Internet Draft
Internet Engineering Task Force Alan Johnston
Internet Draft WorldCom
Document: draft-ietf-sip-call-flows-01.txt Steve Donovan
Category: Informational Robert Sparks
July 2000 Chris Cunningham
Expires: January 2001 Dean Willis
Jonathan Rosenberg
dynamicsoft
Kevin Summers
TTI
Henning Schulzrinne
Columbia University
SIP Telephony Call Flow Examples
Status of this Memo
This document is an Internet-Draft and is in full conformance with
all provisions of Section 10 of RFC2026[1].
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF), its areas, and its working groups. Note that other
groups may also distribute working documents as Internet-Drafts.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet- Drafts as reference
material or to cite them other than as "work in progress."
The list of current Internet-Drafts can be accessed at
http://www.ietf.org/ietf/1id-abstracts.txt
The list of Internet-Draft Shadow Directories can be accessed at
http://www.ietf.org/shadow.html.
Abstract
This document gives examples of SIP (Session Initiation Protocol)
call flows for IP telephony. Elements in these
call flows include SIP User Agents and Clients, SIP Proxy and
Redirect Servers, and Gateways to the PSTN (Public Switch Telephone
Network). IP telephony scenarios include SIP Registration, SIP to
SIP calling, SIP to Gateway, Gateway to SIP, and Gateway to Gateway
via SIP. Call flow diagrams and message details are shown. PSTN
telephony protocols are illustrated using ISDN (Integrated Services
Digital Network), ANSI ISUP (ISDN User Part), and FGB (Feature Group
B) circuit associated signaling. PSTN calls are illustrated using
global telephone numbers from the PSTN and private extensions served
on by a PBX (Private Branch Exchange). Example SIP messages used for
testing during SIP "bakeoff" events include SIP "torture test"
messages, and messages with invalid parameters, methods, and tags.
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Table of Contents
1 Overview...................................................3
1.1 General Assumptions........................................4
1.2 Legend for Message Flows...................................6
1.3 SIP Protocol Assumptions...................................6
1.4 Changes to 00 draft .......................................7
2 SIP Registration Services..................................8
2.1 Success Scenarios..........................................8
2.1.1 SIP Client New Registration................................8
2.1.2 User updates contact list.................................11
2.1.3 User Requests Current Contact List........................13
2.1.4 User Cancels Registration.................................15
2.2 Failure Scenarios.........................................16
2.2.1 Unsuccessful SIP registration.............................16
3 SIP to SIP Dialing........................................18
3.1 Success Scenarios.........................................18
3.1.1 Successful Simple SIP to SIP..............................19
3.1.2 Successful SIP to SIP through two proxies.................22
3.1.3 Successful SIP to SIP with Proxy failure..................31
3.1.4 Successful SIP to SIP through SIP Firewall Proxy..........39
3.1.5 Successful SIP to SIP via Redirect and Proxy..............47
3.2 Failure Scenarios.........................................53
3.2.1 Unsuccessful SIP to SIP no answer.........................53
3.2.2 Unsuccessful SIP to SIP busy..............................60
3.2.3 Unsuccessful SIP to SIP no response.......................64
3.2.4 Unsuccessful SIP to SIP Temporarily Unavailable...........69
4 SIP to Gateway Dialing....................................74
4.1 Success Scenarios.........................................74
4.1.1 Successful SIP to ISUP PSTN call..........................75
4.1.2 Successful SIP to ISDN PBX call...........................82
4.1.3 Successful SIP to ISUP PSTN call with overflow............93
4.2 Failure Scenarios........................................101
4.2.1 Unsuccessful SIP to PSTN call: Treatment from PSTN.......102
4.2.2 Unsuccessful SIP to PSTN: REL w/Cause from PSTN..........108
4.2.3 Unsuccessful SIP to PSTN: ANM Timeout....................112
5 Gateway to SIP Dialing...................................117
5.1 Success Scenarios........................................117
5.1.1 Successful PSTN to SIP call..............................118
5.1.2 Successful PSTN to SIP call, Fast Answer.................125
