only one of the possibly several
responding user agent servers. It also allows several
instances of the callee to send requests that can be
distinguished.
6.45 Unsupported
The Unsupported response-header field lists the features not
supported by the server. See Section 6.36 for a usage example and
motivation.
Syntax:
Unsupported = "Unsupported" ":" 1#option-tag
6.46 User-Agent
The User-Agent general-header field contains information about the
client user agent originating the request. The syntax and semantics
are defined in [H14.42].
6.47 Via
The Via field indicates the path taken by the request so far. This
prevents request looping and ensures replies take the same path as
the requests, which assists in firewall traversal and other unusual
routing situations.
6.47.1 Requests
The client originating the request MUST insert into the request a Via
field containing its host name or network address and, if not the
default port number, the port number at which it wishes to receive
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responses. (Note that this port number can differ from the UDP source
port number of the request.) A fully-qualified domain name is
RECOMMENDED. Each subsequent proxy server that sends the request
onwards MUST add its own additional Via field before any existing Via
fields. A proxy that receives a redirection (3xx) response and then
searches recursively, MUST use the same Via headers as on the
original proxied request.
A proxy SHOULD check the top-most Via header field to ensure that it
contains the sender's correct network address, as seen from that
proxy. If the sender's address is incorrect, the proxy MUST add an
additional "received" attribute, as described 6.47.2.
A host behind a network address translator (NAT) or
firewall may not be able to insert a network address into
the Via header that can be reached by the next hop beyond
the NAT. Use of the received attribute allows SIP requests
to traverse NAT's which only modify the source IP address.
NAT's which modify port numbers, called Network Address
Port Translators (NAPTs) will not properly pass SIP when
transported on UDP, so that an application-layer gateway is
required. When run over TCP, SIP stands a better chance of
traversing NAPTs, since its behavior is similar to HTTP in
this case, albeit using different ports.
A client that sends a request to a multicast address MUST add the
"maddr" parameter to its Via header field, and SHOULD add the "ttl"
parameter. (In that case, the maddr parameter SHOULD contain the
destination multicast address, although under exceptional
circumstances it MAY contain a unicast address.) If a server receives
a request which contained an "maddr" parameter in the topmost Via
field, it SHOULD send the response to the address listed in the
"maddr" parameter.
Loop detection is described in Section 12.3.1.
6.47.2 Receiver-tagged Via Header Fields
Every host that sends or forwards a SIP request adds a Via field
indicating the host's address. However, it is possible that Network
Address Translators (NATs) change the source address and port of the
request (e.g., from a net-10 to a globally routable address), in
which case the Via header field cannot be relied on to route replies.
To prevent this, a proxy SHOULD check the top-most Via header field
to ensure that it contains the sender's correct network address, as
seen from that proxy. If the sender's address is incorrect, the proxy
MUST add a "received" parameter to the Via header field inserted by
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the previous hop. Such a modified Via header field is known as a
receiver-tagged Via header field. If and only if the source address
and port differ from the sent-by address, the proxy also includes the
source port in the received parameter.
An example is:
Via: SIP/2.0/UDP erlang.bell-telephone.com:5060
Via: SIP/2.0/UDP 10.0.0.1:5060 ;received=199.172.136.3
In this example, the message originated from 10.0.0.1 and traversed a
NAT with the external address border.ieee.org (199.172.136.3) to
reach erlang.bell-telephone.com. The latter noticed the mismatch,
and added a parameter to the previous hop's Via header field,
containing the address that the packet actually came from. (Note that
the NAT border.ieee.org is not a SIP server.)
6.47.3 Responses
Via header fields in responses are processed by a proxy or UAC
according to the following rules:
1. The first Via header field should indicate the proxy or
client processing this response. If it does not, discard
the message. Otherwise, remove this Via field.
2. If there is no second Via header field, this response is
destined for this client. Otherwise, the processing depends
on whether the Via field contains a "maddr" parameter or is
a receiver-tagged field:
- If the second Via header field contains a "maddr"
parameter, forward the response to the address listed
there, using the port indicated in "sent-by", or port
5060 if none is present. If the address is a multicast
address, the response SHOULD be sent using the TTL
indicated in the "ttl" parameter, or with a TTL of 1 if
that parameter is not present.
- If the second Via header field does not contain a "maddr"
parameter and is a receiver-tagged field (Section
6.47.2), send the message to the address in the
"received" parameter, using the port indicated in the
"sent-by" value, or using port 5060 if none is present.
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- If neither of the previous cases apply, send the message
to the address indicated by the "sent-by" value in the
second Via header field.
6.47.4 User Agent and Redirect Servers
A UAS or redirect server copies the Via header fields into the
response, without changing their order, and then sends a response
based on one of the following rules:
o If the first Via header field in the request contains a
"maddr" parameter, send the response to the address listed
there, using the port indicated in "sent-by", or port 5060 if
none is present. If the address is a multicast address, the
response SHOULD be sent using the TTL indicated in the "ttl"
parameter, or with a TTL of 1 if that parameter is not
present.
o If the address in the "sent-by" value of the first Via field
differs from the source address of the packet, send the
response to the actual packet source address, similar to the
treatment for receiver-tagged Via header fields (Section
6.47.2). However, the port number is taken from the sent-by
part of the Via header.
This mode of operation supports unaided traversal of
SIP responses through NATs, but does not work through
NAPTs.
o If neither of these conditions is true, send the response to
the address contained in the "sent-by" value. If the request
was sent using TCP, use the existing TCP connection if
available.
6.47.5 Syntax
The format for a Via header field is shown in Fig. 10. The "maddr"
parameter, designating the multicast address, and the "ttl"
parameter, designating the time-to-live (TTL) value, are included
only if the request was sent via multicast. The "received" parameter
is added only for receiver-added Via fields (Section 6.47.2). For
reasons of privacy, a client or proxy may wish to hide its Via
information by encrypting it (see Section 6.26). The "hidden"
parameter is included if this header field was hidden by the upstream
proxy (see 6.26). Note that privacy of the proxy relies on the
cooperation of the next hop, as the next-hop proxy will, by
necessity, know the IP address and port number of the source host.
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Via = ( "Via" | "v") ":" 1#( sent-protocol sent-by
*( ";" via-params ) [ comment ] )
via-params = via-hidden | via-ttl | via-maddr
| via-received | via-branch | via-extension
via-hidden = "hidden"
via-ttl = "ttl" "=" ttl
via-maddr = "maddr" "=" maddr
via-received = "received" "=" host [ ":" port ]
via-branch = "branch" "=" token
via-extension = token [ "=" ( token | quoted-string ) ]
sent-protocol = protocol-name "/" protocol-version "/" transport
protocol-name = "SIP" | token
protocol-version = token
transport = "UDP" | "TCP" | token
sent-by = ( host [ ":" port ] ) | ( concealed-host )
concealed-host = token
ttl = 1*3DIGIT ; 0 to 255
Figure 10: Syntax of Via header field
The "branch" parameter is included by every proxy. The token MUST be
unique for each distinct request. The precise format of the token is
implementation-defined. In order to be able to both detect loops and
associate responses with the corresponding request, the parameter
SHOULD consist of two parts separable by the implementation. One
part, used for loop detection (Section 12.3.1), MAY be computed as a
cryptographic hash of the To, From, Call-ID header fields, the
Request-URI of the request received (before translation) and the
sequence number from the CSeq header field. The algorithm used to
compute the hash is implementation-dependent, but MD5 [38], expressed
in hexadecimal, is a reasonable choice. (Note that base64 is not
permissible for a token.) The other part, used for matching responses
to requests, is a function of the branch taken, for example, a
sequence number or the MD5 hash of the request-URI of the request
sent on the branch.
For example: 7a83e5750418bce23d5106b4c06cc632.1
The "branch" parameter MUST depend on the incoming
request-URI to distinguish looped requests from requests
whose request-URI is changed and which then reach a server
visited earlier.
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CANCEL requests MUST have the same branch value as the corresponding
forked request. When a response arrives at the proxy it can use the
branch value to figure out which branch the response corresponds to.
Via: SIP/2.0/UDP first.example.com:4000;ttl=16
;maddr=224.2.0.1 ;branch=a7c6a8dlze.1 (Acme server)
Via: SIP/2.0/UDP adk8
6.48 Warning
The Warning response-header field is used to carry additional
information about the status of a response. Warning headers are sent
with responses and have the following format:
Warning = "Warning" ":" 1#warning-value
warning-value = warn-code SP warn-agent SP warn-text
warn-code = 3DIGIT
warn-agent = ( host [ ":" port ] ) | pseudonym
; the name or pseudonym of the server adding
; the Warning header, for use in debugging
warn-text = quoted-string
A response MAY carry more than one Warning header.
The "warn-text" should be in a natural language that is most likely
to be intelligible to the human user receiving the response. This
decision can be based on any available knowledge, such as the
location of the cache or user, the Accept-Language field in a
request, or the Content-Language field in a response. The default
language is i-default [39].
Any server MAY add Warning headers to a response. Proxy servers MUST
place additional Warning headers before any Authorization headers.
Within that constraint, Warning headers MUST be added after any
existing Warning headers not covered by a signature. A proxy server
MUST NOT delete any Warning header field that it received with a
response.
When multiple Warning headers are attached to a response, the user
agent SHOULD display as many of them as possible, in the order that
they appear in the response. If it is not possible to display all of
the warnings, the user agent first displays warnings that appear
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early in the response.
The warn-code consists of three digits. A first digit of "3"
indicates warnings specific to SIP.
This is a list of the currently-defined "warn-code"s, each with a
recommended warn-text in English, and a description of its meaning.
Note that these warnings describe failures induced by the session
description.
Warnings 300 through 329 are reserved for indicating problems with
keywords in the session description, 330 through 339 are warnings
related to basic network services requested in the session
description, 370 through 379 are warnings related to quantitative QoS
parameters requested in the session description, and 390 through 399
are miscellaneous warnings that do not fall into one of the above
categories.
300 Incompatible network protocol: One or more network protocols
contained in the session description are not available.
301 Incompatible network address formats: One or more network
address formats contained in the session description are
not available.
302 Incompatible transport protocol: One or more transport
protocols described in the session description are not
available.
303 Incompatible bandwidth units: One or more bandwidth
measurement units contained in the session description were
not understood.
304 Media type not available: One or more media types contained
in the session description are not available.
305 Incompatible media format: One or more media formats
contained in the session description are not available.
306 Attribute not understood: One or more of the media
attributes in the session description are not supported.
307 Session description parameter not understood: A parameter
other than those listed above was not understood.
330 Multicast not available: The site where the user is located
does not support multicast.
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331 Unicast not available: The site where the user is located
does not support unicast communication (usually due to the
presence of a firewall).
370 Insufficient bandwidth: The bandwidth specified in the
session description or defined by the media exceeds that
known to be available.
399 Miscellaneous warning: The warning text can include
arbitrary information to be presented to a human user, or
logged. A system receiving this warning MUST NOT take any
automated action.
1xx and 2xx have been taken by HTTP/1.1.
Additional "warn-code"s, as in the example below, can be defined
through IANA.
Examples:
Warning: 307 isi.edu "Session parameter 'foo' not understood"
Warning: 301 isi.edu "Incompatible network address type 'E.164'"
6.49 WWW-Authenticate
The WWW-Authenticate response-header field MUST be included in 401
(Unauthorized) response messages. The field value consists of at
least one challenge that indicates the authentication scheme(s) and
parameters applicable to the Request-URI. See [H14.46] for a
definition of the syntax, and section 14 for an overview of usage.
The content of the "realm" parameter SHOULD be displayed to the user.
A user agent SHOULD cache the authorization credentials for a given
value of the destination (To header) and "realm" and attempt to re-
use these values on the next request for that destination.
In addition to the "basic" and "digest" authentication schemes
defined in the specifications cited above, SIP defines a new scheme,
PGP (RFC 2015, [40]), Section 15. Other schemes, such as S/MIME, are
for further study.
7 Status Code Definitions
The response codes are consistent with, and extend, HTTP/1.1 response
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codes. Not all HTTP/1.1 response codes are appropriate, and only
those that are appropriate are given here. Other HTTP/1.1 response
codes SHOULD NOT be used. Response codes not defined by HTTP/1.1 have
codes x80 upwards to avoid clashes with future HTTP response codes.
Also, SIP defines a new class, 6xx. The default behavior for unknown
response codes is given for each category of codes.
7.1 Informational 1xx
Informational responses indicate that the server or proxy contacted
is performing some further action and does not yet have a definitive
response. The client SHOULD wait for a further response from the
server, and the server SHOULD send such a response without further
prompting. A server SHOULD send a 1xx response if it expects to take
more than 200 ms to obtain a final response. A server MAY issue zero
or more 1xx responses, with no restriction on their ordering or
uniqueness. Note that 1xx responses are not transmitted reliably,
that is, they do not cause the client to send an ACK. Servers are
free to retransmit informational responses and clients can inquire
about the current state of call processing by re-sending the request.
7.1.1 100 Trying
Some unspecified action is being taken on behalf of this call (e.g.,
a database is being consulted), but the user has not yet been
located.
7.1.2 180 Ringing
The called user agent has located a possible location where the user
has registered recently and is trying to alert the user.
7.1.3 181 Call Is Being Forwarded
A proxy server MAY use this status code to indicate that the call is
being forwarded to a different set of destinations.
7.1.4 182 Queued
The called party is temporarily unavailable, but the callee has
decided to queue the call rather than reject it. When the callee
becomes available, it will return the appropriate final status
response. The reason phrase MAY give further details about the status
of the call, e.g., "5 calls queued; expected waiting time is 15
minutes". The server MAY issue several 182 responses to update the
caller about the status of the queued call.
7.1.5 183 Session Progress
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The 183 (Session Progress) response is used to convey information
about the progress of the call which is not otherwise classified. The
Reason-Phrase MAY be used to convey more details about the call
progress. The status response MAY contain a session description that
allows the UAC to receive special announcements or progress tones.
7.2 Successful 2xx
The request was successful and MUST terminate a search.
7.2.1 200 OK
The request has succeeded. The information returned with the response
depends on the method used in the request, for example:
BYE: The call has been terminated. The message body is empty.
CANCEL: The search has been cancelled. The message body is
empty.
INVITE: The callee has agreed to participate; the message body
indicates the callee's capabilities.
OPTIONS: The callee has agreed to share its capabilities,
included in the message body.
REGISTER: The registration has succeeded. The client treats the
message body according to its Content-Type.
7.3 Redirection 3xx
3xx responses give information about the user's new location, or
about alternative services that might be able to satisfy the call.
They SHOULD terminate an existing search, and MAY cause the initiator
to begin a new search if appropriate.
To avoid forwarding loops, a user agent client or proxy MUST check
whether the address returned by a redirect server equals an address
tried earlier.
7.3.1 300 Multiple Choices
The address in the request resolved to several choices, each with its
own specific location, and the user (or user agent) can select a
preferred communication end point and redirect its request to that
location.
The response SHOULD include an entity containing a list of resource
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characteristics and location(s) from which the user or user agent can
choose the one most appropriate, if allowed by the Accept request
header. The entity format is specified by the media type given in the
Content-Type header field. The choices SHOULD also be listed as
Contact fields (Section 6.15). Unlike HTTP, the SIP response MAY
contain several Contact fields or a list of addresses in a Contact
field. User agents MAY use the Contact header field value for
automatic redirection or MAY ask the user to confirm a choice.
However, this specification does not define any standard for such
automatic selection.
This status response is appropriate if the callee can be
reached at several different locations and the server
cannot or prefers not to proxy the request.
7.3.2 301 Moved Permanently
The user can no longer be found at the address in the Request-URI and
the requesting client SHOULD retry at the new address given by the
Contact header field (Section 6.15). The caller SHOULD update any
local directories, address books and user location caches with this
new value and redirect future requests to the address(es) listed.
7.3.3 302 Moved Temporarily
The requesting client SHOULD retry the request at the new address(es)
given by the Contact header field (Section 6.15). The Request-URI of
the new request uses the value of the Contact header in the response.
The new request can take two different forms. In the first approach,
the To, From, Call-ID, and CSeq header fields in the new request are
the same as in the original request, with a new branch identifier in
the Via header field. Proxies MUST follow this behavior and UACs MAY.
UAs MAY also use the Contact information for the To header field, as
well as a new Call-ID value.
Reusing the CSeq value allows proxies to avoid forwarding
the request to the same destination twice, as a proxy will
consider it a retransmission.
The duration of the redirection can be indicated through an Expires
(Section 6.24) header. If there is no explicit expiration time, the
address is only valid for this call and MUST NOT be cached for future
calls.
7.3.4 305 Use Proxy
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The requested resource MUST be accessed through the proxy given by
the Contact field. The Contact field gives the URI of the proxy. The
recipient is expected to repeat this single request via the proxy.
305 responses MUST only be generated by user agent servers.
7.3.5 380 Alternative Service
The call was not successful, but alternative services are possible.
The alternative services are described in the message body of the
response. Formats for such bodies are not defined here, and may be
the subject of future standardization.
7.4 Request Failure 4xx
4xx responses are definite failure responses from a particular
server. The client SHOULD NOT retry the same request without
modification (e.g., adding appropriate authorization). However, the
same request to a different server might be successful.
7.4.1 400 Bad Request
The request could not be understood due to malformed syntax. The
Reason-Phrase SHOULD identify the syntax problem in more detail,
e.g., "Missing Content-Length header".
7.4.2 401 Unauthorized
The request requires user authentication. This response is issued by
user agent servers and registrars, while 407 (Proxy Authentication
Required) is used by proxy servers.
7.4.3 402 Payment Required
Reserved for future use.
7.4.4 403 Forbidden
The server understood the request, but is refusing to fulfill it.
Authorization will not help, and the request SHOULD NOT be repeated.
7.4.5 404 Not Found
The server has definitive information that the user does not exist at
the domain specified in the Request-URI. This status is also returned
if the domain in the Request-URI does not match any of the domains
handled by the recipient of the request.
7.4.6 405 Method Not Allowed
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The method specified in the Request-Line is not allowed for the
address identified by the Request-URI. The response MUST include an
Allow header field containing a list of valid methods for the
indicated address.
7.4.7 406 Not Acceptable
The resource identified by the request is only capable of generating
response entities which have content characteristics not acceptable
according to the accept headers sent in the request.
7.4.8 407 Proxy Authentication Required
This code is similar to 401 (Unauthorized), but indicates that the
client MUST first authenticate itself with the proxy. The proxy MUST
return a Proxy-Authenticate header field (section 6.32) containing a
challenge applicable to the proxy for the requested resource. The
client MAY repeat the request with a suitable Proxy-Authorization
header field (section 6.33). SIP access authentication is explained
in section 13.2 and 14.
This status code is used for applications where access to the
communication channel (e.g., a telephony gateway) rather than the
callee requires authentication.
7.4.9 408 Request Timeout
The server could not produce a response within a suitable amount of
time, for example, since it could not determine the location of the
user in time. The amount of time may have been indicated in the
Expires request-header field or may be set by the server. The client
MAY repeat the request without modifications at any later time.
7.4.10 409 Conflict
The request could not be completed due to a conflict with the current
state of the resource. This response is returned if the action
parameter in a REGISTER request conflicts with existing
registrations.
7.4.11 410 Gone
The requested resource is no longer available at the server and no
forwarding address is known. This condition is expected to be
considered permanent. If the server does not know, or has no facility
to determine, whether or not the condition is permanent, the status
code 404 (Not Found) SHOULD be used instead.
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7.4.12 411 Length Required
The server refuses to accept the request without a defined Content-
Length. The client MAY repeat the request if it adds a valid
Content-Length header field containing the length of the message-body
in the request message.
7.4.13 413 Request Entity Too Large
The server is refusing to process a request because the request
entity is larger than the server is willing or able to process. The
server MAY close the connection to prevent the client from continuing
the request.
If the condition is temporary, the server SHOULD include a Retry-
After header field to indicate that it is temporary and after what
time the client MAY try again.
7.4.14 414 Request-URI Too Long
The server is refusing to service the request because the Request-URI
is longer than the server is willing to interpret.
7.4.15 415 Unsupported Media Type
The server is refusing to service the request because the message
body of the request is in a format not supported by the server for
the requested method. The server SHOULD return a list of acceptable
formats using the Accept, Accept-Encoding and Accept-Language header
fields.
7.4.16 420 Bad Extension
The server did not understand the protocol extension specified in a
Proxy-Require (Section 6.34) or Require (Section 6.36) header field.
7.4.17 480 Temporarily Unavailable
The callee's end system was contacted successfully but the callee is
currently unavailable (e.g., not logged in, logged in in such a
manner as to preclude communication with the callee or activated the
"do not disturb" feature). The response MAY indicate a better time to
call in the Retry-After header. The user could also be available
elsewhere (unbeknownst to this host), thus, this response does not
terminate any searches. The reason phrase SHOULD indicate a more
precise cause as to why the callee is unavailable. This value SHOULD
be setable by the user agent. Status 486 (Busy Here) MAY be used to
more precisely indicate a particular reason for the call failure.
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This status is also returned by a redirect server that recognizes the
user identified by the Request-URI, but does not currently have a
valid forwarding location for that user.
7.4.18 481 Call Leg/Transaction Does Not Exist
This status is returned under three conditions: The server received a
BYE request that does not match any existing call leg, the server
received a CANCEL request that does not match any existing
transaction or the server received an INVITE with a To tag that does
not match the local tag value. (A server simply discards an ACK
referring to an unknown transaction.)
7.4.19 482 Loop Detected
The server received a request with a Via (Section 6.47) path
containing itself.
7.4.20 483 Too Many Hops
The server received a request that contains a Max-Forwards (Section
6.28) header with the value zero.
7.4.21 484 Address Incomplete
The server received a request with a To (Section 6.44) address or
Request-URI that was incomplete. Additional information SHOULD be
provided.
This status code allows overlapped dialing. With overlapped
dialing, the client does not know the length of the dialing
string. It sends strings of increasing lengths, prompting
the user for more input, until it no longer receives a 484
status response.
7.4.22 485 Ambiguous
The callee address provided in the request was ambiguous. The
response MAY contain a listing of possible unambiguous addresses in
Contact headers.
Revealing alternatives can infringe on privacy concerns of the user
or the organization. It MUST be possible to configure a server to
respond with status 404 (Not Found) or to suppress the listing of
possible choices if the request address was ambiguous.
Example response to a request with the URL lee@example.com :
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485 Ambiguous SIP/2.0
Contact: Carol Lee
Contact: Ping Lee
Contact: Lee M. Foote
Some email and voice mail systems provide this
functionality. A status code separate from 3xx is used
since the semantics are different: for 300, it is assumed
that the same person or service will be reached by the
choices provided. While an automated choice or sequential
search makes sense for a 3xx response, user intervention is
required for a 485 response.
7.4.23 486 Busy Here
The callee's end system was contacted successfully but the callee is
currently not willing or able to take additional calls at this end
system. The response MAY indicate a better time to call in the
Retry-After header. The user could also be available elsewhere, such
as through a voice mail service, thus, this response does not
terminate any searches. Status 600 (Busy Everywhere) SHOULD be used
if the client knows that no other end system will be able to accept
this call.
7.4.24 487 Request Terminated
The original request was terminated by a BYE or CANCEL request.
7.4.25 488 Not Acceptable Here
The response has the same meaning as 606 (Not Acceptable), but only
applies to the specific entity addressed by the Request-URI and the
request may succeed elsewhere.
7.5 Server Failure 5xx
5xx responses are failure responses given when a server itself has
erred. They are not definitive failures, and MUST NOT terminate a
search if other possible locations remain untried.
7.5.1 500 Server Internal Error
The server encountered an unexpected condition that prevented it from
fulfilling the request. The client MAY display the specific error
condition, and MAY retry the request after several seconds.
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If the condition is temporary, the server MAY indicate when the
client may retry the request using the Retry-After header.
7.5.2 501 Not Implemented
The server does not support the functionality required to fulfill the
request. This is the appropriate response when the server does not
recognize the request method and is not capable of supporting it for
any user.
7.5.3 502 Bad Gateway
The server, while acting as a gateway or proxy, received an invalid
response from the downstream server it accessed in attempting to
fulfill the request.
7.5.4 503 Service Unavailable
The server is currently unable to handle the request due to a
temporary overloading or maintenance of the server. The implication
is that this is a temporary condition which will be alleviated after
some delay. If known, the length of the delay MAY be indicated in a
Retry-After header. If no Retry-After is given, the client MUST
handle the response as it would for a 500 response.
Note: The existence of the 503 status code does not imply that a
server has to use it when becoming overloaded. Some servers MAY wish
to simply refuse the connection.
7.5.5 504 Server Time-out
The server did not receive a timely response from the server (e.g., a
location server) it accessed in attempting to process the request.
Note that 408 (Request Timeout) should be used if there was no
response within the period specified in the Expires header field from
the upstream server.
7.5.6 505 Version Not Supported
The server does not support, or refuses to support, the SIP protocol
version that was used in the request message. The server is
indicating that it is unable or unwilling to complete the request
using the same major version as the client, other than with this
error message. The response MAY contain an entity describing why that
version is not supported and what other protocols are supported by
that server. The format for such an entity is not defined here and
may be the subject of future standardization.
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7.6 Global Failures 6xx
6xx responses indicate that a server has definitive information about
a particular user, not just the particular instance indicated in the
Request-URI. All further searches for this user are doomed to failure
and pending searches SHOULD be terminated.
7.6.1 600 Busy Everywhere
The callee's end system was contacted successfully but the callee is
busy and does not wish to take the call at this time. The response
MAY indicate a better time to call in the Retry-After header. If the
callee does not wish to reveal the reason for declining the call, the
callee uses status code 603 (Decline) instead. This status response
is returned only if the client knows that no other end point (such as
a voice mail system) will answer the request. Otherwise, 486 (Busy
Here) should be returned.
7.6.2 603 Decline
The callee's machine was successfully contacted but the user
explicitly does not wish to or cannot participate. The response MAY
indicate a better time to call in the Retry-After header.
7.6.3 604 Does Not Exist Anywhere
The server has authoritative information that the user indicated in
the To request field does not exist anywhere. Searching for the user
elsewhere will not yield any results.
7.6.4 606 Not Acceptable
The user's agent was contacted successfully but some aspects of the
session description such as the requested media, bandwidth, or
addressing style were not acceptable.
A 606 (Not Acceptable) response means that the user wishes to
communicate, but cannot adequately support the session described. The
606 (Not Acceptable) response MAY contain a list of reasons in a
Warning header field describing why the session described cannot be
supported. Reasons are listed in Section 6.48. It is hoped that
negotiation will not frequently be needed, and when a new user is
being invited to join an already existing conference, negotiation may
not be possible. It is up to the invitation initiator to decide
whether or not to act on a 606 (Not Acceptable) response.
8 SIP Message Body
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8.1 Body Inclusion
Requests MAY contain message bodies unless otherwise noted. In this
specification, the CANCEL request MUST NOT contain a message body.
The use of message bodies for REGISTER requests is for further study.
For response messages, the request method and the response status
code determine the type and interpretation of any message body. All
responses MAY include a body. Message bodies for 1xx responses
contain advisory information about the progress of the request. 1xx
responses to INVITE requests MAY contain session descriptions. Their
interpretation depends on the response status code, but generally
informs the caller what kind of session the callee is likely to
establish, subject to later modification in the 2xx response.
Request methods not defined in this specification MAY also contain
session descriptions. 2xx responses to INVITE requests contain
session descriptions. In 3xx responses, the message body MAY contain
the description of alternative destinations or services, as described
in Section 7.3. For responses with status 400 or greater, the message
body MAY contain additional, human-readable information about the
reasons for failure. It is RECOMMENDED that information in 1xx and
300 and greater responses be of type text/plain or text/html
8.2 Message Body Type
The Internet media type of the message body MUST be given by the
Content-Type header field. If the body has undergone any encoding
(such as compression) then this MUST be indicated by the Content-
Encoding header field, otherwise Content-Encoding MUST be omitted. If
applicable, the character set of the message body is indicated as
part of the Content-Type header-field value.
The "multipart" MIME type [41] MAY be used within the body of the
message. Clients that send requests containing multipart message
bodies MUST be able to send a session description as a non-multipart
message body if the server requests this through an Accept header
field.
8.3 Message Body Length
The body length in bytes SHOULD be given by the Content-Length header
field. Section 6.19 describes the behavior in detail.
The "chunked" transfer encoding of HTTP/1.1 MUST NOT be used for SIP.
(Note: The chunked encoding modifies the body of a message in order
to transfer it as a series of chunks, each with its own size
indicator.)
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9 Compact Form
When SIP is carried over UDP with authentication and a complex
session description, it may be possible that the size of a request or
response is larger than the MTU. To address this problem, a more
compact form of SIP is also defined by using abbreviations for the
common header fields listed below:
short field name long field name note
c Content-Type
e Content-Encoding
f From
i Call-ID
k Supported from "know"
m Contact from "moved"
l Content-Length
s Subject
t To
v Via
Thus, the message in section 16.2 could also be written:
INVITE sip:bob@example.com SIP/2.0
v:SIP/2.0/UDP 131.215.131.131;maddr=239.128.16.254;ttl=16
v:SIP/2.0/UDP 216.112.6.38
f:sip:alice@wonderland.com
t:sip:bob@example.com
m:sip:alice@mouse.wonderland.com
i:62729-27@216.112.6.38
c:application/sdp
CSeq: 4711 INVITE
l:187
v=0
o=user1 53655765 2353687637 IN IP4 128.3.4.5
s=Mbone Audio
i=Discussion of Mbone Engineering Issues
e=mbone@somewhere.com
c=IN IP4 224.2.0.1/127
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
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Clients MAY mix short field names and long field names within the
same request. Servers MUST accept both short and long field names for
requests. Proxies MAY change header fields between their long and
short forms, but this MUST NOT be done to fields following an
Authorization header.
10 Behavior of SIP Clients and Servers
10.1 General Remarks
SIP is defined so it can use either UDP (unicast or multicast) or TCP
as a transport protocol; it provides its own reliability mechanism.
10.1.1 Requests
Servers discard isomorphic requests, but first retransmit the
appropriate response. (SIP requests are said to be idempotent , i.e.,
receiving more than one copy of a request does not change the server
state.)
An incoming request is accepted unless the Call-ID value matches an
existing call and the request has a tag value of To header that does
not match the user agent server's tag value. In that case, the
request is refused with a 481 (Call Leg/Transaction Does Not Exist)
response.
If the request is accepted and matches an existing call leg, the
server compares the CSeq header field value. If less than or equal to
the current sequence number, the request is a retransmission.
Otherwise, it is a new request. If the request does not match an
existing call leg, a new call leg is created.
If the request did not contain a To tag value, the server returns a
response containing the same To header field value as in the request
and adds a unique tag.
After receiving a CANCEL request from an upstream client, a stateful
proxy server MAY send a CANCEL on all branches where it has not yet
received a final response.
10.1.2 Responses
A server MAY issue one or more provisional responses at any time
before sending a final response. If a stateful proxy, user agent
server, redirect server or registrar cannot respond to a request with
a final response within 200 ms, it SHOULD issue a provisional (1xx)
response as soon as possible. Stateless proxies MUST NOT issue
provisional responses on their own.
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Responses are mapped to requests by the matching To, From, Call-ID,
CSeq headers and the branch parameter of the first Via header.
Responses terminate request retransmissions even if they have Via
headers that cause them to be delivered to an upstream client.
A stateful proxy may receive a response that it does not have state
for, that is, where it has no a record of an associated request. If
the Via header field indicates that the upstream server used TCP, the
proxy actively opens a TCP connection to that address. Thus, proxies
have to be prepared to receive responses on the incoming side of
passive TCP connections, even though most responses will arrive on
the incoming side of an active connection. (An active connection is a
TCP connection initiated by the proxy, a passive connection is one
accepted by the proxy, but initiated by another entity.)
100 responses SHOULD NOT be forwarded, other 1xx responses MAY be
forwarded, possibly after the server eliminates responses with status
codes that had already been sent earlier. 2xx responses are forwarded
according to the Via header. Once a stateful proxy has received a 2xx
response, it MUST NOT forward non-2xx final responses. Responses
with status 300 and higher are retransmitted by each stateful proxy
or UAS until the next upstream proxy or UAC sends an ACK (see below
for timing details) or CANCEL.
A stateful proxy SHOULD maintain state for at least 32 seconds after
the receipt of the first definitive non-200 response, in order to
handle retransmissions of the response.
The 32 second window is given by the maximum retransmission
duration of 200-class responses using the default timers,
in case the ACK is lost somewhere on the way to the called
user agent or the next stateful proxy.
10.2 Source Addresses, Destination Addresses and Connections
10.2.1 Unicast UDP
Responses are sent according to the rules in Section 6.47.3 or
Section 6.47.4.
Recall that responses are not generated by the next-hop
stateless server, but generated by either a proxy server or
the user agent server. Thus, the stateless proxy can only
use the Via header field to forward the response.
10.2.2 Multicast UDP
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Requests MAY be multicast; multicast requests likely feature a host-
independent Request-URI. This request SHOULD be scoped to ensure it
is not forwarded beyond the boundaries of the administrative scope.
This MAY be done with either TTL or administrative scopes [29],
depending on what is implemented in the network.
A client receiving a multicast query does not have to check whether
the host part of the Request-URI matches its own host or domain name.
If the request was received via multicast, the response MUST be
returned to the address listed in the maddr parameter of the Via
header field. Generally, this will be a multicast address. Such
multicast responses are multicast with the same TTL as the request,
where the TTL is derived from the ttl parameter in the Via header
(Section 6.47).
To avoid response implosion, servers MUST NOT answer multicast
requests with a status code other than 2xx, 401, 407, 484 or 6xx. The
server delays its response by a random interval uniformly distributed
between zero and one second. Servers MAY suppress responses if they
hear a lower-numbered or 6xx response from another group member prior
to sending. Servers do not respond to CANCEL requests received via
multicast to avoid request implosion. A proxy or UAC SHOULD send a
CANCEL on receiving the first 2xx, 401, 407 or 6xx response to a
multicast request.
Server response suppression is a MAY since it requires a
server to violate some basic message processing rules. Lets
say A sends a multicast request, and it is received by B,
C, and D. B sends a 200 response. The topmost Via field in
the response will contain the address of A. C will also
receive this response, and could use it to suppress its own
response. However, C would normally not examine this
response, as the topmost Via is not its own. Normally, a
response received with an incorrect topmost Via MUST be
dropped, but not in this case. To distinguish this packet
from a misrouted or multicast looped packet is fairly
complex, and for this reason the procedure is a MAY. The
CANCEL, instead, provides a simpler and more standard way
to perform response suppression. It is for this reason that
the use of CANCEL here is a SHOULD.
10.3 TCP
A single TCP connection can serve one or more SIP transactions. A
transaction contains zero or more provisional responses followed by
one or more final responses. (Typically, transactions contain exactly
one final response, but there are exceptional circumstances, where,
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for example, multiple 200 responses can be generated.)
The client SHOULD keep the connection open at least until the first
final response arrives. If the client closes or resets the TCP
connection prior to receiving the first final response, the server
treats this action as equivalent to a CANCEL request.
This behavior makes it less likely that malfunctioning
clients cause a proxy server to keep connection state
indefinitely.
The server SHOULD NOT close the TCP connection until it has sent its
final response, at which point it MAY close the TCP connection if it
wishes to. However, normally it is the client's responsibility to
close the connection. If the server closes the connection
prematurely, the client SHOULD interpret this as being equivalent to
a 500 (Server Internal Error) response.
If the server leaves the connection open, and if the client so
desires it MAY re-use the connection for further SIP requests or for
requests from the same family of protocols (such as HTTP or stream
control commands).
If a server needs to return a response to a client and no longer has
a connection open to that client, it MAY open a connection to the
address listed in the Via header. Thus, a proxy or user agent MUST be
prepared to receive both requests and responses on a "passive"
connection.
10.4 Reliability for Requests Other Than INVITE
10.4.1 UDP
A SIP client using UDP SHOULD retransmit requests other than INVITE
or ACK with an exponential backoff, starting at a T1 second interval,
doubling the interval for each packet, and capping off at a T2 second
interval. This means that after the first packet is sent, the second
is sent T1 seconds later, the next 2*T1 seconds after that, the next
4*T1 seconds after that, and so on, until the interval reaches T2.
Subsequent retransmissions are spaced by T2 seconds. If the client
receives a provisional response, it continues to retransmit the
request, but with an interval of T2 seconds. Retransmissions cease
when the client has sent a total of eleven packets, or receives a
definitive response. Default values for T1 and T2 are 500 ms and 4 s,
respectively. Clients MAY use larger values, but SHOULD NOT use
smaller ones. Servers retransmit the response upon receipt of a
request retransmission. After the server sends a final response, it
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cannot be sure the client has received the response, and thus SHOULD
cache the results for at least 10*T2 seconds to avoid having to, for
example, contact the user or location server again upon receiving a
request retransmission.
Use of the exponential backoff is for congestion control
purposes. However, the back-off must cap off, since request
retransmissions are used to trigger response
retransmissions at the server. Without a cap, the loss of a
single response could significantly increase transaction
latencies.
The value of the initial retransmission timer is smaller than that
that for TCP since it is expected that network paths suitable for
interactive communications have round-trip times smaller than 500 ms.
For congestion control purposes, the retransmission count has to be
bounded. Given that most transactions are expected to consist of one
request and a few responses, round-trip time estimation is not likely
to be very useful. If RTT estimation is desired to more quickly
discover a missing final response, each request retransmission needs
to be labeled with its own Timestamp (Section 6.43), returned in the
response. The server caches the result until it can be sure that the
client will not retransmit the same request again.
Each server in a proxy chain generates its own final response to a
CANCEL request. The server responds immediately upon receipt of the
CANCEL request rather than waiting until it has received final
responses from the CANCEL requests it generates.
BYE and OPTIONS final responses are generated by redirect and user
agent servers; REGISTER final responses are generated by registrars.
Note that in contrast to the reliability mechanism described in
Section 10.5, responses to these requests are not retransmitted
periodically and not acknowledged via ACK.
10.4.2 TCP
Clients using TCP do not need to retransmit requests, but MAY give up
after receiving no response for an extended period of time.
10.5 Reliability for INVITE Requests
Special considerations apply for the INVITE method.
1. After receiving an invitation, considerable time can elapse
before the server can determine the outcome. For example,
if the called party is "rung" or extensive searches are
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performed, delays between the request and a definitive
response can reach several tens of seconds. If either
caller or callee are automated servers not directly
controlled by a human being, a call attempt could be
unbounded in time.
2. If a telephony user interface is modeled or if we need to
interface to the PSTN, the caller's user interface will
provide "ringback", a signal that the callee is being
alerted. (The status response 180 (Ringing) MAY be used to
initiate ringback.) Once the callee picks up, the caller
needs to know so that it can enable the voice path and stop
ringback. The callee's response to the invitation could get
lost. Unless the response is transmitted reliably, the
caller will continue to hear ringback while the callee
assumes that the call exists.
3. The client has to be able to terminate an on-going request,
e.g., because it is no longer willing to wait for the
connection or search to succeed. The server will have to
wait several retransmission intervals to interpret the lack
of request retransmissions as the end of a call. If the
call succeeds shortly after the caller has given up, the
callee will "pick up the phone" and not be "connected".
10.5.1 UDP
For UDP, A SIP client SHOULD retransmit a SIP INVITE request with an
interval that starts at T1 seconds, and doubles after each packet
transmission. The client ceases retransmissions if it receives a
provisional or definitive response, or once it has sent a total of
seven request packets. A UAC MAY send a BYE or CANCEL request after
the seventh retransmission. It is RECOMMENDED to send both. (This
avoids call establishment in case the network path loses packets
asymmetrically.)
A server which transmits a provisional response should retransmit it
upon reception of a duplicate request. A server which transmits a
final response should retransmit it with an interval that starts at
T1 seconds, and doubles for each subsequent packet until it reaches
T2 seconds. Response retransmissions cease when an ACK request is
received or the response has been retransmitted seven times. The
value of a final response is not changed by the arrival of a BYE or
CANCEL request.
Only the user agent client generates an ACK for 2xx final responses,
If the response contained a Contact header field, the ACK MAY be sent
to the address listed in that Contact header field. If the response
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did not contain a Contact header, the client uses the same To header
field and Request-URI as for the INVITE request and sends the ACK to
the same destination as the original INVITE request. ACKs for final
responses other than 2xx are sent to the same server that the
original request was sent to, using the same Request-URI as the
original request. Note, however, that the To header field in the ACK
is copied from the response being acknowledged, not the request, and
thus MAY additionally contain the tag parameter. Also note than
unlike 2xx final responses, a proxy generates an ACK for non-2xx
final responses.