5.1.3 Successful PBX to SIP call...............................130
5.2 Failure Scenarios........................................136
5.2.1 Unsuccessful PSTN to SIP REL, SIP error mapped to REL....136
5.2.2 Unsuccessful PSTN to SIP REL, SIP busy mapped to REL.....138
5.2.3 Unsuccessful PSTN->SIP, SIP error interworking to tones..142
5.2.4 Unsuccessful PSTN->SIP, ACM timeout......................146
5.2.5 Unsuccessful PSTN->SIP, ACM timeout, stateless Proxy.....151
5.2.6 Unsuccessful PSTN->SIP, ANM timeout......................156
6 Gateway to Gateway Dialing via SIP Network...............162
6.1 Success Scenarios........................................162
6.1.1 Successful ISUP PSTN to ISUP PSTN call...................163
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6.1.2 Successful FGB PBX to ISDN PBX call with overflow........171
7 SIP Test Messages........................................180
7.1 INVITE Parser Torture Test Message.......................180
7.2 INVITE with Proxy-Require and Require....................181
7.3 INVITE with Unknown Schemes in URIs and URLs.............181
7.4 REGISTER with Y2038 Test.................................182
7.5 INVITE with inconsistent Accept and message body.........182
7.6 INVITE with non-SDP message body.........................183
7.7 Unknown Method Message...................................183
7.8 Unknown Method with CSeq Error...........................184
7.9 REGISTER with Unknown Authorization Scheme...............184
7.10 INVITE with Invalid SIP Version Number...................185
7.11 INVITE missing Required Headers..........................186
7.12 INVITE with Duplicate Required Headers...................186
7.13 INVITE with Illegal Expires Header.......................187
7.14 200 OK Response with Broadcast Via Header................187
7.15 INVITE with Invalid Via and Contact Headers..............188
7.16 INVITE with Incorrect Content-Length Header..............188
7.17 INVITE with Invalid Value for Content-Length.............189
7.18 INVITE with Garbage after Message Body...................189
7.19 INVITE with Error in Display Name in To Header...........190
8 Acknowledgements.........................................191
9 References...............................................191
1 Overview
The call flows shown in this document were developed in the design of
a carrier-class SIP IP Telephony network. They represent an example
minimum set of functionality for SIP to be used in IP Telephony
applications. The message examples were developed during the SIP
interoperability testing "bake-offs."
It is the hope of the authors that this document will be useful for
SIP implementors, designers, and protocol researchers alike and will
help further the goal of a standard SIP implementation for IP
Telephony. It is envisioned that as changes to the standard and
additional RFCs are added that this document will reflect those
changes and represent the current state of a standard interoperable
SIP IP Telephony implementation.
These call flows are based on the current version 2.0 of SIP in
RFC2543[2]. Additions and changes to SIP necessary for PSTN
interworking are referenced as IETF Internet-Drafts as they are used
in the call flows.
Various PSTN signaling protocols are illustrated in this document:
ISDN (Integrated Services Digital Network), ANSI ISUP (ISDN User
Part) and FGB (Feature Group B) circuit associated signaling. They
were chosen to illustrate the nature of SIP/PSTN interworking - they
are not a complete or even representative set. Also, some details
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and parameters of these PSTN protocols have been omitted. The intent
of this document was not to provide a complete and exact mapping of
PSTN protocols to SIP. Rather the emphasis is on the SIP signaling,
the message interaction, and the modifications to SIP currently
proposed to solve IP Telephony issues.
Finally, some example messages are given along with expected
behavior of clients and servers.
1.1 General Assumptions
A number of architecture, network, and protocol assumptions underlie
the call flows in this document. They are outlined in this section
so that they may be taken into consideration. Differences in these
assumptions will affect the nature of the call flows.
The authentication of SIP User Agents in these example call flows is
performed using SIP Digest[2].
No authentication of Gateways is shown, since it is assumed that:
. Gateways will only accept calls routed through a trusted Proxy.