The ACK request MUST NOT be acknowledged to prevent a response-ACK
feedback loop. Fig. 11 and 12 show the client and server state
diagram for INVITE transactions. The "terminated" event occurs if the
server receives either a CANCEL or BYE request. Note that the state
diagram only shows the behavior for the INVITE transaction; the
responses for BYE and CANCEL are not shown and follow the rules laid
in Section 10.4.
The mechanism in Sec. 10.4 would not work well for INVITE
because of the long delays between INVITE and a final
response. If the 200 response were to get lost, the callee
would believe the call to exist, but the voice path would
be dead since the caller does not know that the callee has
picked up. Thus, the INVITE retransmission interval would
have to be on the order of a second or two to limit the
duration of this state confusion. Retransmitting the
response with an exponential back-off helps ensure that the
response is received, without placing an undue burden on
the network.
10.5.2 TCP
A user agent using TCP MUST NOT retransmit requests, but uses the
same algorithm as for UDP (Section 10.5.1) to retransmit responses
until it receives an ACK. A client MAY give up on the request if
there is no response within a client-defined timeout interval.
It is necessary to retransmit 2xx responses as their
reliability is assured end-to-end only. If the chain of
proxies has a UDP link in the middle, it could lose the
response, with no possibility of recovery. For simplicity,
we also retransmit non-2xx responses, although that is not
strictly necessary.
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+===========+
* *
* Initial *
* *
+===========+
|
| -
| INVITE
|
v
*************
T1*2^n <--* *
INVITE -->* Calling *--------+
* * |
************* |
: | |
..............: | 1xx xxx |
: 7 INVITE sent | - ACK |
: | |
: v |
: ************* |
: * Call * |
: * proceeding*<->1xx |
: * * |
: ************* |
: | |
: |<-------------+
:.............. |
. v
*************
xxx <--* *
ACK -->* Completed *
* *
*************
event (xxx=status)
message
Figure 11: State transition diagram of client for INVITE method
10.6 Reliability for ACK Requests
The ACK request does not generate responses. It is only generated
when a response to an INVITE request arrives (see Section 10.5). This
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+===============+
* *
* Initial *
* *
+===============+
!
! INVITE/1xx
!
!
v
*****************
|------->* *<----------|
INVITE/1xx * Proceeding * status change/1xx
^--------* *-----------^
;;;;;;;;;;;;;;;;;;;;;;*****************
; terminated/487 ! !
: ! !
; ! !
; failure/>=300 ! ! picks up/2xx
; +-------------+ +-----------+
; v v
; *********** ***********
;INVITE/>* *<-min(T1*2^n,T2)/->* *<-----|
;status <* failure *-> status <-* success * INVITE/2xx
; * * * *------^
;;;;;;;;;*********** ***********
! !
! !
! !
! !
+--------------+--------------+
event/message sent ! ACK/-
v
*****************
|--->* *
ACK/- * Confirmed *
^--->* *
*****************
Figure 12: State transition diagram of server for INVITE method
behavior is independent of the transport protocol. Note that the ACK
request MAY take a different path than the original INVITE request,
and MAY even cause a new TCP connection to be opened in order to send
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it.
10.7 ICMP Handling
Handling of ICMP messages in the case of UDP messages is
straightforward. For requests, a host, network, port, or protocol
unreachable error SHOULD be treated as if a 400-class response was
received. For responses, these errors SHOULD cause the server to
cease retransmitting the response.
Source quench ICMP messages SHOULD be ignored. TTL exceeded errors
SHOULD be ignored. Parameter problem errors SHOULD be treated as if a
400-class response was received.
11 Behavior of SIP User Agents
This section describes the rules for user agent client and servers
for generating and processing requests and responses.
11.1 Caller Issues Initial INVITE Request
When a user agent client desires to initiate a call, it formulates an
INVITE request. The To field in the request contains the address of
the callee, and remains unaltered as the request traverses proxies.
The Request-URI contains the same address, but may be rewritten by
proxies. The From field contains the address of the caller. If the
From address can appear in requests generated by other user agent
clients for the same call, the caller MUST insert the tag parameter
in the From field. A UAC MAY optionally add a Contact header
containing an address where it would like to be contacted for
transactions from the callee back to the caller.
11.2 Callee Issues Response
When the initial INVITE request is received at the callee, the callee
can accept, redirect, or reject the call. In all of these cases, it
formulates a response. The response MUST copy the To, From, Call-ID,
CSeq and Via fields from the request. Additionally, the responding
UAS MUST add the tag parameter to the To field in the response if the
request contained more than one Via header field. Since a request
from a UAC may fork and arrive at multiple hosts, the tag parameter
serves to distinguish, at the UAC, multiple responses from different
UAS's. The UAS MAY add a Contact header field in the response. It
contains an address where the callee would like to be contacted for
subsequent transactions, including the ACK for the current INVITE.
The UAS stores the values of the To and From field, including any
tags. These become the local and remote addresses of the call leg,
respectively.
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11.3 Caller Receives Response to Initial Request
Multiple responses may arrive at the UAC for a single INVITE request,
due to a forking proxy. Each response is distinguished by the "tag"
parameter in the To header field, and each represents a distinct call
leg. The caller MAY choose to acknowledge or terminate the call with
each responding UAS. To acknowledge, it sends an ACK request, and to
terminate it sends a BYE request. The To header field in the ACK or
BYE MUST be the same as the To field in the 200 response, including
any tag. The From header field MUST be the same as the From header
field in the 200 (OK) response, including any tag. The Request-URI of
the ACK or BYE request MAY be set to whatever address was found in
the Contact header field in the 200 (OK) response, if present.
Alternately, a UAC may copy the address from the To header field into
the Request-URI. The UAC also notes the value of the To and From
header fields in each response. For each call leg, the To header
field becomes the remote address, and the From header field becomes
the local address.
11.4 Caller or Callee Generate Subsequent Requests
Once the call has been established, either the caller or callee MAY
generate INVITE or BYE requests to change or terminate the call.
Regardless of whether the caller or callee is generating the new
request, the header fields in the request are set as follows. For the
desired call leg, the To header field is set to the remote address,
and the From header field is set to the local address (both including
any tags). A UAC copies the tag from the final response into the ACK,
but it MUST NOT copy the tag into any subsequent requests unless the
response was a 200-class response to an INVITE request. The To field
of CANCEL requests always contain exactly the same value as the
request it is cancelling.
The Contact header field MAY be different than the Contact header
field sent in a previous response or request. The Request-URI MAY be
set to the value of the Contact header field received in a previous
request or response from the remote party.
The network destination and Request-URI of requests is determined
according to the following rules:
o If the response from the previous request contained a Record-
Route header field, the UAC sends the request to the last
entry in the list and removes that entry. As described in
Section 6.35, the Request-URI is set to that value.
o Otherwise, if the response from the previous response
contained a Contact header field, the request is directed to
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the host and port identified there. The Request-URI is set to
the value of the Contact header. The request does not contain
a Route header field in this case.
o Otherwise, if the UAC is configured with the address of an
outbound proxy server, the UAC sends the request there. The
Request-URI contains the same URL as the To header field if
the UAC is the callee. It copies the URL and To header field
from the caller's From header if the UAC is the callee.
o Otherwise, if the UAC is the caller, it copies the To header
field into the Request-URI If the UAC is the callee, it copies
the From value of the caller's request into the To header
field and the Request-URI. In both cases, the UAC sends the
request to the host identified in the URL.
If a UAC does not support DNS resolution or the full Record-
Route/Route mechanism, it MAY send all requests to a locally
configured outbound proxy. In that case, that proxy behaves as
described above. The UAC MUST, however, perform the mapping of
Record-Route to Route header fields and MUST include all Route header
fields, i.e., the UAC does not remove the first Route header field.
11.5 Receiving Subsequent Requests
When a request is received during a call, the following checks are
made:
1. If the Call-ID is new, the request is for a new call,
regardless of the values of the To and From header fields.
It is possible that the To header in an INVITE request has
a tag, but the UAS believes this to be a new call. This
will occur if the UAS crashed and rebooted in the middle of
a call, and the UAC has sent what it believes to be a re-
INVITE. The UAS MAY either accept or reject the request.
Accepting the request provides robustness, so that calls
can persist even through crashes. UAs wishing to support
this capability must choose monotonically increasing CSeq
numbers even across reboots. This is because subsequent
requests from the crashed-and-rebooted UA towards the other
UA need to have a CSeq number higher than previous requests
in that direction.
Note also that the crashed-and-rebooted UA will have lost
any Route headers which would need to be inserted into a
subsequent request. Therefore, it is possible that the
requests may not be properly forwarded by proxies.
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RTP media agents allowing restarts need to be robust
by accepting out-of-range timestamps and sequence
numbers.
2. If the Call-ID exists, the request is for an existing call.
If the To, From, Call-ID, and CSeq values exactly match
(including tags) those of any requests received previously,
the request is a retransmission.
3. If there was no match to the previous step, the To and From
fields are compared against existing call leg local and
remote addresses. If there is a match, and the CSeq in the
request is higher than the last CSeq received on that leg,
the request is a new transaction for an existing call leg.
12 Behavior of SIP Proxy and Redirect Servers
This section describes behavior of SIP redirect and proxy servers in
detail. Proxy servers can "fork" connections, i.e., a single incoming
request spawns several outgoing (client) requests.
12.1 Redirect Server
A redirect server does not issue any SIP requests of its own. After
receiving a request other than CANCEL, the server gathers the list of
alternative locations and returns a final response of class 3xx or it
refuses the request. For well-formed CANCEL requests, it SHOULD
return a 2xx response. This response ends the SIP transaction. The
redirect server maintains transaction state for the whole SIP
transaction. It is up to the client to detect forwarding loops
between redirect servers.
12.2 User Agent Server
User agent servers behave similarly to redirect servers, except that
they also accept requests and can return a response of class 2xx.
12.3 Proxy Server
This section outlines processing rules for proxy servers. A proxy
server can either be stateful or stateless. When stateful, a proxy
remembers the incoming request which generated outgoing requests, and
the outgoing requests. A stateless proxy forgets all information once
an outgoing request is generated. A forking proxy SHOULD be stateful.
Proxies that accept TCP connections MUST be stateful when handling
the TCP connection.
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Otherwise, if the proxy were to lose a request, the TCP
client would never retransmit it.
A stateful proxy SHOULD NOT become stateless until after it sends a
definitive response upstream, and at least 32 seconds after it
received a definitive response.
A stateful proxy acts similar to a virtual UAS/UAC, but cannot be
viewed as just a UAS and UAC glued together at the back. (In
particular, it does not originate requests except ACK and CANCEL.)
It implements the server state machine when receiving requests, and
the client state machine for generating outgoing requests, with the
exception of receiving a 2xx response to an INVITE. Instead of
generating an ACK, the 2xx response is always forwarded upstream
towards the caller. Furthermore, ACK's for 200 responses to INVITE's
are always proxied downstream towards the UAS, as they would be for a
stateless proxy.
A stateless proxy forwards every request it receives downstream, and
every response it receives upstream.
12.3.1 Proxying Requests
A proxy server MUST check for forwarding loops before proxying a
request. A request has been looped if the server finds its own
address in the Via header field and the hash computation over the
fields enumerated in Section 6.47.5 yields the same value as the hash
part of the "branch" parameter in the Via entry containing the proxy
server's address.
A proxy server MUST NOT forward a request to a multicast group which
already appears in any of the Via headers.
The To, From, Call-ID, and Contact tags are copied exactly from the
original request. The proxy SHOULD change the Request-URI to indicate
the server where it intends to send the request.
A proxy server always inserts a Via header field containing its own
address into those requests that are caused by an incoming request.
Each proxy MUST insert a "branch" parameter (Section 6.47).
12.3.2 Proxying Responses
A proxy only processes a response if the topmost Via field matches
one of its addresses. A response with a non-matching top Via field
MUST be dropped.
12.3.3 Stateless Proxy: Proxying Responses
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A stateless proxy removes its own Via field, and checks the address
in the next Via field. In the case of UDP, the response is sent to
the address listed in the "maddr" tag if present, otherwise to the
"received" tag if present, and finally to the address in the "sent-
by" field. A proxy MUST remain stateful when handling requests
received via TCP.
A stateless proxy MUST NOT generate its own provisional responses.
12.3.4 Stateful Proxy: Receiving Requests
When a stateful proxy receives a request, it checks the To, From
(including tags), Call-ID and CSeq against existing request records.
If the tuple exists, the request is a retransmission. The provisional
or final response sent previously is retransmitted, as per the server
state machine. If the tuple does not exist, the request corresponds
to a new transaction, and the request should be proxied.
A stateful proxy server MAY generate its own provisional (1xx)
responses.
12.3.5 Stateful Proxy: Receiving ACKs
When an ACK request is received, it is either processed locally or
proxied. To make this determination, the To, From, CSeq and Call-ID
fields are compared against those in previous requests. If there is
no match, the ACK request is proxied as if it were an INVITE request.
If there is a match, and if the server had ever sent a 200 response
upstream, the ACK is proxied. If the server had never sent any
responses upstream, the ACK is also proxied. If the server had sent a
3xx, 4xx, 5xx or 6xx response, but no 2xx response, the ACK is
processed locally if the tag in the To field of the ACK matches the
tag sent by the proxy in the response.
12.3.6 Stateful Proxy: Receiving Responses
When a proxy server receives a response that has passed the Via
checks, the proxy server checks the To (without the tag), From
(including the tag), Call-ID and CSeq against values seen in previous
requests. If there is no match, the response is forwarded upstream to
the address listed in the Via field. If there is a match, the
"branch" tag in the Via field is examined. If it matches a known
branch identifier, the response is for the given branch, and
processed by the virtual client for the given branch. Otherwise, the
response is dropped.
A stateful proxy should obey the rules in Section 12.4 to determine
if the response should be proxied upstream. If it is to be proxied,
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the same rules for stateless proxies above are followed, with the
following addition for TCP. If a request was received via TCP
(indicated by the protocol in the top Via header), the proxy checks
to see if it has a connection currently open to that address. If so,
the response is sent on that connection. Otherwise, a new TCP
connection is opened to the address and port in the Via field, and
the response is sent there. Note that this implies that a UAC or
proxy MUST be prepared to receive responses on the incoming side of a
TCP connection. Definitive non 200-class responses MUST be
retransmitted by the proxy, even over a TCP connection.
12.3.7 Stateless, Non-Forking Proxy
Proxies in this category issue at most a single unicast request for
each incoming SIP request, that is, they do not "fork" requests.
However, servers MAY choose to always operate in a mode that allows
issuing of several requests, as described in Section 12.4.
The server can forward the request and any responses. It does not
have to maintain any state for the SIP transaction. Reliability is
assured by the next redirect or stateful proxy server in the server
chain.
A proxy server SHOULD cache the result of any address translations
and the response to speed forwarding of retransmissions. After the
cache entry has been expired, the server cannot tell whether an
incoming request is actually a retransmission of an older request.
The server will treat it as a new request and commence another
search.
12.4 Forking Proxy
The server must respond to the request (other than ACK) immediately
with a 100 (Trying) response if it expects to take more than 200 ms
to obtain a final response.
Successful responses to an INVITE request MAY contain a Contact
header field so that the following ACK or BYE bypasses the proxy
search mechanism. If the proxy requires future requests to be routed
through it, it adds a Record-Route header to the request (Section
6.35).
The following C-code describes the behavior of a proxy server issuing
several requests in response to an incoming INVITE request with
method R which is to be proxied to a list of N destination enumerated
in ' address .I expires
The function request(r, a, b) sends a SIP request of type r to
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address a, with branch id b. await_response() waits until a response
is received and returns the response. close(a) closes the TCP
connection to client with address a. response(r) sends a response to
the client. ismulticast() returns 1 if the location is a multicast
address and zero otherwise. The variable timeleft indicates the
amount of time left until the maximum response time has expired. The
variable recurse indicates whether the server will recursively try
addresses returned through a 3xx response. A server MAY decide to
recursively try only certain addresses, e.g., those which are within
the same domain as the proxy server. Thus, an initial multicast
request can trigger additional unicast requests.
/* request type */
typedef enum {INVITE, ACK, BYE, OPTIONS, CANCEL, REGISTER} Method;
process_request(Method R, int N, address_t address[], int expires)
{
struct {
char *branch; /* branch token */
int branch_seq; /* branch sequence number part */
int done; /* has responded */
} outgoing[];
char *location[]; /* list of locations */
int heard = 0; /* number of sites heard from */
int class; /* class of status code */
int timeleft = expires; /* expiration value */
int loc = 0; /* number of locations */
struct { /* response */
int status; /* response: CANCEL=-1 */
int locations; /* number of redirect locations */
char *location[]; /* redirect locations */
address_t a; /* address of respondent */
char *branch; /* branch token */
int branch_seq; /* branch sequence number */
} r, best; /* response, best response */
int i;
best.status = 1000;
for (i = 0; i < N; i++) {
request(R, address[i], i);
outgoing[i].done = 0;
outgoing[i].branch = "";
outgoing[i].branch_seq = i;
}
while (timeleft > 0 && heard < N) {
r = await_response();
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class = r.status / 100;
/* If final response, mark branch as done. */
if (class >= 2) {
heard++;
for (i = 0; i < N; i++) {
if (r.branch_seq == outgoing[i].branch_seq) {
outgoing[i].done = 1;
break;
}
}
}
/* CANCEL: respond, fork and wait for responses */
/* terminate INVITE with 40
else if (class < 0) {
best.status = 200;
response(best);
for (i = 0; i < N; i++) {
if (!outgoing[i].done)
request(CANCEL, address[i], outgoing[i].branch);
}
best.status = -1;
}
/* Send an ACK */
if (class != 2) {
if (R == INVITE) request(ACK, r.a, r.branch);
}
if (class == 2) {
if (r.status < best.status) best = r;
break;
}
else if (class == 3) {
/* A server MAY optionally recurse. The server MUST check
* whether it has tried this location before and whether the
* location is part of the Via path of the incoming request.
* This check is omitted here for brevity. Multicast locations
* MUST NOT be returned to the client if the server is not
* recursing.
*/
if (recurse) {
multicast = 0;
N += r.locations;
for (i = 0; i < r.locations; i++) {
request(R, r.location[i]);
}
} else if (!ismulticast(r.location)) {
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best = r;
}
}
else if (class == 4) {
if (best.status >= 400) best = r;
}
else if (class == 5) {
if (best.status >= 500) best = r;
}
else if (class == 6) {
best = r;
break;
}
}
/* We haven't heard anything useful from anybody. */
if (best.status == 1000) {
best.status = 408; /* request expired */
}
if (best.status/100 != 3) loc = 0;
response(best);
}
Responses are processed as follows. The process completes (and state
can be freed) when all requests have been answered by final status
responses (for unicast) or 60 seconds have elapsed (for multicast). A
proxy MAY send a CANCEL to all incomplete branches and return the
best available final status to the client if not all responses have
been received after 60 seconds or the expiration period specified in
the Expires header field of the request. If no responses have been
received, the proxy returns a 408 (Timeout) response to the client.
When forwarding responses, a proxy MUST forward the whole response,
including all header fields of the selected response as well as the
body.
1xx: The proxy SHOULD forward provisional responses greater than
100 upstream towards the client and SHOULD NOT forward 100
(Trying) responses.
2xx: The proxy MUST forward the response upstream towards the
client, without sending an ACK downstream. After receiving
a 2xx, the server MAY terminate all other pending requests
by sending a CANCEL request and closing the TCP connection,
if applicable. (Terminating pending requests is advisable
as searches consume resources. Also, INVITE requests could
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"ring" on a number of workstations if the callee is
currently logged in more than once.)
3xx: For INVITE requests, the proxy MUST send an ACK. It MAY
recurse on the listed Contact addresses. Otherwise, the
lowest-numbered response is returned if there were no 2xx
or 6xx responses.
Location lists are not merged as that would prevent
forwarding of authenticated responses. Also, responses
can have message bodies, so that merging is not
feasible.
4xx, 5xx: For INVITE requests, the proxy MUST send an ACK. It
remembers the response if it has a lower status code class
than any previous 4xx and 5xx response. On completion, a
response with the lowest response class is returned if
there were no 2xx, 3xx or 6xx responses. Within the set of
responses from the lowest-numbered class, the proxy server
may choose any response.