. Proxies will perform the Client authentication.
. The Proxy and the Gateway will authenticate each other using
IPSec[4].
The SIP Proxy Server has access to a Location Manager and other
databases. Information present in the Request-URI and the context
(From header) is sufficient to determine to which proxy or gateway
the message should be routed. In most cases, a primary and secondary
route will be determined in case of Proxy or Gateway failure
downstream.
The Proxy Servers in these call flows insert Record-Route headers
into requests to ensure that they are in the signaling path for
future message exchanges. This allows them to implement features
later in the call.
Gateways receive enough information in the Request-URI field to
determine how to route a call, i.e. what trunk group or link to
select, what digits to outpulse, etc.
Gateways provide tones (ringing, busy, etc) and announcements to the
PSTN side based on SIP response messages, or pass along audio in-band
tones (ringing, busy tone, etc.) in an early media stream to the SIP
side.
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Two types of Gateways are described in this document. The actual
names of Gateways will be vendor and implementation specific.
However, two catagories are described here since the type of Gateway
determines the form of the SIP URL used to identify them. The two
types are:
. Network Gateway. This high port count PSTN gateway originates
and terminates calls to the PSTN. Its use is shared by many
customers. Incoming calls from the PSTN have the From header
populated with a SIP URL containing the telephone number from
the calling party telephone number, if available. A Network
Gateway typically uses carrier protocols such as SS7.
. Enterprise Gateway. This low port count PBX (Private Branch
Exchange) gateway has trunks or lines for a single customer or
user. Incoming calls from the PBX have the From header
populated with a provisionable string which uniquely identifies
the customer, trunk group, or carrier. This allows private
numbers to be interpreted in their correct context. An
Enterprise Gateway typically uses SS7, ISDN, circuit associated
signaling, or other PBX interfaces.
The interactions between the Proxy and Gateway can be summarized as
follows:
. The SIP Proxy Server performs digit analysis and lookup and
locates the correct gateway.
. The SIP Proxy Server performs gateway location based on primary
and secondary routing.
Digit handling by the Gateways will be as follows:
. Dialed digits received from a Network or Enterprise Gateway will
be put in a SIP URL with a telephone number. The number will
either be globalized (e.g. sip:+1-314-555-1111@ngw.wcom.com
;user=phone) or left as a private number (sip:555-
6666@gw.wcom.com;phone-context=p1234) which will require
interpretation based on From header. The "phone-context"
qualifier is used to interpret the private number. It is used
the same as the tag of the same name from the tel URL draft[5].
All Gateways will need to be provisioned to be able to parse the
user portion of a Request-URI to determine the customer, trunk
group, or circuit referenced.
. The From header will be populated with a SIP URL with a
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telephone number if it is Calling Party number (CgPN) from the
PSTN. If it is an Enterprise Gateway, a provisionable string
which uniquely identifies the customer, trunk group, or carrier
will be used in the sip URI (e.g. From:
sip:ProvisionableString@gw1.wcom.com;user=phone).
. Note that an alternative to using a SIP URL for telephone
numbers is the tel URL[5]. The major difference between using
the SIP URL and the tel URL is that the SIP URL is routable in a
SIP network (resolves down to an IP address) where the tel URL
is not (it just represents digits). For example, a SIP URL can
be used in a Contact header, but a tel URL can not.
These flows show UDP for transport. TCP could also be used.
1.2 Legend for Message Flows
Dashed lines (---) represent control messages that are mandatory to
the call scenario. These control messages can be SIP or PSTN
signaling.
Double dashed lines (===) represent media paths between network
elements.
Messages with parenthesis around name represent optional control
messages.
Messages are identified in the Figures as F1, F2, etc. This
references the message details in the table that follows the Figure.
Comments in the message details are shown in the following form:
/* Comments. */
1.3 SIP Protocol Assumptions
Except for the following, this call flows document uses the April
1999 version 2.0 of SIP defined by RFC2543[2]. The following
changes/extensions are assumed throughout:
. A Contact header is included with every INVITE message.