The proxy SHOULD collect all WWW-Authenticate and Proxy-
Authenticate headers from all 401 and 407 responses and
return all of them in the response if either 401 or 407 is
the lowest-numbered response.
6xx: For INVITE requests, the proxy sends an ACK. It forwards
the 6xx response unless a 2xx response has been received.
Other pending requests MAY be terminated with CANCEL as
described for 2xx responses. Unlike for 2xx responses, only
one 6xx response is forwarded, since ACKs are generated
locally.
A proxy server forwards any response for Call-IDs for which it does
not have a pending transaction according to the response's Via
header. User agent servers respond to BYE requests for unknown call
legs with status code 481 (Transaction Does Not Exist); they drop ACK
requests with unknown call legs silently.
Special considerations apply for choosing forwarding destinations for
ACK and BYE requests. In most cases, these requests will bypass
proxies and reach the desired party directly, keeping proxies from
having to make forwarding decisions.
A proxy MAY maintain call state for a period of its choosing. If a
proxy still has list of destinations that it forwarded the last
INVITE to, it SHOULD direct ACK requests only to those downstream
servers.
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13 Security Considerations
13.1 Confidentiality and Privacy: Encryption
13.1.1 End-to-End Encryption
SIP requests and responses can contain sensitive information about
the communication patterns and communication content of individuals.
The SIP message body MAY also contain encryption keys for the session
itself. SIP supports three complementary forms of encryption to
protect privacy:
o End-to-end encryption of the SIP message body and certain
sensitive header fields;
o hop-by-hop encryption to prevent eavesdropping that tracks who
is calling whom;
o hop-by-hop encryption of Via fields to hide the route a
request has taken.
Not all of the SIP request or response can be encrypted end-to-end
because header fields such as To and Via need to be visible to
proxies so that the SIP request can be routed correctly. Hop-by-hop
encryption encrypts the entire SIP request or response on the wire so
that packet sniffers or other eavesdroppers cannot see who is calling
whom. Hop-by-hop encryption can also encrypt requests and responses
that have been end-to-end encrypted. Note that proxies can still see
who is calling whom, and this information is also deducible by
performing a network traffic analysis, so this provides a very
limited but still worthwhile degree of protection.
SIP Via fields are used to route a response back along the path taken
by the request and to prevent infinite request loops. However, the
information given by them can also provide useful information to an
attacker. Section 6.26 describes how a sender can request that Via
fields be encrypted by cooperating proxies without compromising the
purpose of the Via field.
End-to-end encryption relies on keys shared by the two user agents
involved in the request. Typically, the message is sent encrypted
with the public key of the recipient, so that only that recipient can
read the message. All implementations SHOULD support PGP-based
encryption [42] and MAY implement other schemes.
A SIP request (or response) is end-to-end encrypted by splitting the
message to be sent into a part to be encrypted and a short header
that will remain in the clear. Some parts of the SIP message, namely
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the request line, the response line and certain header fields marked
with "n" in the "enc." column in Table 4 and 5 need to be read and
returned by proxies and thus MUST NOT be encrypted end-to-end.
Possibly sensitive information that needs to be made available as
plaintext include destination address (To) and the forwarding path
(Via) of the call. The Authorization header field MUST remain in the
clear if it contains a digital signature as the signature is
generated after encryption, but MAY be encrypted if it contains
"basic" or "digest" authentication.
Other header fields MAY be encrypted or MAY travel in the clear as
desired by the sender. The Subject, Allow and Content-Type header
fields will typically be encrypted. The Accept, Accept-Language,
Date, Expires, Priority, Require, Call-ID, Cseq, and Timestamp header
fields will remain in the clear.
All fields that will remain in the clear MUST precede those that will
be encrypted. The message is encrypted starting with the first
character of the first header field that will be encrypted and
continuing through to the end of the message body. If no header
fields are to be encrypted, encrypting starts with the second CRLF
pair after the last header field, as shown below. Carriage return and
line feed characters have been made visible as "$", and the encrypted
part of the message is outlined.
INVITE sip:watson@boston.bell-telephone.com SIP/2.0$
Via: SIP/2.0/UDP 169.130.12.5$
To: T. A. Watson $
From: A. Bell $
Encryption: PGP version=5.0$
Content-Length: 224$
Call-ID: 187602141351@worcester.bell-telephone.com$
Content-Type: message/sip
CSeq: 488$
$
*******************************************************
* Subject: Mr. Watson, come here.$ *
* Content-Type: application/sdp$ *
* $ *
* v=0$ *
* o=bell 53655765 2353687637 IN IP4 128.3.4.5$ *
* s=Mr. Watson, come here.$ *
* t=0 0$ *
* c=IN IP4 135.180.144.94$ *
* m=audio 3456 RTP/AVP 0 3 4 5$ *
*******************************************************
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An Encryption header field MUST be added to indicate the encryption
mechanism used. A Content-Length field is added that indicates the
length of the encrypted body. The encrypted body is preceded by a
blank line as a normal SIP message body would be.
Upon receipt by the called user agent possessing the correct
decryption key, the message body as indicated by the Content-Length
field is decrypted, and the now-decrypted body is appended to the
clear-text header fields. There is no need for an additional
Content-Length header field within the encrypted body because the
length of the actual message body is unambiguous after decryption.
A Content-Type indication of "message/sip" MAY be added, but will be
overridden after receipt.
Had no SIP header fields required encryption, the message would have
been as below. Note that the encrypted body MUST then include a blank
line (start with CRLF) to disambiguate between any possible SIP
header fields that might have been present and the SIP message body.
INVITE sip:watson@boston.bell-telephone.com SIP/2.0$
Via: SIP/2.0/UDP 169.130.12.5$
To: T. A. Watson $
From: A. Bell $
Encryption: PGP version=5.0$
Content-Type: application/sdp$
Content-Length: 107$
Call-ID: 187602141351@worcester.bell-telephone.com$
CSeq: 488$
$
*************************************************
* $ *
* v=0$ *
* o=bell 53655765 2353687637 IN IP4 128.3.4.5$ *
* c=IN IP4 135.180.144.94$ *
* m=audio 3456 RTP/AVP 0 3 4 5$ *
*************************************************
13.1.2 Privacy of SIP Responses
SIP requests can be sent securely using end-to-end encryption and
authentication to a called user agent that sends an insecure
response. This is allowed by the SIP security model, but is not a
good idea. However, unless the correct behavior is explicit, it
would not always be possible for the called user agent to infer what
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a reasonable behavior was. Thus, when end-to-end encryption is used
by the request originator, the encryption key to be used for the
response SHOULD be specified in the request (Section 6.37). If this
were not done, it might be possible for the called user agent to
incorrectly infer an appropriate key to use in the response. Thus, to
prevent key-guessing becoming an acceptable strategy, we specify that
a called user agent receiving a request that does not specify a key
to be used for the response SHOULD send that response unencrypted.
Any SIP header fields that were encrypted in a request SHOULD also be
encrypted in an encrypted response. Contact response fields MAY be
encrypted if the information they contain is sensitive, or MAY be
left in the clear to permit proxies more scope for localized
searches.
13.1.3 Encryption by Proxies
Normally, proxies are not allowed to alter end-to-end header fields
and message bodies. Proxies MAY, however, encrypt an unsigned request
or response with the key of the call recipient.
Proxies need to encrypt a SIP request if the end system
cannot perform encryption or to enforce organizational
security policies.
13.1.4 Hop-by-Hop Encryption
SIP requests and responses MAY also be protected by security
mechanisms at the transport or network layer. No particular mechanism
is defined or recommended here. Two possibilities are IPSEC [43] or
TLS [24]. The use of a particular mechanism will generally need to be
specified out of band, through manual configuration, for example.
13.1.5 Via field encryption
When Via header fields are to be hidden, a proxy that receives a
request containing an appropriate "Hide: hop" header field (as
specified in section 6.26) SHOULD encrypt the header field. As only
the proxy that encrypts the field will decrypt it, the algorithm
chosen is entirely up to the proxy implementor. Two methods satisfy
these requirements:
o The server keeps a cache of Via header fields and the
associated To header field, and replaces the Via header field
with an index into the cache. On the reverse path, take the
Via header field from the cache rather than the message.
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This is insufficient to prevent message looping, and so an
additional ID MUST be added so that the proxy can detect
loops. This SHOULD NOT normally be the address of the proxy as
the goal is to hide the route, so instead a sufficiently large
random number SHOULD be used by the proxy and maintained in
the cache.
It is possible for replies to get directed to the wrong
originator if the cache entry gets reused, so great care needs
to be taken to ensure this does not happen.
o The server MAY use a secret key to encrypt the Via field, a
timestamp and an appropriate checksum in any such message with
the same secret key. The checksum is needed to detect whether
successful decoding has occurred, and the timestamp is
required to prevent possible replay attacks and to ensure that
no two requests from the same previous hop have the same
encrypted Via field. This is the preferred solution.
13.2 Message Integrity and Access Control: Authentication
Protective measures need to be taken to prevent an active attacker
from modifying and replaying SIP requests and responses. The same
cryptographic measures that are used to ensure the authenticity of
the SIP message also serve to authenticate the originator of the
message. However, the "basic" and "digest" authentication mechanism
offer authentication only, without message integrity.
Transport-layer or network-layer authentication MAY be used for hop-
by-hop authentication. SIP also extends the HTTP WWW-Authenticate
(Section 6.49) and Authorization (Section 6.12) header field and
their Proxy counterparts to include cryptographically strong
signatures. SIP also supports the HTTP "basic" and "digest" schemes
(see Section 14) and other HTTP authentication schemes to be defined
that offer a rudimentary mechanism of ascertaining the identity of
the caller.
SIP requests MAY be authenticated using the Authorization header
field to include a digital signature of certain header fields, the
request method and version number and the payload, none of which are
modified between client and called user agent. The Authorization
header field is used in requests to authenticate the request
originator end-to-end to proxies and the called user agent, and in
responses to authenticate the called user agent or proxies returning
their own failure codes. If required, hop-by-hop authentication can
be provided, for example, by the IPSEC Authentication Header.
SIP does not dictate which digital signature scheme is used for
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authentication, but does define how to provide authentication using
PGP in Section 15. As indicated above, SIP implementations MAY also
use "basic" and "digest" authentication and other authentication
mechanisms defined for HTTP [44]. Note that "basic" authentication
has severe security limitations. The following does not apply to
these schemes.
To cryptographically sign a SIP request, the order of the SIP header
fields is important. When an Authorization header field is present,
it indicates that all header fields following the Authorization
header field have been included in the signature. Therefore, hop-
by-hop header fields which MUST or SHOULD be modified by proxies MUST
precede the Authorization header field as they will generally be
modified or added-to by proxy servers. Hop-by-hop header fields
which MAY be modified by a proxy MAY appear before or after the
Authorization header. When they appear before, they MAY be modified
by a proxy. When they appear after, they MUST NOT be modified by a
proxy. To sign a request, a client constructs a message from the
request method (in upper case) followed, without LWS, by the SIP
version number, followed, again without LWS, by the request headers
to be signed and the message body. The message thus constructed is
then signed.
For example, if the SIP request is to be:
INVITE sip:watson@boston.bell-telephone.com SIP/2.0
Via: SIP/2.0/UDP 169.130.12.5
Authorization: PGP version=5.0, signature=...
From: A. Bell
To: T. A. Watson
Call-ID: 187602141351@worcester.bell-telephone.com
Subject: Mr. Watson, come here.
Content-Type: application/sdp
Content-Length: ...
v=0
o=bell 53655765 2353687637 IN IP4 128.3.4.5
s=Mr. Watson, come here.
t=0 0
c=IN IP4 135.180.144.94
m=audio 3456 RTP/AVP 0 3 4 5
Then the data block that is signed is:
INVITESIP/2.0From: A. Bell
To: T. A. Watson
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Call-ID: 187602141351@worcester.bell-telephone.com
Subject: Mr. Watson, come here.
Content-Type: application/sdp
Content-Length: ...
v=0
o=bell 53655765 2353687637 IN IP4 128.3.4.5
s=Mr. Watson, come here.
t=0 0
c=IN IP4 135.180.144.94
m=audio 3456 RTP/AVP 0 3 4 5
Clients wishing to authenticate requests MUST construct the portion
of the message below the Authorization header using a canonical form.
This allows a proxy to parse the message, take it apart, and
reconstruct it, without causing an authentication failure due to
extra white space, for example. Canonical form consists of the
following rules:
o No short form header fields;
o Header field names are capitalized as shown in this document;
o No white space between the header name and the colon;
o A single space after the colon;
o Line termination with a CRLF;
o No line folding;
o No comma separated lists of header values; each must appear as
a separate header;
o Only a single SP between tokens, between tokens and quoted
strings, and between quoted strings; no SP after last token or
quoted string;
o No LWS between tokens and separators, except as described
above for after the colon in header fields;
o The To and From header fields always include the < and >
delimiters even if the display-name is empty.
Note that if a message is encrypted and authenticated using a digital
signature, when the message is generated encryption is performed
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before the digital signature is generated. On receipt, the digital
signature is checked before decryption.
A client MAY require that a server sign its response by including a
Require: signed-response request header field. The client indicates
the desired authentication method via the WWW-Authenticate header.
The correct behavior in handling unauthenticated responses to a
request that requires authenticated responses is described in section
13.2.1.
13.2.1 Trusting responses
There is the possibility that an eavesdropper listens to requests and
then injects unauthenticated responses that terminate, redirect or
otherwise interfere with a call. (Even encrypted requests contain
enough information to fake a response.)
Clients need to be particularly careful with 3xx redirection
responses. Thus a client receiving, for example, a 301 (Moved
Permanently) which was not authenticated when the public key of the
called user agent is known to the client, and authentication was
requested in the request SHOULD be treated as suspicious. The correct
behavior in such a case would be for the called-user to form a dated
response containing the Contact field to be used, to sign it, and
give this signed stub response to the proxy that will provide the
redirection. Thus the response can be authenticated correctly. A
client SHOULD NOT automatically redirect such a request to the new
location without alerting the user to the authentication failure
before doing so.
Another problem might be responses such as 6xx failure responses
which would simply terminate a search, or "4xx" and "5xx" response
failures.
If TCP is being used, a proxy SHOULD treat 4xx and 5xx responses as
valid, as they will not terminate a search. However, fake 6xx
responses from a rogue proxy terminate a search incorrectly. 6xx
responses SHOULD be authenticated if requested by the client, and
failure to do so SHOULD cause such a client to ignore the 6xx
response and continue a search.
With UDP, the same problem with 6xx responses exists, but also an
active eavesdropper can generate 4xx and 5xx responses that might
cause a proxy or client to believe a failure occurred when in fact it
did not. Typically 4xx and 5xx responses will not be signed by the
called user agent, and so there is no simple way to detect these
rogue responses. This problem is best prevented by using hop-by-hop
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encryption of the SIP request, which removes any additional problems
that UDP might have over TCP.
These attacks are prevented by having the client require response
authentication and dropping unauthenticated responses. A server user
agent that cannot perform response authentication responds using the
normal Require response of 420 (Bad Extension).
13.3 Callee Privacy
User location and SIP-initiated calls can violate a callee's privacy.
An implementation SHOULD be able to restrict, on a per-user basis,
what kind of location and availability information is given out to
certain classes of callers.
13.4 Known Security Problems
With either TCP or UDP, a denial of service attack exists by a rogue
proxy sending 6xx responses. Although a client SHOULD choose to
ignore such responses if it requested authentication, a proxy cannot
do so. It is obliged to forward the 6xx response back to the client.
The client can then ignore the response, but if it repeats the
request it will probably reach the same rogue proxy again, and the
process will repeat.
14 SIP Authentication using HTTP Basic and Digest Schemes
SIP implementations MAY use HTTP's basic and digest authentication
mechanisms (RFC 2617 [44]) to provide a rudimentary form of security.
This section overviews usage of these mechanisms in SIP. The basic
operation is almost completely identical to that for HTTP [44]. This
section outlines this operation, pointing to RFC 2617 [44] for
details, and noting the differences when used in SIP.
14.1 Framework
The framework for SIP authentication parallels that for HTTP (RFC
2617 [44]). In particular, the BNF for auth-scheme, auth-param,
challenge, realm, realm-value, and credentials is identical. The 401
response is used by user agent servers in SIP to challenge the
authorization of a user agent client. Additionally, registrars and
redirect servers MAY make use of 401 responses for authorization, but
proxies MUST NOT, and instead MAY use the 407 response. The
requirements for inclusion of the Proxy-Authenticate, Proxy-
Authorization, WWW-Authenticate, and Authorization in the various
messages is identical to RFC 2617 [44].
Since SIP does not have the concept of a canonical root URL, the
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notion of protections spaces are interpreted differently for SIP. The
realm is a protection domain for all SIP URIs with the same value for
the userinfo, host and port part of the SIP Request-URI. For example:
INVITE sip:alice.wonderland@example.com SIP/2.0
WWW-Authenticate: Basic realm="business"
and
INVITE sip:aw@example.com SIP/2.0
WWW-Authenticate: Basic realm="business"
define different protection realms according to this rule.
When a UAC resubmits a request with its credentials after receiving a
401 or 407 response, it MUST increment the CSeq header field as it
would normally do when sending an updated request.
14.2 Basic Authentication
The rules for basic authentication follow those defined in [44] but
with the words "origin server" replaced with "user agent server,
redirect server , or registrar".
Since SIP URIs are not hierarchical, the paragraph in [44] that
states that "all paths at or deeper than the depth of the last
symbolic element in the path field of the Request-URI also are within
the protection space specified by the Basic realm value of the
current challenge" does not apply for SIP. SIP clients MAY
preemptively send the corresponding Authorization header with
requests for SIP URIs within the same protection realm (as defined
above) without receipt of another challenge from the server.
14.3 Digest Authentication
The rules for digest authentication follow those defined in [44],
with "HTTP 1.1" replaced by "SIP/2.0" in addition to the following
differences:
1. The URI included in the challenge has the following BNF:
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URI = SIP-URL
2. The BNF in RFC 2617 has an error in that the URI is not
enclosed in quotation marks. (The example in Section 3.5 is
correct.) For SIP, the URI MUST be enclosed in quotation
marks.
3. The BNF for digest-uri-value is:
digest-uri-value = Request-URI ; as defined in Section
4.3
4. The example procedure for choosing a nonce based on Etag
does not work for SIP.
5. The Authentication-Info and Proxy-Authentication-Info
fields are not used in SIP.
6. The text in RFC 2617 [44] regarding cache operation does
not apply to SIP.
7. RFC 2617 [44] requires that a server check that the URI in
the request line, and the URI included in the Authorization
header, point to the same resource. In a SIP context, these
two URI's may actually refer to different users, due to
forwarding at some proxy. Therefore, in SIP, a server MAY
check that the request-uri in the Authorization header
corresponds to a user that the server is willing to accept
forwarded or direct calls for.
14.4 Proxy-Authentication
The use of the Proxy-Authentication and Proxy-Authorization parallel
that as described in [44], with one difference. Proxies MUST NOT add
the Proxy-Authorization header. 407 responses MUST be forwarded
upstream towards the client following the procedures for any other
response. It is the client's responsibility to add the Proxy-
Authorization header containing credentials for the proxy which has
asked for authentication.
If a proxy were to resubmit a request with a Proxy-
Authorization header field, it would need to increment the
CSeq in the new request. However, this would mean that the
UAC which submitted the original request would discard a
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response from the UAS, as the CSeq value would be
different.
See sections 6.32 and 6.33 for additional information on usage of
these fields as they apply to SIP.
It is also possible that a 401 (Unauthorized) response contains
several challenges, from a mixture of proxies and user agent servers,
if the request was forked.
15 SIP Security Using PGP
15.1 PGP Authentication Scheme
The "pgp" authentication scheme is based on the model that the client
authenticates itself with a request signed with the client's private
key. The server can then ascertain the origin of the request if it
has access to the public key, preferably signed by a trusted third
party. Implementations supporting this scheme MUST implement the
definitions and default algorithms of RFC 2440 [45] and MAY implement
the older version, based upon PGP 2.6, described in RFC 1991 [42].