. A Contact header is included in every 200 OK Response.
. The 183 Session Progress response message[5] is used in SIP to
Gateway and Gateway to Gateway via SIP calling (Sections 4 and
6). The 183 response with SDP will cause the User Agent to
immediately play the SDP media stream to hear in-band call
progress information. See Section 4 for more information.
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. A Content-Length header is present in every message, set to zero
if there is no message body.
. The final entry in a Route header is always the Contact
information obtained from the INVITE or 200 OK messages.
. In the SDP message bodies, the time field is "t=0 0" It is
expected that an actual SDP message body would have a non-zero
start timestamp.
1.4 Changes to 00 draft
The major changes between this draft and the previous draft are
listed below:
- SIP Telephony Service Examples have been removed from the draft.
They will be revised using the TRANSFER header in a separate
draft.
- Updated draft with RFC2543bis changes including: adding maddr to
all Record-Route and Route headers, adding branch tags to Via
headers inserted by proxies, added 487 response to CANCEL
scenarios.
- Added example of INFO method in 5.1.1.
- Added Session: media to all 183 messages.
- Corrected a number of typos including putting user=phone tags
inside <>, fixing Request-URI on PRACK, added missing tags, fixed
Request-URIs that did not match To header in initial INVITE.
- Corrected all registrations to have same Call-ID.
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2 SIP Registration Services
2.1 Success Scenarios
Registration either validates or invalidates a SIP client for user
services provided by the SIP proxy and/or SIP server. Additionally,
the client provides one or more contact locations to the SIP server
with the registration request. Registration is used by a Proxy to
route incoming calls in an IP Telephony network. All these call
flows assume that registration requires authentication, otherwise an
imposter could "hijack" someone else's calls.
2.1.1 SIP Client New Registration
User B SIP Server
| |
| REGISTER F1 |
|------------------------------>|
| |
| 401 Unauthorized F2 |
|<------------------------------|
| |
| REGISTER F3 |
|------------------------------>|
| |
| 200 OK F4 |
|<------------------------------|
| |
User B initiates a new SIP session with the SIP Server (i.e. the user
"logs on to" the SIP server). User B sends a SIP REGISTER request to
the SIP server. The request includes the user's contact list. The
SIP server provides a challenge to User B. User B enters her/his
valid user ID and password. User B's SIP client encrypts the user
information according to the challenge issued by the SIP server and
sends the response to the SIP server. The SIP server validates the
user's credentials. It registers the user in its contact database
and returns a response (200 OK) to User B's SIP client. The response
includes the user's current contact list in Contact headers. The
format of the authentication shown is SIP digest as described by
RFC2543[2].
Message Details
F1 REGISTER B -> SIP Server
REGISTER sip:ss2.wcom.com SIP/2.0
Via: SIP/2.0/UDP there.com:5060
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From: LittleGuy
To: LittleGuy
Call-ID: 123456789@here.com
CSeq: 1 REGISTER
Contact: LittleGuy
Contact: sip:+1-972-555-2222@gw1.wcom.com;user=phone
Contact: tel:+1-972-555-2222
Content-Length: 0
F2 401 Unauthorized SIP Server -> User B
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP there.com:5060
From: LittleGuy
To: LittleGuy
Call-ID: 123456789@here.com
CSeq: 1 REGISTER
WWW-Authenticate: Digest realm="MCI WorldCom SIP", domain="wcom.com",
nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", stale="FALSE",
algorithm="MD5"
Content-Length: 0
F3 REGISTER B -> SIP Server
REGISTER sip:ss2.wcom.com SIP/2.0
Via: SIP/2.0/UDP there.com:5060
From: LittleGuy
To: LittleGuy
Call-ID: 123456789@here.com
CSeq: 1 REGISTER
Contact: LittleGuy
Contact: sip:+1-972-555-2222@gw1.wcom.com;user=phone
Contact: tel:+1-972-555-2222
Authorization:Digest username="UserB", realm="MCI WorldCom SIP",
nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="",
uri="sip:ss2.wcom.com", response="dfe56131d1958046689cd83306477ecc"
Content-Length: 0
F4 200 OK SIP Server -> B
SIP/2.0 200 OK
Via: SIP/2.0/UDP there.com:5060
From: LittleGuy
To: LittleGuy
Call-ID: 123456789@here.com
CSeq: 1 REGISTER
Contact: LittleGuy
Contact: sip:+1-972-555-2222@gw1.wcom.com;user=phone
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Contact: tel:+1-972-555-2222
Content-Length: 0
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2.1.2 User updates contact list
User B SIP Server
| |
| REGISTER F1 |
|------------------------------>|
| |
| 200 OK F2 |
|<------------------------------|
| |
User B wishes to update the list of addresses where the SIP server
will redirect or forward INVITE requests.