15.1.1 The WWW-Authenticate Response Header
WWW-Authenticate = "WWW-Authenticate" ":" "pgp" pgp-challenge
pgp-challenge = # pgp-params
pgp-params = realm | pgp-version | pgp-micalgorithm
| pgp-pubalgorithm | nonce
realm = "realm" "=" realm-value
realm-value = quoted-string
pgp-version = "version" "="
<"> digit *( "." digit ) *letter <">
pgp-micalgorithm = "algorithm" "=" ( "md5" | "sha1" | token
| "ripemd160" | "MD2" | "TIGER192" | "HAVAL-5-160")
pgp-pubalgorithm = "pubkey" "=" ( "rsa" | "rsa-encrypt"
| "rsa-sign" | "elgamal" | "dsa" | token )
nonce = "nonce" "=" nonce-value
nonce-value = quoted-string
The meanings of the values of the parameters used above are as
follows:
realm: A string to be displayed to users so they know which
identity to use. This string SHOULD contain at least the
name of the host performing the authentication and MAY
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additionally indicate the collection of users who might
have access. An example might be "Users with call-out
privileges".
pgp-micalgorithm: The value of this parameter indicates the PGP
message integrity check (MIC) to be used to produce the
signature. If this parameter is not present, it is assumed
to be "md5". The currently defined values are "md5" for
the MD5 checksum, and "sha1" for the SHA.1 algorithm.
pgp-pubalgorithm: The value of this parameter indicates the PGP
public-key algorithm to be used for signing and encrypting
messages. If this parameter is not present, it is assumed
to be "rsa" for the RSA algorithm. The value "dsa" defines
the DSS/DH algorithm.
pgp-version: The version of PGP that the client MUST use. Common
values are "2.6.2" and "5.0". The default is 5.0.
nonce: A server-specified data string which should be uniquely
generated each time a 401 response is made. It is
RECOMMENDED that this string be base64 [46] or hexadecimal
data. Specifically, since the string is passed in the
header lines as a quoted string, the double-quote character
is not allowed. The contents of the nonce are
implementation dependent. The quality of the implementation
depends on a good choice. Since the nonce is used only to
prevent replay attacks and is signed, a time stamp in units
convenient to the server is sufficient.
Replay attacks within the duration of the call setup
are of limited interest, so that timestamps with a
resolution of a few seconds are often sufficient. In
that case, the server does not have to keep a record
of the nonces.
Example:
WWW-Authenticate: pgp version="5.0"
realm="Your Startrek identity, please", algorithm=md5,
nonce="913082051"
15.1.2 The Authorization Request Header
The client is expected to retry the request, passing an Authorization
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header line, which is defined as follows.
Authorization = "Authorization" ":" "pgp" # pgp-response
pgp-response = realm | pgp-version | pgp-signature
| signed-by | nonce
pgp-signature = "signature" "=" quoted-string
signed-by = "signed-by" "=" <"> URI <">
The client MUST increment the CSeq header before resubmitting the
request. The signature MUST correspond to the From header of the
request unless the signed-by parameter is provided.
pgp-signature: The PGP ASCII-armored signature [42] and [45], as
it appears between the "BEGIN PGP MESSAGE" and "END PGP
MESSAGE" delimiters, without the version indication. The
signature is included without any linebreaks.
The signature is computed, in order, across the nonce (if
present), realm, request method, request version and header
fields following the Authorization header and the message
body, in the same order as they appear in the message. The
nonce, realm, request method and version are prepended to
the header fields without any white space. The signature is
computed across the headers as sent, and the terminating
CRLF. The CRLF following the Authorization header is NOT
included in the signature.
UACs MAY attempt to authenticate themselves without a nonce
on the first INVITE request rather than waiting for a 401
response if the UAC knows, e.g., from past requests or
local configuration, that the UAS supports PGP
authentication. A server MAY be configured not to generate
nonces only if replay attacks are not a concern.
Not generating nonces avoids the additional set of
request, 401 response and possibly ACK messages and
reduces delay by one round-trip time.
Using the ASCII-armored version is about 25% less
space-efficient than including the binary signature,
but it is significantly easier for the receiver to
piece together. Versions of the PGP program always
include the full (compressed) signed text in their
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output unless ASCII-armored mode ( -sta ) is
specified. Typical signatures are about 200 bytes
long. -- The PGP signature mechanism allows the client
to simply pass the request to an external PGP program.
This relies on the requirement that proxy servers are
not allowed to reorder or change header fields.
realm: The realm is copied from the corresponding WWW-
Authenticate header field parameter.
signed-by: If and only if the request was not signed by the
entity listed in the From header, the signed-by header
indicates the name of the signing entity, expressed as a
URI.
Receivers of signed SIP messages SHOULD discard any end-to-end header
fields above the Authorization header, as they may have been
maliciously added en route by a proxy.
Example:
Authorization: pgp version="5.0",
realm="Your Startrek identity, please",
nonce="913082051"
signature="iQB1AwUBNNJiUaYBnHmiiQh1AQFYsgL/Wt3dk6TWK81/b0gcNDf
VAUGU4rhEBW972IPxFSOZ94L1qhCLInTPaqhHFw1cb3lB01rA0RhpV4t5yCdUt
SRYBSkOK29o5e1KlFeW23EzYPVUm2TlDAhbcjbMdfC+KLFX
=aIrx"
15.2 PGP Encryption Scheme
The PGP encryption scheme uses the following syntax:
Encryption = "Encryption" ":" "pgp" pgp-eparams
pgp-eparams = 1# ( pgp-version | pgp-encoding )
pgp-encoding = "encoding" "=" "ascii" | token
encoding: Describes the encoding or "armor" used by PGP. The
value "ascii" refers to the standard PGP ASCII armor,
without the lines containing "BEGIN PGP MESSAGE" and "END
PGP MESSAGE" and without the version identifier. By
default, the encrypted part is included as binary.
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Example:
Encryption: pgp version="2.6.2", encoding="ascii"
15.3 Response-Key Header Field for PGP
Response-Key = "Response-Key" ":" "pgp" pgp-eparams
pgp-eparams = 1# ( pgp-version | pgp-encoding | pgp-key)
pgp-key = "key" "=" quoted-string
If ASCII encoding has been requested via the encoding parameter, the
key parameter contains the user's public key as extracted from the
pgp key ring with the "pgp -kxa user ".
Example:
Response-Key: pgp version="2.6.2", encoding="ascii",
key="mQBtAzNWHNYAAAEDAL7QvAdK2utY05wuUG+ItYK5tCF8HNJM60sU4rLaV+eUnkMk
mOmJWtc2wXcZx1XaXb2lkydTQOesrUR75IwNXBuZXPEIMThEa5WLsT7VLme7njnx
sE86SgWmAZx5ookIdQAFEbQxSGVubmluZyBTY2h1bHpyaW5uZSA8c2NodWx6cmlu
bmVAY3MuY29sdW1iaWEuZWR1Pg==
=+y19"
16 Examples
In the following examples, we often omit the message body and the
corresponding Content-Length and Content-Type headers for brevity.
16.1 Registration
A user at host saturn.bell-tel.com registers on start-up, via
multicast, with the local SIP server named bell-tel.com. In the
example, the user agent on saturn expects to receive SIP requests on
UDP port 3890.
C->S: REGISTER sip:bell-tel.com SIP/2.0
Via: SIP/2.0/UDP saturn.bell-tel.com
From: sip:watson@bell-tel.com
To: sip:watson@bell-tel.com
Call-ID: 70710@saturn.bell-tel.com
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CSeq: 1 REGISTER
Contact:
Expires: 7200
The registration expires after two hours. Any future invitations for
watson@bell-tel.com arriving at sip.bell-tel.com will now be
redirected to watson@saturn.bell-tel.com, UDP port 3890.
If Watson wants to be reached elsewhere, say, an on-line service he
uses while traveling, he updates his reservation after first
cancelling any existing locations:
C->S: REGISTER sip:bell-tel.com SIP/2.0
Via: SIP/2.0/UDP saturn.bell-tel.com
From: sip:watson@bell-tel.com
To: sip:watson@bell-tel.com
Call-ID: 70710@saturn.bell-tel.com
CSeq: 2 REGISTER
Contact: *
Expires: 0
C->S: REGISTER sip:bell-tel.com SIP/2.0
Via: SIP/2.0/UDP saturn.bell-tel.com
From: sip:watson@bell-tel.com
To: sip:watson@bell-tel.com
Call-ID: 70710@saturn.bell-tel.com
CSeq: 3 REGISTER
Contact: sip:tawatson@example.com
Now, the server will forward any request for Watson to the server at
example.com, using the Request-URI tawatson@example.com. For the
server at example.com to reach Watson, he will need to send a
REGISTER there, or inform the server of his current location through
some other means.
It is possible to use third-party registration. Here, the secretary
jon.diligent registers his boss, T. Watson:
C->S: REGISTER sip:bell-tel.com SIP/2.0
Via: SIP/2.0/UDP pluto.bell-tel.com
From: sip:jon.diligent@bell-tel.com
To: sip:watson@bell-tel.com
Call-ID: 17320@pluto.bell-tel.com
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CSeq: 1 REGISTER
Contact: sip:tawatson@example.com
The request could be sent to either the registrar at bell-tel.com or
the server at example.com. In the latter case, the server at
example.com would proxy the request to the address indicated in the
Request-URI. Then, Max-Forwards header could be used to restrict the
registration to that server.
16.2 Invitation to a Multicast Conference
The first example invites bob@example.com to a multicast session.
All examples use the Session Description Protocol (SDP) (RFC 2327
[6]) as the session description format.
16.2.1 Request
C->S: INVITE sip:bob@one.example.com SIP/2.0
Via: SIP/2.0/UDP sip.example.com;branch=7c337f30d7ce.1
;maddr=239.128.16.254;ttl=16
Via: SIP/2.0/UDP mouse.wonderland.com
From: Alice
To: Bob
Call-ID: 602214199@mouse.wonderland.com
CSeq: 1 INVITE
Contact: Alice
Subject: SIP will be discussed, too
Content-Type: application/sdp
Content-Length: 187
v=0
o=user1 53655765 2353687637 IN IP4 128.3.4.5
s=Mbone Audio
t=3149328700 0
i=Discussion of Mbone Engineering Issues
e=mbone@somewhere.com
c=IN IP4 224.2.0.1/127
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
The From request header above states that the request was initiated
by alice@wonderland.com and addressed to bob@example.com (From header
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fields). The Via fields list the hosts along the path from invitation
initiator (the last element of the list) towards the callee. In the
example above, the message was last multicast to the administratively
scoped group 239.128.16.254 with a ttl of 16 from the host
sip.example.com. The second Via header field indicates that it was
originally sent from the outbound proxy mouse.wonderland.com. The
Request-URI indicates that the request is currently being being
addressed to bob@one.example.com, the local address that the SIP
server for the example.com domain looked up for the callee.
In this case, the session description is using the Session
Description Protocol (SDP), as stated in the Content-Type header.
The header is terminated by an empty line and is followed by a
message body containing the session description.
16.2.2 Response
The called user agent, directly or indirectly through proxy servers,
indicates that it is alerting ("ringing") the called party:
S->C: SIP/2.0 180 Ringing
Via: SIP/2.0/UDP csvax.cs.caltech.edu;branch=7c337f30d7ce.1
;maddr=239.128.16.254;ttl=16
Via: SIP/2.0/UDP north.east.isi.edu
From: Alice
To: Bob ;tag=3141593
Call-ID: 602214199@mouse.wonderland.com
CSeq: 1 INVITE
A sample response to the invitation is given below. The first line of
the response states the SIP version number, that it is a 200 (OK)
response, which means the request was successful. The Via headers are
taken from the request, and entries are removed hop by hop as the
response retraces the path of the request. A new authentication field
MAY be added by the invited user's agent if required. The Call-ID is
taken directly from the original request, along with the remaining
fields of the request message. The original sense of From field is
preserved (i.e., it is the session initiator).
In addition, the Contact header gives details of the host where the
user was located, or alternatively the relevant proxy contact point
which should be reachable from the caller's host.
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S->C: SIP/2.0 200 OK
Via: SIP/2.0/UDP csvax.cs.caltech.edu;branch=7c337f30d7ce.1
;maddr=239.128.16.254;ttl=16
Via: SIP/2.0/UDP north.east.isi.edu
From: Alice
To: Bob ;tag=3141593
Call-ID: 602214199@mouse.wonderland.com
CSeq: 1 INVITE
Contact:
The caller confirms the invitation by sending an ACK request to the
location named in the Contact header:
C->S: ACK sip:bob@one.example.com SIP/2.0
Via: SIP/2.0/UDP north.east.isi.edu
From: Alice
To: Bob ;tag=3141593
Call-ID: 602214199@mouse.wonderland.com
CSeq: 1 ACK
16.3 Two-party Call
For two-party Internet phone calls, the response must contain a
description of where to send the data. In the example below, Bell
calls Watson. Bell indicates that he can receive RTP audio codings 0
(PCMU), 3 (GSM), 4 (G.723) and 5 (DVI4).
C->S: INVITE sip:watson@boston.bell-tel.com SIP/2.0
Via: SIP/2.0/UDP kton.bell-tel.com
From: A. Bell
To: T. Watson
Call-ID: 662606876@kton.bell-tel.com
CSeq: 1 INVITE
Contact:
Subject: Mr. Watson, come here.
Content-Type: application/sdp
Content-Length: ...
v=0
o=bell 53655765 2353687637 IN IP4 128.3.4.5
s=Mr. Watson, come here.
t=3149328600 0
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c=IN IP4 kton.bell-tel.com
m=audio 3456 RTP/AVP 0 3 4 5
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=rtpmap:5 DVI4/8000
S->C: SIP/2.0 100 Trying
Via: SIP/2.0/UDP kton.bell-tel.com
From: A. Bell
To: T. Watson ;tag=37462311
Call-ID: 662606876@kton.bell-tel.com
CSeq: 1 INVITE
Content-Length: 0
S->C: SIP/2.0 180 Ringing
Via: SIP/2.0/UDP kton.bell-tel.com
From: A. Bell
To: T. Watson ;tag=37462311
Call-ID: 662606876@kton.bell-tel.com
CSeq: 1 INVITE
Content-Length: 0
S->C: SIP/2.0 182 Queued, 2 callers ahead
Via: SIP/2.0/UDP kton.bell-tel.com
From: A. Bell
To: T. Watson ;tag=37462311
Call-ID: 3298420296@kton.bell-tel.com
CSeq: 1 INVITE
Content-Length: 0
S->C: SIP/2.0 182 Queued, 1 caller ahead
Via: SIP/2.0/UDP kton.bell-tel.com
From: A. Bell
To: T. Watson ;tag=37462311
Call-ID: 3298420296@kton.bell-tel.com
CSeq: 1 INVITE
Content-Length: 0
S->C: SIP/2.0 200 OK
Via: SIP/2.0/UDP kton.bell-tel.com
From: A. Bell
To: ;tag=37462311
Call-ID: 3298420296@kton.bell-tel.com
CSeq: 1 INVITE
Contact: sip:watson@boston.bell-tel.com
Content-Type: application/sdp
Content-Length: ...
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v=0
o=watson 4858949 4858949 IN IP4 192.1.2.3
s=I'm on my way
t=3149329600 0
c=IN IP4 boston.bell-tel.com
m=audio 5004 RTP/AVP 0 3
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
The example illustrates the use of informational status responses.
Here, the reception of the call is confirmed immediately (100), then,
possibly after some database mapping delay, the call rings (180) and
is then queued, with periodic status updates.
Watson can only receive PCMU and GSM. Note that Watson's list of
codecs may or may not be a subset of the one offered by Bell, as each
party indicates the data types it is willing to receive. Watson will
send audio data to port 3456 at c.bell-tel.com, Bell will send to
port 5004 at boston.bell-tel.com.
By default, the media session is one RTP session. Watson will receive
RTCP packets on port 5005, while Bell will receive them on port 3457.
Since the two sides have agreed on the set of media, Bell confirms
the call without enclosing another session description:
C->S: ACK sip:watson@boston.bell-tel.com SIP/2.0
Via: SIP/2.0/UDP kton.bell-tel.com
From: A. Bell
To: T. Watson ;tag=37462311
Call-ID: 3298420296@kton.bell-tel.com
CSeq: 1 ACK
16.4 Terminating a Call
To terminate a call, caller or callee can send a BYE request:
C->S: BYE sip:watson@boston.bell-tel.com SIP/2.0
Via: SIP/2.0/UDP kton.bell-tel.com
From: A. Bell
To: T. A. Watson ;tag=37462311
Call-ID: 3298420296@kton.bell-tel.com
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CSeq: 2 BYE
If the callee wants to abort the call, it simply reverses the To and
From fields. Note that it is unlikely that a BYE from the callee will
traverse the same proxies as the original INVITE.
16.5 Forking Proxy
In this example, Bell (a.g.bell@bell-tel.com) (C), currently seated
at host c.bell-tel.com wants to call Watson (t.watson@ieee.org). At
the time of the call, Watson is logged in at two workstations,
t.watson@x.bell-tel.com (X) and watson@y.bell-tel.com (Y), and has
registered with the IEEE proxy server (P) called sip.ieee.org. The
IEEE server also has a registration for the home machine of Watson,
at watson@h.bell-tel.com (H), as well as a permanent registration at
watson@acm.org (A). For brevity, the examples omit the message bodies
containing the session descriptions.
Bell's user agent sends the invitation to the SIP server for the
ieee.org domain:
C->P: INVITE sip:t.watson@ieee.org SIP/2.0
Via: SIP/2.0/UDP c.bell-tel.com
From: A. Bell
To: T. Watson
Call-ID: 31415@c.bell-tel.com
CSeq: 1 INVITE
Contact: a.g.bell@c.bell-tel.com
The SIP server at ieee.org tries the four addresses in parallel. It
sends the following message to the home machine:
P->H: INVITE sip:watson@h.bell-tel.com SIP/2.0
Via: SIP/2.0/UDP sip.ieee.org ;branch=3d8a50dbf5a28d.1
Via: SIP/2.0/UDP c.bell-tel.com
From: A. Bell
To: T. Watson
Call-ID: 31415@c.bell-tel.com
CSeq: 1 INVITE
Contact: a.g.bell@c.bell-tel.com
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This request immediately yields a 404 (Not Found) response, since
Watson is not currently logged in at home:
H->P: SIP/2.0 404 Not Found
Via: SIP/2.0/UDP sip.ieee.org ;branch=3d8a50dbf5a28d.1
Via: SIP/2.0/UDP c.bell-tel.com
From: A. Bell
To: T. Watson ;tag=87454273
Call-ID: 31415@c.bell-tel.com
CSeq: 1 INVITE
The proxy ACKs the response so that host H can stop retransmitting
it:
P->H: ACK sip:watson@h.bell-tel.com SIP/2.0
Via: SIP/2.0/UDP sip.ieee.org ;branch=3d8a50dbf5a28d.1
From: A. Bell
To: T. Watson ;tag=87454273
Call-ID: 31415@c.bell-tel.com
CSeq: 1 ACK
Also, P attempts to reach Watson through the ACM server:
P->A: INVITE sip:watson@acm.org SIP/2.0
Via: SIP/2.0/UDP sip.ieee.org ;branch=3d8a50dbf5a28d.2
Via: SIP/2.0/UDP c.bell-tel.com
From: A. Bell
To: T. Watson
Call-ID: 31415@c.bell-tel.com
CSeq: 1 INVITE
Contact: a.g.bell@c.bell-tel.com
In parallel, the next attempt proceeds, with an INVITE to X and Y:
P->X: INVITE sip:t.watson@x.bell-tel.com SIP/2.0
Via: SIP/2.0/UDP sip.ieee.org ;branch=3d8a50dbf5a28d.3
Via: SIP/2.0/UDP c.bell-tel.com
From: A. Bell
To: T. Watson
Call-ID: 31415@c.bell-tel.com
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CSeq: 1 INVITE
Contact: a.g.bell@c.bell-tel.com
P->Y: INVITE sip:watson@y.bell-tel.com SIP/2.0
Via: SIP/2.0/UDP sip.ieee.org ;branch=3d8a50dbf5a28d.4
Via: SIP/2.0/UDP c.bell-tel.com
From: A. Bell
To: T. Watson
Call-ID: 31415@c.bell-tel.com
CSeq: 1 INVITE
Contact: a.g.bell@c.bell-tel.com
As it happens, both Watson at X and a colleague in the other lab at
host Y hear the phones ringing and pick up. Both X and Y return 200s
via the proxy to Bell.
X->P: SIP/2.0 200 OK
Via: SIP/2.0/UDP sip.ieee.org ;branch=3d8a50dbf5a28d.3
Via: SIP/2.0/UDP c.bell-tel.com
From: A. Bell
To: T. Watson ;tag=192137601
Call-ID: 31415@c.bell-tel.com
CSeq: 1 INVITE
Contact: sip:t.watson@x.bell-tel.com
Y->P: SIP/2.0 200 OK
Via: SIP/2.0/UDP sip.ieee.org ;branch=3d8a50dbf5a28d.4
Via: SIP/2.0/UDP c.bell-tel.com
Contact: sip:t.watson@y.bell-tel.com
From: A. Bell
To: T. Watson ;tag=35253448
Call-ID: 31415@c.bell-tel.com
CSeq: 1 INVITE
Both responses are forwarded to Bell, using the Via information. At
this point, the ACM server is still searching its database. P can now
cancel this attempt:
P->A: CANCEL sip:watson@acm.org SIP/2.0
Via: SIP/2.0/UDP sip.ieee.org ;branch=3d8a50dbf5a28d.2
From: A. Bell
To: T. Watson
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Call-ID: 31415@c.bell-tel.com
CSeq: 1 CANCEL
The ACM server gladly stops its neural-network database search and
responds with a 200. The 200 will not travel any further, since P is
the last Via stop.
A->P: SIP/2.0 200 OK
Via: SIP/2.0/UDP sip.ieee.org ;branch=3d8a50dbf5a28d.2
From: A. Bell
To: T. Watson
Call-ID: 31415@c.bell-tel.com
CSeq: 1 CANCEL
In addition, P responds to the original INVITE request with a 487
(Request Terminated):
A->P: SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP sip.ieee.org ;branch=3d8a50dbf5a28d.2
Via: SIP/2.0/UDP c.bell-tel.com
From: A. Bell
To: T. Watson
Call-ID: 31415@c.bell-tel.com
CSeq: 1 INVITE
This response terminates at P.
Bell gets the two 200 responses from X and Y in short order. Bell's
reaction now depends on his software. He can either send an ACK to
both if human intelligence is needed to determine who he wants to
talk to or he can automatically reject one of the two calls. Here, he
acknowledges both, separately and directly to the final destination:
C->X: ACK sip:t.watson@x.bell-tel.com SIP/2.0
Via: SIP/2.0/UDP c.bell-tel.com
From: A. Bell
To: T. Watson ;tag=192137601
Call-ID: 31415@c.bell-tel.com
CSeq: 1 ACK
C->Y: ACK sip:watson@y.bell-tel.com SIP/2.0
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Via: SIP/2.0/UDP c.bell-tel.com
From: A. Bell
To: T. Watson ;tag=35253448
Call-ID: 31415@c.bell-tel.com
CSeq: 1 ACK
After a brief discussion between Bell with X and Y, it becomes clear
that Watson is at X. (Note that this is not a three-way call; only
Bell can talk to X and Y, but X and Y cannot talk to each other.)
Thus, Bell sends a BYE to Y, which is replied to:
C->Y: BYE sip:watson@y.bell-tel.com SIP/2.0
Via: SIP/2.0/UDP c.bell-tel.com
From: A. Bell
To: T. Watson ;tag=35253448
Call-ID: 31415@c.bell-tel.com
CSeq: 2 BYE
Y->C: SIP/2.0 200 OK
Via: SIP/2.0/UDP c.bell-tel.com
From: A. Bell
To: T. Watson ;tag=35253448
Call-ID: 31415@c.bell-tel.com
CSeq: 2 BYE
16.6 Redirects
Replies with status codes 301 (Moved Permanently) or 302 (Moved
Temporarily) specify another location using the Contact field.
Continuing our earlier example, the server P at ieee.org decides to
redirect rather than proxy the request:
P->C: SIP/2.0 302 Moved temporarily
Via: SIP/2.0/UDP c.bell-tel.com
From: A. Bell
To: T. Watson ;tag=72538263
Call-ID: 31415@c.bell-tel.com
CSeq: 1 INVITE
Contact: sip:watson@h.bell-tel.com,
sip:watson@acm.org, sip:t.watson@x.bell-tel.com,
sip:watson@y.bell-tel.com
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As another example, assume Alice (A) wants to delegate her calls to
Bob (B) while she is on vacation until July 29th, 1998. Any calls
meant for her will reach Bob with Alice's To field, indicating to him
what role he is to play. Charlie (C) calls Alice (A), whose server
returns:
A->C: SIP/2.0 302 Moved temporarily
From: Charlie
To: Alice ;tag=2332462
Call-ID: 27182@caller.com
Contact: sip:bob@anywhere.com
Expires: Wed, 29 Jul 1998 9:00:00 GMT
CSeq: 1 INVITE
Charlie then sends the following request to the SIP server of the
anywhere.com domain. Note that the server at anywhere.com forwards
the request to Bob based on the Request-URI.
C->B: INVITE sip:bob@anywhere.com SIP/2.0
From: sip:charlie@caller.com
To: sip:alice@anywhere.com
Call-ID: 27182@caller.com
CSeq: 2 INVITE
Contact: sip:charlie@h.caller.com
In the third redirection example, we assume that all outgoing
requests are directed through a local firewall F at caller.com, with
Charlie again inviting Alice:
C->F: INVITE sip:alice@anywhere.com SIP/2.0
From: sip:charlie@caller.com
To: Alice
Call-ID: 27182@caller.com
CSeq: 1 INVITE
Contact: sip:charlie@h.caller.com
The local firewall at caller.com happens to be overloaded and thus
redirects the call from Charlie to a secondary server S:
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F->C: SIP/2.0 302 Moved temporarily
From: sip:charlie@caller.com
To: Alice
Call-ID: 27182@caller.com
CSeq: 1 INVITE
Contact:
Based on this response, Charlie directs the same invitation to the
secondary server spare.caller.com at port 5080, but maintains the
same Request-URI as before:
C->S: INVITE sip:alice@anywhere.com SIP/2.0
From: sip:charlie@caller.com
To: Alice
Call-ID: 27182@caller.com
CSeq: 2 INVITE
Contact: sip:charlie@h.caller.com
16.7 Negotiation
An example of a 606 (Not Acceptable) response is:
S->C: SIP/2.0 606 Not Acceptable
From: sip:mjh@isi.edu
To: ;tag=7434264
Call-ID: 14142@north.east.isi.edu
CSeq: 1 INVITE
Contact: sip:mjh@north.east.isi.edu
Warning: 370 "Insufficient bandwidth (only have ISDN)",
305 "Incompatible media format",
330 "Multicast not available"
Content-Type: application/sdp
Content-Length: 50
v=0
o=schooler 3149329138 3149329165 IN IP4 38.245.76.2
s=Let's talk
t=3149328630 0
b=CT:128
c=IN IP4 north.east.isi.edu
m=audio 3456 RTP/AVP 5 0 7
a=rtpmap:5 DVI4/8000
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a=rtpmap:0 PCMU/8000
a=rtpmap:7 LPC/8000
m=video 2232 RTP/AVP 31
a=rtpmap:31 H261/90000
In this example, the original request specified a bandwidth that was
higher than the access link could support, requested multicast, and
requested a set of media encodings. The response states that only 128
kb/s is available and that (only) DVI, PCM or LPC audio could be
supported in order of preference.
The response also states that multicast is not available. In such a
case, it might be appropriate to set up a transcoding gateway and
re-invite the user.
16.8 OPTIONS Request
A caller Alice can use an OPTIONS request to find out the
capabilities of a potential callee Bob, without "ringing" the
designated address. Bob returns a description indicating that he is
capable of receiving audio encodings PCM mu-law (RTP payload type 0),
1016 (payload type 1), GSM (payload type 3), and SX7300/8000 (dynamic
payload type 99), and video encodings H.261 (payload type 31) and
H.263 (payload type 34).
C->S: OPTIONS sip:bob@example.com SIP/2.0
From: Alice
To: Bob
Call-ID: 6378@host.anywhere.org
CSeq: 1 OPTIONS
Accept: application/sdp
S->C: SIP/2.0 200 OK
From: Alice
To: Bob ;tag=376364382
Call-ID: 6378@host.anywhere.org
Content-Length: 81
Content-Type: application/sdp
v=0
o=alice 3149329138 3149329165 IN IP4 24.124.37.3
s=Security problems
t=3149328650 0
c=IN IP4 24.124.37.3
m=audio 0 RTP/AVP 0 1 3 99
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a=rtpmap:0 PCMU/8000
a=rtpmap:1 1016/8000
a=rtpmap:3 GSM/8000
a=rtpmap:99 SX7300/8000
m=video 0 RTP/AVP 31 34
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
A Minimal Implementation
A.1 Transport Protocol Support
User agents and stateless proxies MUST support UDP and MAY support
TCP or other transport protocols, stateful proxies MUST support both
UDP and TCP.
A.2 Client
All clients MUST be able to generate the INVITE and ACK requests.
Clients MUST generate and parse the Call-ID, Content-Length,
Content-Type, CSeq, From and To headers. Clients MUST also parse the
Require header. A minimal implementation MUST understand SDP (RFC
2327, [6]). It MUST be able to recognize the status code classes 1
through 6 and act accordingly.
The following capability sets build on top of the minimal
implementation described in the previous paragraph. In general, each
capability listed below builds on the ones above it:
Basic: A basic implementation adds support for the BYE method to
allow the interruption of a pending call attempt. It
includes a User-Agent header in its requests and indicates
its preferred language in the Accept-Language header.
Redirection: To support call forwarding, a client needs to be
able to understand the Contact header, but only the SIP-URL
part, not the parameters.
Firewall-friendly: A firewall-friendly client understands the
Route and Record-Route header fields and can be configured
to use a local proxy for all outgoing requests.
Negotiation: A client MUST be able to request the OPTIONS method
and understand the 380 (Alternative Service) status and the
Contact parameters to participate in terminal and media
negotiation. It SHOULD be able to parse the Warning
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response header to provide useful feedback to the caller.
Authentication: If a client wishes to invite callees that
require caller authentication, it MUST be able to recognize
the 401 (Unauthorized) status code, MUST be able to
generate the Authorization request header and MUST
understand the WWW-Authenticate response header.
If a client wishes to use proxies that require caller
authentication, it MUST be able to recognize the 407 (Proxy
Authentication Required) status code, MUST be able to
generate the Proxy-Authorization request header and
understand the Proxy-Authenticate response header.
A.3 Server
A minimally compliant server implementation MUST understand the
INVITE, ACK, OPTIONS and BYE requests. A proxy server MUST also
understand CANCEL. It MUST parse and generate, as appropriate, the
Call-ID, Content-Length, Content-Type, CSeq, Expires, From, Max-
Forwards, Require, To and Via headers. It MUST echo the CSeq and
Timestamp headers in the response. It SHOULD include the Server
header in its responses.
A.4 Header Processing
Table 6 lists the headers that different implementations support. UAC
refers to a user-agent client (calling user agent), UAS to a user-
agent server (called user-agent).
The fields in the table have the following meaning. Type is as in
Table 4 and 5. "-" indicates the field is not meaningful to this
system (although it might be generated by it). "m" indicates the
field MUST be understood. "b" indicates the field SHOULD be
understood by a Basic implementation. "r" indicates the field SHOULD
be understood if the system claims to understand redirection. "a"
indicates the field SHOULD be understood if the system claims to
support authentication. "e" indicates the field SHOULD be understood
if the system claims to support encryption. "o" indicates support of
the field is purely optional. Headers whose support is optional for
all implementations are not shown.
B Usage of the Session Description Protocol (SDP)
This section describes the use of the Session Description Protocol
(SDP) (RFC 2327 [6]). SDP is identified as Content-Type
"application/sdp". Each SIP message body may only contain one SDP
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type UAC proxy UAS registrar
_____________________________________________________
Accept R - o m m
Accept-Encoding R - - m m
Accept-Language R - b b b
Allow 405 o - - -
Authorization R a o a a
Call-ID g m m m m
Content-Encoding g m - m m
Content-Length g m m m m
Content-Type g m - m m
CSeq g m m m m
Encryption g e - e e
Expires g - o o m
From g m o m m
Hide R - m - -
Contact R - - - m
Contact r r r - -
Max-Forwards R - b - -
Proxy-Authenticate 407 a - - -
Proxy-Authorization R - a - -
Proxy-Require R - m - -
Require R m - m m
Response-Key R - - e e
Route R - m - -
Timestamp g o o m m
To g m m m m
Unsupported r b b - -
User-Agent g b - b -
Via g m m m m
WWW-Authenticate 401 a - - -
Table 6: Header Field Processing Requirements
message unless the SIP message body is of type "multipart". SDP
messages in later SIP request supersede earlier ones for the same
call leg.
B.1 Configuring Media Streams
The caller and callee align their media descriptions so that the nth
media stream ("m=" line) in the caller's session description
corresponds to the nth media stream in the callee's description.
All media descriptions SHOULD contain "a=rtpmap" mappings from RTP
payload types to encodings.
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This allows easier migration away from static payload
types.
User agents MUST NOT reuse dynamic payload types for different
encodings when issuing a re-INVITE.
If the callee wants to neither send nor receive a stream offered by
the caller, the callee sets the port number of that stream to zero in
its media description.
There currently is no other way than port zero for the
callee to refuse a bidirectional stream offered by the
caller. Both caller and callee need to be aware what media
tools are to be started.
For example, assume that the caller Alice has included the following
description in her INVITE request. It includes a bidirectional audio
stream and two bidirectional video streams, using H.261 (payload type
31) and MPEG (payload type 32).
v=0
o=alice 2890844526 2890844526 IN IP4 host.anywhere.com
s=New board design
t=0 0
c=IN IP4 host.anywhere.com
m=audio 49170 RTP/AVP 0
a=rtpmap:0 PCMU/8000
m=video 51372 RTP/AVP 31
a=rtpmap:31 H261/90000
m=video 53000 RTP/AVP 32
a=rtpmap:32 MPV/90000
The callee, Bob, does not want to receive or send the first video
stream, so it returns the media description below:
v=0
o=bob 2890844730 2890844730 IN IP4 host.example.com
s=New board design
t=0 0
c=IN IP4 host.example.com
m=audio 47920 RTP/AVP 0 1
a=rtpmap:0 PCMU/8000
m=video 0 RTP/AVP 31
m=video 53000 RTP/AVP 32
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a=rtpmap:32 MPV/90000
B.2 Setting SDP Values for Unicast
If a session description from a caller contains a media stream which
is listed as send (receive) only, it means that the caller is only
willing to send (receive) this stream, not receive (send). The same
is true for the callee.
For receive-only and send-or-receive streams, the port number and
address in the session description indicate where the media stream
should be sent to by the recipient of the session description, either
caller or callee. For send-only RTP streams, the address and port
number indicate where RTCP reports are to be sent. (RTCP reports are
sent to the port number one higher than the number indicated.)
The list of payload types for each media stream conveys two pieces of
information, namely the set of codecs that the caller or callee is
capable of sending or receiving, and the RTP payload type numbers
used to identify those codecs. For receive-only or send-and-receive
media streams, a caller SHOULD list all of the codecs it is capable
of supporting in the session description in an INVITE or ACK. For
send-only streams, the caller SHOULD indicate only those it wishes to
send for this session. For receive-only streams, the payload type
numbers indicate the value of the payload type field in RTP packets
the caller is expecting to receive for that codec type. For send-only
streams, the payload type numbers indicate the value of the payload
type field in RTP packets the caller is planning to send for that
codec type. For send-and-receive streams, the payload type numbers
indicate the value of the payload type field the caller expects to
both send and receive.
If a media stream is listed as receive-only by the caller, the callee
lists, in the response, those codecs it intends to use from among the
ones listed in the request. If a media stream is listed as send-only
by the caller, the callee lists, in the response, those codecs it is
willing to receive among the ones listed in the the request. If the
media stream is listed as both send and receive, the callee lists
those codecs it is capable of sending or receiving among the ones
listed by the caller in the INVITE. The actual payload type numbers
in the callee's session description corresponding to a particular
codec MUST be the same as the caller's session description.
If caller and callee have no media formats in common for a particular
stream, the callee MUST return a session description containing the
particular "m=" line, but with the port number set to zero, and no
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payload types listed.
If there are no media formats in common for all streams, the callee
SHOULD return a 488 or 606 response, with a 304 Warning header field.
B.3 Multicast Operation
The interpretation of send-only and receive-only for multicast media
sessions differs from that for unicast sessions. For multicast,
send-only means that the recipient of the session description (caller
or callee) SHOULD only send media streams to the address and port
indicated. Receive-only means that the recipient of the session
description SHOULD only receive media on the address and port
indicated.
For multicast, receive and send multicast addresses are the same and
all parties use the same port numbers to receive media data. If the
session description provided by the caller is acceptable to the
callee, the callee can choose not to include a session description or
MAY echo the description in the response.
A callee MAY, in the response, return a session description with some
of the payload types removed, or port numbers set to zero (but no
other value). This indicates to the caller that the callee does not
support the given stream or media types which were removed. A callee
MUST NOT change whether a given stream is send-only, receive-only, or
send-and-receive.
If a callee does not support multicast at all, it SHOULD return a 400
status response and include a 330 Warning.
B.4 Delayed Media Streams
In some cases, a caller may not know the set of media formats which
it can support at the time it would like to issue an invitation. This
is the case when the caller is actually a gateway to another protocol
which performs media format negotiation after call setup. When this
occurs, a caller MAY issue an INVITE with a session description that
contains no media lines. The callee SHOULD interpret this to mean
that the caller wishes to participate in a multimedia session
described by the session description, but that the media streams are
not yet known. The callee SHOULD return a session description
indicating the streams and media formats it is willing to support,
however. The caller MAY update the session description either in the
ACK request or in a re-INVITE at a later time, once the streams are
known.
B.5 Adding and Deleting Media Streams
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To add a stream to an existing call leg, either party appends an
additional "m" line to the previous session description when sending
a re-INVITE.
To remove a stream from a call leg, either party sets its port to
zero in the session description when sending a re-INVITE.
UAs receiving SDP SHOULD accept descriptions that are not aligned, in
terms of "m=" lines, with earlier descriptions. If a UA receives such
a description, it SHOULD line up "m=" lines by media type ("audio",
"video", ...). If it receives a new description in an INVITE request,
it MAY line up the "m=" lines by media type, address and port and
position in the description and start or delete media streams
accordingly.
Receiving a new description in an INVITE request that omits
media with zero ports or is otherwise different may occur
if the UA crashed and restarted.
B.6 Putting Media Streams on Hold
If a party in a call wants to put the other party "on hold", i.e.,
request that it temporarily stops sending one or more media streams,
a party re-invites the other by sending an INVITE request with a
modified session description. The session description is the same as
in the original invitation (or response), but the "c" destination
addresses for the media streams to be put on hold are set to zero
(0.0.0.0).
B.7 Subject and SDP "s=" Line
The SDP "s=" line and the SIP Subject header field have different
meanings when inviting to a multicast session. The session
description line describes the subject of the multicast session,
while the SIP Subject header field describes the reason for the
invitation. The example in Section 16.2 illustrates this point. For
invitations to two-party sessions, the SDP "s=" line MAY consist of a
single space character (0x20).
Unfortunately, SDP does not allow to leave the "s=" line
empty.
B.8 The SDP "o=" Line
The "o=" line MUST be present for all sessions, including two-party
sessions, to allow re-use of SDP-based tools.
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C Summary of Augmented BNF
All of the mechanisms specified in this document are described in
both prose and an augmented Backus-Naur Form (BNF) similar to that
used by RFC 822 [28] and RFC 2234 [47]. Implementors will need to be
familiar with the notation in order to understand this specification.
The augmented BNF includes the following constructs:
name = definition
The name of a rule is simply the name itself (without any enclosing
"<" and ">") and is separated from its definition by the equal "="
character. White space is only significant in that indentation of
continuation lines is used to indicate a rule definition that spans
more than one line. Certain basic rules are in uppercase, such as SP,
LWS, HT, CRLF, DIGIT, ALPHA, etc. Angle brackets are used within
definitions whenever their presence will facilitate discerning the
use of rule names.
"literal"
Quotation marks surround literal text. Unless stated otherwise, the
text is case-insensitive.
rule1 | rule2
Elements separated by a bar ("|") are alternatives, e.g., "yes | no"
will accept yes or no.
(rule1 rule2)
Elements enclosed in parentheses are treated as a single element.
Thus, "(elem (foo | bar) elem)" allows the token sequences "elem foo
elem" and "elem bar elem".
*rule
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The character "*" preceding an element indicates repetition. The full
form is "*element" indicating at least and at most
occurrences of element. Default values are 0 and infinity so that
"*(element)" allows any number, including zero; "1*element" requires
at least one; and "1*2element" allows one or two.
[rule]
Square brackets enclose optional elements; "[foo bar]" is equivalent
to "*1(foo bar)".
N rule
Specific repetition: "(element)" is equivalent to
"*(element)"; that is, exactly occurrences of (element).
Thus 2DIGIT is a 2-digit number, and 3ALPHA is a string of three
alphabetic characters.
#rule
A construct "#" is defined, similar to "*", for defining lists of
elements. The full form is "# element" indicating at least
and at most elements, each separated by one or more commas (",")
and OPTIONAL linear white space (LWS). This makes the usual form of
lists very easy; a rule such as
( *LWS element *( *LWS "," *LWS element ))
can be shown as 1# element. Wherever this construct is used, null
elements are allowed, but do not contribute to the count of elements
present. That is, "(element), , (element)" is permitted, but counts
as only two elements. Therefore, where at least one element is
required, at least one non-null element MUST be present. Default
values are 0 and infinity so that "#element" allows any number,
including zero; "1#element" requires at least one; and "1#2element"
allows one or two.
; comment
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A semi-colon, set off some distance to the right of rule text, starts
a comment that continues to the end of line. This is a simple way of
including useful notes in parallel with the specifications.
implied *LWS
The grammar described by this specification is word-based. Except
where noted otherwise, linear white space (LWS) can be included
between any two adjacent words (token or quoted-string), and between
adjacent tokens and separators, without changing the interpretation
of a field. At least one delimiter (LWS and/or separators) MUST exist
between any two tokens (for the definition of "token" below), since
they would otherwise be interpreted as a single token.
C.1 Basic Rules
The following rules are used throughout this specification to
describe basic parsing constructs. The US-ASCII coded character set
is defined by ANSI X3.4-1986.
OCTET = %x00-ff ; any 8-bit sequence of data
CHAR = %x00-7f ; any US-ASCII character (octets 0 - 127)
upalpha = "A" | "B" | "C" | "D" | "E" | "F" | "G" | "H" | "I" |
"J" | "K" | "L" | "M" | "N" | "O" | "P" | "Q" | "R" |
"S" | "T" | "U" | "V" | "W" | "X" | "Y" | "Z"
lowalpha = "a" | "b" | "c" | "d" | "e" | "f" | "g" | "h" | "i" |
"j" | "k" | "l" | "m" | "n" | "o" | "p" | "q" | "r" |
"s" | "t" | "u" | "v" | "w" | "x" | "y" | "z"
alpha = lowalpha | upalpha
DIGIT = "0" | "1" | "2" | "3" | "4" | "5" | "6" | "7" |
"8" | "9"
alphanum = alpha | DIGIT
CTL = %x00-1f | 0x7f ; (octets 0 -- 31) and DEL (127)
CR = %d13 ; US-ASCII CR, carriage return character
LF = %d10 ; US-ASCII LF, line feed character
SP = %d32 ; US-ASCII SP, space character
HT = %d09 ; US-ASCII HT, horizontal tab character
CRLF = CR LF ; typically the end of a line
The following are defined in RFC 2396 [12] for the SIP URI:
unreserved = alphanum | mark
mark = "-" | "_" | "." | "!" | "~" | "*" | "'"
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| "(" | ")"
escaped = "%" hex hex
SIP header field values can be folded onto multiple lines if the
continuation line begins with a space or horizontal tab. All linear
white space, including folding, has the same semantics as SP. A
recipient MAY replace any linear white space with a single SP before
interpreting the field value or forwarding the message downstream.
LWS = [CRLF] 1*( SP | HT ) ; linear whitespace
The TEXT-UTF8 rule is only used for descriptive field contents and
values that are not intended to be interpreted by the message parser.
Words of *TEXT-UTF8 contain characters from the UTF-8 character set
(RFC 2279 [25]). The TEXT-UTF8-TRIM rule is used for descriptive
field contents that are not quoted strings, where leading and
trailing LWS is not meaningful. In this regard, SIP differs from
HTTP, which uses the ISO 8859-1 character set.
TEXT-UTF8 = *(TEXT-UTF8char | LWS)
TEXT-UTF8-TRIM = *TEXT-UTF8char *(*LWS TEXT-UTF8char)
TEXT-UTF8char = %x21-7e
| UTF8-NONASCII
UTF8-NONASCII = %xc0-df 1UTF8-CONT
| %xe0-ef 2UTF8-CONT
| %xf0-f7 3UTF8-CONT
| %xf8-fb 4UTF8-CONT
| %xfc-fd 5UTF8-CONT
UTF8-CONT = %x80-bf
A CRLF is allowed in the definition of TEXT-UTF8 only as part of a
header field continuation. It is expected that the folding LWS will
be replaced with a single SP before interpretation of the TEXT-UTF8
value.
Hexadecimal numeric characters are used in several protocol elements.
HEX = "A" | "B" | "C" | "D" | "E" | "F"
| "a" | "b" | "c" | "d" | "e" | "f" | DIGIT
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Many SIP header field values consist of words separated by LWS or
special characters. These special characters MUST be in a quoted
string to be used within a parameter value.
token = 1*(alphanum | "-" | "." | "!" | "%" | "*"
| "_" | "+" | "`" | "'" | "~" )
separators = "(" | ")" | "<" | ">" | "@" |
"," | ";" | ":" | "\" | <"> |
"/" | "[" | "]" | "?" | "=" |
"{" | "}" | SP | HT
Comments can be included in some SIP header fields by surrounding the
comment text with parentheses. Comments are only allowed in fields
containing "comment" as part of their field value definition. In all
other fields, parentheses are considered part of the field value.
comment = "(" *(ctext | quoted-pair | comment) ")"
ctext = < any TEXT-UTF8 excluding "(" and ")">
A string of text is parsed as a single word if it is quoted using
double-quote marks. In quoted strings, quotation marks (") and
backslashes (
) need to be escaped.
quoted-string = ( <"> *(qdtext | quoted-pair ) <"> )
qdtext = LWS | %x21 | %x23-5b | %x5d-7e
| UTF8-NONASCII
The backslash character ("\") MAY be used as a single-character
quoting mechanism only within quoted-string and comment constructs.
quoted-pair = " \ " CHAR
D IANA Considerations
Section 4.4 describes a name space and mechanism for registering SIP
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options.
Section 6.48 describes the name space for registering SIP warn-codes.
E Changes from RFC 2543
In addition to editorial clarifications, this document changes or
adds the following features to SIP as specified in RFC 2543:
o The Record-Route header field needs to be added to each
request and applies only to the next request.
o Extensions developed by the IETF no longer use the org.ietf
prefix.
o tag syntax
o A new optional header field, In-Reply-To (Section 6.27.
F Changes Made in Version 00
o In Sec. 10.5.1, indicated that UAC should send both CANCEL and
BYE after a retransmission fails.
o Added semicolon and question mark to the list of unreserved
characters for the user part of SIP URLs to handle tel: URLs
properly.
o Uniform handling of if hop count Max-Forwards: return 483.
Note that this differs from HTTP/1.1 behavior, where only
OPTIONS and TRACE allow this header, but respond as the final
recipient when the value reaches zero.
o Clarified that a forking proxy sends ACKs only for INVITE
requests.
o Clarified wording of DNS caching. Added paragraph on "negative
caching", i.e., what to do if one of the hosts failed. It is
probably not a good idea to simply drop this host from the
list if the DNS ttl value is more than a few minutes, since
that would mean that load balancing may not work for quite a
while after a server is brought back on line. This will be
true in particular if a server group receives a large number
of requests from a small number of upstream servers, as is
likely to be the case for calls between major consumer ISPs.
However, without getting into arbitrary and complicated retry
rules, it seems hard to specify any general algorithm. Might
it be worthwhile to simply limit the "black list" interval to
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a few minutes?
o Added optional From-Info and Alert-Info header fields that
describe the caller and information to be used in alerting.
(Currently, avoided use of "purpose" qualification since it is
not yet clear whether rendering content without understanding
its meaning is always appropriate. For example, if a UAS does
not understand that this header is to replace ringing, it
would mix both local ring tone and the indicated sound URL.)
TBD!
o SDP "s=" lines can't be empty, unfortunately. (Section B.7)
o Noted that maddr could also contain a unicast address, but
SHOULD contain the multicast address if the request is sent
via multicast (Section 6.47, 10.2.2).
o Clarified that responses are sent to port in Via sent-by
value.
o Added "other-*" to the user URL parameter and the Hide and
Content-Disposition headers.
o Clarify generation of timeout (408) responses in forking
proxies and mention the Expires header. (Section 12.4)
o Clarified that CANCEL and INVITE are separate transactions
(Fig. 12). Thus, the INVITE request generates a 487
(Transaction Terminated) if a CANCEL or BYE arrives.
o Clarified that Record-Route SHOULD be inserted in every
request, but that the route, once established, persists. This
provides robustness if the called UAS crashes.
o Emphasized that proxy, redirect, registrar and location
servers are logical, not physical entities and that UAC and
UAS roles are defined on a request-by-request basis. (Section
1.3)
o In Section 6.47, noted that the maddr and received parameters
also need to be encrypted when doing Via hiding.
o Simplified Fig. 12 to only show INVITE transaction.
o Added definition of the use of Contact (Section 6.15) for
OPTIONS.
o Added HTTP/RFC822 headers Content-Language and MIME-Version.
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o Added note in Section A indicating that UAs need to support
UDP.
o Added explanation in Section 11.5 explaining what a UA should
do when receiving an initial INVITE with a tag.
o Clarified UA and proxy behavior for 302 responses (Section
7.3.3).
o Added details on what a UAS should do when receiving a tagged
INVITE request for an unknown call leg. This could occur if
the UAS had crashed and the UAC sends a re-INVITE or if the
BYE got lost and the UAC still believes to be in the call.
o Added definition of Contact in 4xx, 5xx and 6xx to "redirect"
to more error details.
o Added note to forking proxy description in Section 12.4 to
gather *-Authenticate from responses. This allows several
branches to be authenticated simultaneously.
o Changed URI syntax to use URL escaping instead of quotation
marks.
o Changed SIP URL definition to reference RFC 2806 for
telephone-subscriber part.
o Clarify that the To URI should basically be ignored by the
receiving UAS except for matching requests to call legs. In
particular, To headers with a scheme or name unknown to the
callee should be accepted.
o Clarify in Section 6.47.1 that maddr is to be added by any
client, either proxy or UAC.
o Added response code 488 to indicate that there was no common
media at the particular destination. (606 indicates such
failure globally.)
o In Section 6.24, noted that registration updates can shorten
the validity period.
o Added note to Section 14.3 to enclose the URI in quotation
marks. The BNF in RFC 2617 is in error.
o Clarified that registrars use Authorization and WWW-
Authenticate, not proxy authentication.
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o Added note in Section 6.15 that "headers" are copied from
Contact into the new request.
o Changed URL syntax so that port specifications have to have at
least one digit, in line with other URL formats such as
"http". Previously, an empty port number was permissible.
o In Section B, added a section on how to add and delete streams
in re-INVITEs.
o IETF-blessed extensions now have short names, without
org.ietf. prefix.
o Cseq is unique within a call leg, not just within a call
(Section 6.21).
o Added IPv6 literal addresses to the SIP URL definition in
Section 2, according to RFC 2732 [48]. Modified the IPv4
address to limit segments to at most three digits.
o In Section 4.2.6, modify registration procedure so that it
explicitly references the URL comparison. Updates with shorter
expiration time are now allowed.
o For send-only media, SDP still must indicate the address and
port, since these are needed as destinations for RTCP
messages. (Section B)
o Changed references regarding DNS SRV records from RFC 2052 to
RFC 2782, which is now a Proposed Standard. Integrated SRV
into the search procedure in Section 1 and removed the SRV
appendix. The only visible change is that protocol and service
names are now prefixed by an underscore. Added wording that
incorporates the precedence of maddr.
o Allow parameters in Record-Route and Route headers.
o In Table 2, list udp as the default value for the transport
parameter in SIP URI.
o Removed sentence that From can be encrypted. It cannot, since
the header is needed for call-leg identification.
o Added note that a UAC only copies a To tag into subsequent
transactions if it arrives in a 200 OK to an INVITE in Section
11. This avoids the problem that occurs when requests get
resubmitted after receiving, say, a 407 (or possibly 500, 503,
504, 305, 400, 411, 413, maybe even 408). Under the old rules,
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these requests would have a tag, which would force the called
UAS to reject the request, since it doesn't have an entry for
this tag.
o Loop detection has been modified to take the request-URI into
account (Section 12.3 and 6.47.5). This allows the same
request to visit the server twice, but with different request
URIs.
o Elaborated on URL comparison and comparison of From/To fields.
o Added np-queried user parameter.
o Changed tag syntax from UUID to token, since there's no reason
to restrict it to hex.
o Added Content-Disposition header based on earlier discussions
about labeling what to do with a message body (part).
o Clarification: proxies must insert To tags for locally
generated responses.
o Clarification: multicast may be used for subsequent
registrations.
o Feature: Added Supported header. Needed if client wants to
indicate things the server can usefully return in the
response.
o Bug: The From, To, and Via headers were missing extension
parameters. The Encryption and Response-Key header fields now
"officially" allow parameters consisting only of a token,
rather than just "token = value".
o Bug: Allow was listed as optional in 405 responses in Table 4.
It is mandatory.
o Added in Section 4.2.4: "A BYE request from either called or
calling party terminates any pending INVITE, but the INVITE
request transaction MUST be completed with a final response."
o Clarified in Section 4.2.1: "If an INVITE request for an
existing session fails, the session description agreed upon in
the last successful INVITE transaction remains in force."
o Clarified in Section 4.2.1 what happens if two INVITE requests
meet each other on the wire, either traveling the same or in
opposite directions:
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A UAC MUST NOT issue another INVITE request for the
same call leg before the previous transaction has
completed. A UAS that receives an INVITE before it
sent the final response to an INVITE with a lower CSeq
number MUST return a 400 (Bad Request) response and
MUST include a Retry-After header field with a
randomly chosen value of between 0 and 10 seconds. A
UA that receives an INVITE while it has an INVITE
transaction pending, returns a 500 (Internal Server
Error) and also includes a Retry-After header field.
o Expires header clarified: limits only duration of INVITE
transaction, not the actual session. SDP does the latter.
o The In-Reply-To header was added (Section 6.27).
o There were two incompatible BNFs for WWW-Authenticate. One
defined for PGP, and the other borrowed from HTTP. For basic
or digest:
WWW-Authenticate: basic realm="Wallyworld"
and for pgp:
WWW-Authenticate: pgp; realm="Wallyworld"
The latter is incorrect and the semicolon has been removed.
o Added rules for Route construction from called to calling UA.
o We now allow Accept and Accept-Encoding in BYE and CANCEL
requests. There is no particular reason not to allow them, as
both requests could theoretically return responses,
particularly when interworking with other signaling systems.
o PGP "pgp-pubalgorithm" allows server to request the desired
public-key algorithm.
o ABNF rules now describe tokens explicitly rather than by
subtraction; explicit character enumeration for CTL, etc.
o Registrars should be careful to check the Date header as the
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expiration time may well be in the past, as seen by the
client.
o Content-Length is mandatory; Table 4 erroneously marked it as
optional.
o User-Agent was classified in a syntax definition as a request
header rather than a general header.
o Clarify ordering of items to be signed and include realm in
list.
o Allow Record-Route in 401 and 484 responses.
o Hop-by-hop need to preceded end-to-end headers only if
authentication is used (Section 6).
o 1xx message bodies MAY now contain session descriptions.
o Changed references to HTTP/1.1 and authentication to point to
the latest RFCs.
o Added 487 (Request terminated) status response. It is issued
if the original request was terminated via CANCEL or BYE.
o The spec was not clear on the identification of a call leg.
Section 1.3 says it's the combination of To, From, and Call-
ID. However, requests from the callee to the caller have the
To and From reversed, so this definition is not quite
accurate. Additionally, the "tag" field should be included in
the definition of call leg. The spec now says that a call leg
is defined as the combination of local-address, remote-
address, and call-id, where these addresses include tags.
Text was added to Section 6.21 to emphasize that the From and
To headers designate the originator of the request, not that
of the call leg.
G Changes To Be Made
NOTE: Almost all of these require further discussion or may already
have been integrated into the main spec.
o Bug: All URI parameters, except method, are allowed in a
Request-URI. Consequently, also updated the description of
which parameters are copied from 3xx responses in Sec. 6.15.
o Bug: Consider a proxy that implements an ACD service. It
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proxies call requests to operators based on some complex
logic. After receiving and INVITE, proxying it, and forwarding
the response, the proxy eventually times out its state. Later
on, an ACK (a late one, or one that has been wandering in the
network) arrives. Where should the proxy send this? Ideally,
the proxy should know where it sent the INVITE, and send it to
the same place. This is not workable for complex forwarding
logic. To fix this, the ACKs should go directly to the UAS.
This will occur if a UAS inserts a Contact into the 200 OK,
and if the UAC honors this in subsequent requests (including
ACK). However, insertion of Contact, and honoring of Contact,
are SHOULD, not MUST, so this "lost ACK" case is still
possible in conditionally compliant implementations. TO fix
this, (1) insertion of a Contact header, and (2) honoring of a
Contact header, should be made MUST strength. (Q: is this the
only solution???)
o Bug: The use of CRLF,CR,or LF to terminate lines is confusing.
Basically, each header line can be terminated by a CR, LF, or
CRLF. Furthermore, the end of the headers is signified by a
"double return". Thus, if the BNF is defined as:
return = (CR | LF | CRLF)
and double-return is (CRLF CRLF)
This grammar is not context-free, since CR LF could be either
return or double return, and thus you may not be able to
distinguish the end of line with end of headers. In fact, the
header section should be terminated by a double-return of the
form: double-return = (CR CR) | (LF LF) | (CR LF CR LF)
This is inconsistent with the BNF currently defined in the
spec, which reads like the former definition. This needs to be
explicitly clarified in the spec.
o Bug: Section 4.2.6 specifies that, for a REGISTER request, the
user name of the Request-URI MUST be empty. While it does not
mention anything about the "@" sign, the SIP-URL BNF is
defined so that an "@" sign is used only in conjunction with a
user ID. The implication that no "@" sign appears in a
REGISTER Request-URI is never explicitly stated.
o Bug: Round brackets in Contact header are sort of legacy, and
very hard to implement. They are also not that useful. Perhaps
they should be removed. Vote during bakeoff was that this is
OK.
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o Bug: The spec says that a proxy is a back to back UAS/UAC.
This is almost, but not quite, true. For example, a UAS should
insert a tag into a provisional response, but a proxy should
not. This should be clarified.
o Issue: Usage of Content-Length. Spec is not that clear under
what conditions its needed, particularly with TCP. This should
be clarified. Another issue: should Content-Length just be
made mandatory? It's not as a carryover from HTTP; do we have
the same assumptions they do? Also, the section on Content-
Length does not discuss behavior for requests over TCP that do
not contain Content-Lengths (Answer: if no Content-Length is
present in requests, the request packet must be terminated
with a close connection. However, having the client close a
connection is the same as sending a CANCEL, so no response is
ever received. Now, a client *can* do this, but it makes no
sense)
o Issue: In REGISTER requests, should the request URI be a fully
qualified hostname, or the domainname:
REGISTER sip:machine.company.com SIP/2.0
or
REGISTER sip:company.com SIP/2.0
It matters since registrars check if the domain name is
there's before accepting the registration. The comparison
might fail if the wrong thing is sent. Answer: probably a
proxy/UAC SHOULD send it to company.com (and then use SRV to
find the right machine), but a registrar SHOULD accept both
fully qualified host names and domain names in comparisons. In
fact, the registrar probably should be making this comparison
on the To field, not the Request-URI!
o Issue: A sends an INVITE to B, and B returns a 180 with a tag.
Then, A sends BYE to B, without a tag in the To field. Should
B accept this BYE or not? Answer: not sure; I think actually
it says somewhere in the spec you are not supposed to initiate
a new transaction until the previous completes.
o Bug: No default value for the q parameter in Contact is
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defined. This is not strictly needed, but is useful for
consistent behaviors at recursive proxies and at UAC's
o Bug: A INVITEs B through a proxy, B responds to A with a tag
in the To field. Later, A sends a re-INVITE to B, with a tag
in the To field, through the proxy. This request is malformed,
so the proxy returns a 400 error. Normally, a proxy will
insert a tag into the To field of a 400 response. But here, it
cannot, since a tag is already present in the request. Thus,
the rule should be that if a request arrives with a tag, the
entity sending a response MUST NOT insert an additional tag,
even if it a different entity than the one that created the
tag in the first place.
o Bug: When UAS mirrors the To field in the response, and the
UAC "matches" it to determine what call and call-leg it is, it
MUST make the comparison based on the user, host, and tag of
the URI only - not based on the display name, existence of
brackets, etc. (Basically, don't do a pure string match).
o Bug: When multicasting a register, if a server wishes to
authenticate the user it, must return a 401. However, it is
not permitted to send a 400 response via multicast. How should
this be handled?
o Bug: If a proxy sends a request by UDP (TCP), the spec does
not disallow placing TCP (UDP) in the transport parameter of
the Via field, which it should.
o Bug: The spec says that if a server receives a CANCEL for
which it has no pending transaction, it returns an error.
This, however, is not true for a proxy. A stateless proxy
SHOULD forward the CANCEL as it would any non-INVITE request.
o Bug: Currently, the spec says a proxy should not insert a
Record-Route into a request with Route headers. I think this
is wrong. The Route should be refreshed with each new request
by a new Record-Route. The primary motivation here is
mobility; if a user moves and sends a new request with a new
Contact header, the Route must change. Similarly, this new
route may require new proxies on the route (or old ones to
drop off).
o Section 6.13 in the rfc begins mid-paragraph after the BNF.
The following text seems to have been misplaced in the
conversion to ASCII:
Even if the "display-name" is empty, the "name-addr"
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Internet Draft SIP July 13, 2000
form MUST be used if the "addr-spec" contains a comma,
semicolon or question mark.
o Table 5 lists WWW-Authenticate as a response-header allowed
only in 401 responses. However, section 13.2 allows its reuse
in requests containing Require: signed-response header. Also
examples in section 14.1 have WWW-Authenticate in INVITE
requests with no Require: present.
A client MAY require that a server sign its response by
including a Require: signed-response request header field.
The client indicates the desired authentication method via the
WWW- Authenticate header.
This is also an inconsistency. If we do allow this, then WWW-
Authenticate must be a request and response header field, and
the Authorization header as well, since then it is included in
a signed response.
o Text on including Authorization in responses and WWW-
Authenticate in requests needs to be added.
o Ping liveness detection. Motivation:
A proxy may be keeping track of a call during its
entire duration, for example to maintain consistent
routing for all subsequent requests. The proxy
enforces that requests traverse it by using Record-
Route. If, however, a BYE fails to arrive at the
termination of a call (e.g., because of application
failure or network partition), the proxy must know to
delete the call record. If the proxy deletes a call
record before a session is terminated (by timeout, for
example), it makes the routing of subsequent call
messages impossible (since the information regarding
the ultimate destination of an INVITE may no longer be
available). Without the ability for a proxy server to
check the liveness of a session, there is no way to
rectify this problem.
Solutions suggested:
o proxy sends re-INVITE;
o session lifetime;
o A PING request would be retransmitted (T1, doubled each time,
max of T2, up to 11 times) until a provisional or final
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response is received. Once that is accomplished, it is only
retransmitted at the frequency specified in the "Period"
header. As long as another provisional response is received in
at the freqency specified in the "Period" header, the PING is
not retransmitted.
o The OPTIONS request is answered with a special 2xx status if
the call leg exists at the server. (What about changes in
SDP?)
o received-port for NAPT support in Via header field;
o Proxy authentication: If a forking proxy adds or changes an
Authorization header, it MUST change the branch parameter in
the Via header to avoid being confused by retransmission of
earlier 407 responses. If a request triggered a 407 response,
the downstream proxy inspects retransmissions of the the same
request to see if it contains new authorization header. If so,
it re-evaluates its authentication decision. This is a slight
deviation from the usual idempotency rules.
H Acknowledgments
We wish to thank the members of the IETF MMUSIC and SIP WGs for their
comments and suggestions. Detailed comments were provided by Jim
Buller, Neil Deason, Dave Devanathan, Cédric Fluckiger, Yaron Goland,
Bernie Höneisen, Phil Hoffer, Christian Huitema, Jean Jervis, Gadi
Karmi, Anders Kristensen, Jonathan Lennox, Gethin Liddell, Keith
Moore, Vern Paxson, Moshe J. Sambol, Chip Sharp, Igor Slepchin,
Robert Sparks, Eric Tremblay., and Rick Workman.
This work is based, inter alia, on [49,50].
I Authors' Addresses
Mark Handley
ACIRI
electronic mail: mjh@aciri.org
Henning Schulzrinne
Dept. of Computer Science
Columbia University
1214 Amsterdam Avenue
New York, NY 10027
USA
electronic mail: schulzrinne@cs.columbia.edu
Eve Schooler
Handley/Schulzrinne/Schooler/Rosenberg [Page 168]
Internet Draft SIP July 13, 2000
Computer Science Department 256-80
California Institute of Technology
Pasadena, CA 91125
USA
electronic mail: schooler@cs.caltech.edu
Jonathan Rosenberg
dynamicsoft
72 Eagle Rock Ave
East Hanover, NJ 07936
USA
electronic mail: jdrosen@dynamicsoft.com
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Internet Draft SIP July 13, 2000
packetization and synchronization on non-guaranteed quality of
service LANs," Recommendation H.225.0, Telecommunication
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locators (URL)," Request for Comments 1738, Internet Engineering Task
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services," Request for Comments 2219, Internet Engineering Task
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[16] P. V. Mockapetris, "Domain names - implementation and
specification," Request for Comments 1035, Internet Engineering Task
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[17] D. Zimmerman, "The finger user information protocol," Request
for Comments 1288, Internet Engineering Task Force, Dec. 1991.
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Computer Science, California Institute of Technology, Pasadena,
California, Aug. 1996.
Handley/Schulzrinne/Schooler/Rosenberg [Page 170]
Internet Draft SIP July 13, 2000
[21] A. Vaha-Sipila, "URLs for telephone calls," Request for Comments
2806, Internet Engineering Task Force, Apr. 2000.
[22] J. Postel, "User datagram protocol," Request for Comments 768,
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[29] D. Meyer, "Administratively scoped IP multicast," Request for
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scheme," Request for Comments 2368, Internet Engineering Task Force,
July 1998.
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[35] R. Troost and S. Dorner, "Communicating presentation information
in internet messages: The content-disposition header," Request for
Comments 1806, Internet Engineering Task Force, June 1995.
[36] R. T. Braden, "Requirements for internet hosts - application and
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Oct. 1989.
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Comments 1321, Internet Engineering Task Force, Apr. 1992.
[39] H. Alvestrand, "IETF policy on character sets and languages,"
Request for Comments 2277, Internet Engineering Task Force, Jan.
1998.
[40] M. Elkins, "MIME security with pretty good privacy (PGP),"
Request for Comments 2015, Internet Engineering Task Force, Oct.
1996.
[41] N. Freed and N. Borenstein, "Multipurpose internet mail
extensions (MIME) part two: Media types," Request for Comments 2046,
Internet Engineering Task Force, Nov. 1996.
[42] D. Atkins, W. Stallings, and P. Zimmermann, "PGP message
exchange formats," Request for Comments 1991, Internet Engineering
Task Force, Aug. 1996.
[43] R. Atkinson, "Security architecture for the internet protocol,"
Request for Comments 1825, Internet Engineering Task Force, Aug.
1995.
[44] J. Franks, P. Hallam-Baker, J. Hostetler, S. Lawrence, P. Leach,
A. Luotonen, and L. Stewart, "HTTP authentication: Basic and digest
access authentication," Request for Comments 2617, Internet
Engineering Task Force, June 1999.
[45] J. Callas, L. Donnerhacke, H. Finney, and R. Thayer, "OpenPGP
message format," Request for Comments 2440, Internet Engineering Task
Force, Nov. 1998.
[46] N. Freed and N. Borenstein, "Multipurpose internet mail
extensions (MIME) part one: Format of internet message bodies,"
Request for Comments 2045, Internet Engineering Task Force, Nov.
1996.
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[47] D. Crocker, Ed., and P. Overell, "Augmented BNF for syntax
specifications: ABNF," Request for Comments 2234, Internet
Engineering Task Force, Nov. 1997.
[48] R. Hinden, B. Carpenter, and L. Masinter, "Format for literal
IPv6 addresses in URL's," Request for Comments 2732, Internet
Engineering Task Force, Dec. 1999.
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packet-switched teleconferencing system," Journal of Internetworking:
Research and Experience , Vol. 4, pp. 99--120, June 1993. ISI
reprint series ISI/RS-93-359.
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1996.
Full Copyright Statement
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This document and translations of it may be copied and furnished to
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or assist in its implementation may be prepared, copied, published
and distributed, in whole or in part, without restriction of any
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Handley/Schulzrinne/Schooler/Rosenberg [Page 173]
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Table of Contents
1 Introduction ........................................ 1
1.1 Overview of SIP Functionality ....................... 1
1.2 Terminology ......................................... 3
1.3 Definitions ......................................... 3
1.4 Overview of SIP Operation ........................... 7
1.4.1 SIP Addressing ...................................... 8
1.4.2 Locating a SIP Server ............................... 8
1.4.3 SIP Transaction ..................................... 11
1.4.4 SIP Invitation ...................................... 11
1.4.5 Locating a User ..................................... 13
1.4.6 Changing an Existing Session ........................ 15
1.4.7 Registration Services ............................... 15
1.5 Protocol Properties ................................. 16
1.5.1 Minimal State ....................................... 16
1.5.2 Lower-Layer-Protocol Neutral ........................ 16
1.5.3 Text-Based .......................................... 16
2 SIP Uniform Resource Locators ....................... 16
2.1 SIP URL Comparison .................................. 21
2.2 Non-SIP URLs ........................................ 22
3 SIP Message Overview ................................ 22
4 Request ............................................. 24
4.1 Request-Line ........................................ 24
4.2 Methods ............................................. 24
4.2.1 INVITE .............................................. 26
4.2.2 ACK ................................................. 28
4.2.3 OPTIONS ............................................. 29
4.2.4 BYE ................................................. 29
4.2.5 CANCEL .............................................. 30
4.2.6 REGISTER ............................................ 31
4.3 Request-URI ......................................... 35
4.3.1 SIP Version ......................................... 36
4.4 Option Tags ......................................... 36
4.4.1 Registering New Option Tags with IANA ............... 36
5 Response ............................................ 37
5.1 Status-Line ......................................... 37
5.1.1 Status Codes and Reason Phrases ..................... 38
6 Header Field Definitions ............................ 40
6.1 General Header Fields ............................... 42
6.2 Entity Header Fields ................................ 42
6.3 Request Header Fields ............................... 43
6.4 Response Header Fields .............................. 44
6.5 End-to-end and Hop-by-hop Headers ................... 45
6.6 Header Field Format ................................. 45
6.7 Accept .............................................. 46
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6.8 Accept-Encoding ..................................... 46
6.9 Accept-Language ..................................... 47
6.10 Alert-Info .......................................... 47
6.11 Allow ............................................... 47
6.12 Authorization ....................................... 48
6.13 Call-ID ............................................. 48
6.14 Call-Info ........................................... 50
6.15 Contact ............................................. 50
6.16 Content-Disposition ................................. 54
6.17 Content-Encoding .................................... 54
6.18 Content-Language .................................... 55
6.19 Content-Length ...................................... 55
6.20 Content-Type ........................................ 56
6.21 CSeq ................................................ 56
6.22 Date ................................................ 58
6.23 Encryption .......................................... 59
6.24 Expires ............................................. 60
6.25 From ................................................ 61
6.26 Hide ................................................ 63
6.27 In-Reply-To ......................................... 64
6.28 Max-Forwards ........................................ 64
6.29 MIME-Version ........................................ 65
6.30 Organization ........................................ 65
6.31 Priority ............................................ 65
6.32 Proxy-Authenticate .................................. 66
6.33 Proxy-Authorization ................................. 67
6.34 Proxy-Require ....................................... 67
6.35 Record-Route ........................................ 67
6.35.1 Operation ........................................... 67
6.35.2 Construction of Route Header ........................ 68
6.35.3 Request Destination ................................. 69
6.35.4 Syntax .............................................. 69
6.35.5 Example ............................................. 69
6.36 Require ............................................. 69
6.37 Response-Key ........................................ 70
6.38 Retry-After ......................................... 71
6.39 Route ............................................... 72
6.40 Server .............................................. 72
6.41 Subject ............................................. 72
6.42 Supported ........................................... 72
6.43 Timestamp ........................................... 73
6.44 To .................................................. 73
6.45 Unsupported ......................................... 75
6.46 User-Agent .......................................... 75
6.47 Via ................................................. 75
6.47.1 Requests ............................................ 75
6.47.2 Receiver-tagged Via Header Fields ................... 76
6.47.3 Responses ........................................... 77
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6.47.4 User Agent and Redirect Servers ..................... 78
6.47.5 Syntax .............................................. 78
6.48 Warning ............................................. 80
6.49 WWW-Authenticate .................................... 82
7 Status Code Definitions ............................. 82
7.1 Informational 1xx ................................... 83
7.1.1 100 Trying .......................................... 83
7.1.2 180 Ringing ......................................... 83
7.1.3 181 Call Is Being Forwarded ......................... 83
7.1.4 182 Queued .......................................... 83
7.1.5 183 Session Progress ................................ 83
7.2 Successful 2xx ...................................... 84
7.2.1 200 OK .............................................. 84
7.3 Redirection 3xx ..................................... 84
7.3.1 300 Multiple Choices ................................ 84
7.3.2 301 Moved Permanently ............................... 85
7.3.3 302 Moved Temporarily ............................... 85
7.3.4 305 Use Proxy ....................................... 85
7.3.5 380 Alternative Service ............................. 86
7.4 Request Failure 4xx ................................. 86
7.4.1 400 Bad Request ..................................... 86
7.4.2 401 Unauthorized .................................... 86
7.4.3 402 Payment Required ................................ 86
7.4.4 403 Forbidden ....................................... 86
7.4.5 404 Not Found ....................................... 86
7.4.6 405 Method Not Allowed .............................. 86
7.4.7 406 Not Acceptable .................................. 87
7.4.8 407 Proxy Authentication Required ................... 87
7.4.9 408 Request Timeout ................................. 87
7.4.10 409 Conflict ........................................ 87
7.4.11 410 Gone ............................................ 87
7.4.12 411 Length Required ................................. 88
7.4.13 413 Request Entity Too Large ........................ 88
7.4.14 414 Request-URI Too Long ............................ 88
7.4.15 415 Unsupported Media Type .......................... 88
7.4.16 420 Bad Extension ................................... 88
7.4.17 480 Temporarily Unavailable ......................... 88
7.4.18 481 Call Leg/Transaction Does Not Exist ............. 89
7.4.19 482 Loop Detected ................................... 89
7.4.20 483 Too Many Hops ................................... 89
7.4.21 484 Address Incomplete .............................. 89
7.4.22 485 Ambiguous ....................................... 89
7.4.23 486 Busy Here ....................................... 90
7.4.24 487 Request Terminated .............................. 90
7.4.25 488 Not Acceptable Here ............................. 90
7.5 Server Failure 5xx .................................. 90
7.5.1 500 Server Internal Error ........................... 90
7.5.2 501 Not Implemented ................................. 91
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7.5.3 502 Bad Gateway ..................................... 91
7.5.4 503 Service Unavailable ............................. 91
7.5.5 504 Server Time-out ................................. 91
7.5.6 505 Version Not Supported ........................... 91
7.6 Global Failures 6xx ................................. 92
7.6.1 600 Busy Everywhere ................................. 92
7.6.2 603 Decline ......................................... 92
7.6.3 604 Does Not Exist Anywhere ......................... 92
7.6.4 606 Not Acceptable .................................. 92
8 SIP Message Body .................................... 92
8.1 Body Inclusion ...................................... 93
8.2 Message Body Type ................................... 93
8.3 Message Body Length ................................. 93
9 Compact Form ........................................ 94
10 Behavior of SIP Clients and Servers ................. 95
10.1 General Remarks ..................................... 95
10.1.1 Requests ............................................ 95
10.1.2 Responses ........................................... 95
10.2 Source Addresses, Destination Addresses and
Connections .................................................... 96
10.2.1 Unicast UDP ......................................... 96
10.2.2 Multicast UDP ....................................... 96
10.3 TCP ................................................. 97
10.4 Reliability for Requests Other Than INVITE .......... 98
10.4.1 UDP ................................................. 98
10.4.2 TCP ................................................. 99
10.5 Reliability for INVITE Requests ..................... 99
10.5.1 UDP ................................................. 100
10.5.2 TCP ................................................. 101
10.6 Reliability for ACK Requests ........................ 102
10.7 ICMP Handling ....................................... 104
11 Behavior of SIP User Agents ......................... 104
11.1 Caller Issues Initial INVITE Request ................ 104
11.2 Callee Issues Response .............................. 104
11.3 Caller Receives Response to Initial Request ......... 105
11.4 Caller or Callee Generate Subsequent Requests ....... 105
11.5 Receiving Subsequent Requests ....................... 106
12 Behavior of SIP Proxy and Redirect Servers .......... 107
12.1 Redirect Server ..................................... 107
12.2 User Agent Server ................................... 107
12.3 Proxy Server ........................................ 107
12.3.1 Proxying Requests ................................... 108
12.3.2 Proxying Responses .................................. 108
12.3.3 Stateless Proxy: Proxying Responses ................. 108
12.3.4 Stateful Proxy: Receiving Requests .................. 109
12.3.5 Stateful Proxy: Receiving ACKs ...................... 109
12.3.6 Stateful Proxy: Receiving Responses ................. 109
12.3.7 Stateless, Non-Forking Proxy ........................ 110
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12.4 Forking Proxy ....................................... 110
13 Security Considerations ............................. 115
13.1 Confidentiality and Privacy: Encryption ............. 115
13.1.1 End-to-End Encryption ............................... 115
13.1.2 Privacy of SIP Responses ............................ 117
13.1.3 Encryption by Proxies ............................... 118
13.1.4 Hop-by-Hop Encryption ............................... 118
13.1.5 Via field encryption ................................ 118
13.2 Message Integrity and Access Control:
Authentication ................................................. 119
13.2.1 Trusting responses .................................. 122
13.3 Callee Privacy ...................................... 123
13.4 Known Security Problems ............................. 123
14 SIP Authentication using HTTP Basic and Digest
Schemes ........................................................ 123
14.1 Framework ........................................... 123
14.2 Basic Authentication ................................ 124
14.3 Digest Authentication ............................... 124
14.4 Proxy-Authentication ................................ 125
15 SIP Security Using PGP .............................. 126
15.1 PGP Authentication Scheme ........................... 126
15.1.1 The WWW-Authenticate Response Header ................ 126
15.1.2 The Authorization Request Header .................... 127
15.2 PGP Encryption Scheme ............................... 129
15.3 Response-Key Header Field for PGP ................... 130
16 Examples ............................................ 130
16.1 Registration ........................................ 130
16.2 Invitation to a Multicast Conference ................ 132
16.2.1 Request ............................................. 132
16.2.2 Response ............................................ 133
16.3 Two-party Call ...................................... 134
16.4 Terminating a Call .................................. 136
16.5 Forking Proxy ....................................... 137
16.6 Redirects ........................................... 141
16.7 Negotiation ......................................... 143
16.8 OPTIONS Request ..................................... 144
A Minimal Implementation .............................. 145
A.1 Transport Protocol Support .......................... 145
A.2 Client .............................................. 145
A.3 Server .............................................. 146
A.4 Header Processing ................................... 146
B Usage of the Session Description Protocol (SDP)
................................................................ 146
B.1 Configuring Media Streams ........................... 147
B.2 Setting SDP Values for Unicast ...................... 149
B.3 Multicast Operation ................................. 150
B.4 Delayed Media Streams ............................... 150
B.5 Adding and Deleting Media Streams ................... 150
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B.6 Putting Media Streams on Hold ....................... 151
B.7 Subject and SDP "s=" Line ........................... 151
B.8 The SDP "o=" Line ................................... 151
C Summary of Augmented BNF ............................ 152
C.1 Basic Rules ......................................... 154
D IANA Considerations ................................. 156
E Changes from RFC 2543 ............................... 157
F Changes Made in Version 00 .......................... 157
G Changes To Be Made .................................. 163
H Acknowledgments ..................................... 168
I Authors' Addresses .................................. 168
J Bibliography ........................................ 169
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