User B sends a SIP REGISTER request to the SIP server. User B's
request includes an updated contact list. Since the user already has
authenticated with the server, the user supplies authentication
credentials with the request and is not challenged by the server.
The SIP server validates the user's credentials. It registers the
user in its contact database, updates the user's contact list, and
returns a response (200 OK) to User B's SIP client. The response
includes the user's current contact list in Contact headers.
Message Details
F1 REGISTER B -> SIP Server
REGISTER sip:ss2.wcom.com SIP/2.0
Via: SIP/2.0/UDP there.com:5060
From: LittleGuy
To: LittleGuy
Call-ID: 123456789@here.com
CSeq: 1 REGISTER
Contact: mailto:UserB@there.com
Authorization:Digest username="UserB", realm="MCI WorldCom SIP",
nonce="1cec4341ae6cbe5a359ea9c8e88df84f", opaque="",
uri="sip:ss2.wcom.com", response="71ba27c64bd01de719686aa4590d5824"
Content-Length: 0
F2 200 OK SIP Server -> B
SIP/2.0 200 OK
Via: SIP/2.0/UDP there.com:5060
From: LittleGuy
To: LittleGuy
Call-ID: 123456789@here.com
CSeq: 1 REGISTER
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Contact: LittleGuy
Contact: sip:+1-972-555-2222@gw1.wcom.com;user=phone
Contact: tel:+1-972-555-2222
Contact: mailto:UserB@there.com
Content-Length: 0
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2.1.3 User Requests Current Contact List
User B SIP Server
| |
| REGISTER F1 |
|------------------------------>|
| |
| 200 OK F2 |
|<------------------------------|
| |
User B sends a register request to the Proxy Server containing no
Contact headers, indicating the user wishes to query the server for
the user's current contact list. Since the user already has
authenticated with the server, the user supplies authentication
credentials with the request and is not challenged by the server. The
SIP server validates the user's credentials. It registers the user
in its contact database and returns a response (200 OK) to User B's
SIP client. The response includes the user's current contact list in
Contact headers.
Message Details
F1 REGISTER B -> SIP Server
REGISTER sip:ss2.wcom.com SIP/2.0Via: SIP/2.0/UDP there.com:5060
From: LittleGuy
To: LittleGuy
Call-ID: 123456789@here.com
CSeq: 1 REGISTER
Authorization:Digest username="UserB", realm="MCI WorldCom SIP",
nonce="df84f1cec4341ae6cbe5ap359a9c8e88", opaque="",
uri="sip:ss2.wcom.com", response="aa7ab4678258377c6f7d4be6087e2f60"
Content-Length: 0
F2 200 OK SIP Server -> B
SIP/2.0 200 OK
Via: SIP/2.0/UDP there.com:5060
From: LittleGuy
To: LittleGuy
Call-ID: 123456789@here.com
CSeq: 1 REGISTER
Contact: LittleGuy
Contact: sip:+1-972-555-2222@gw1.wcom.com;user=phone
Contact: tel:+1-972-555-2222
Contact: mailto:UserB@there.com
Content-Length: 0
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2.1.4 User Cancels Registration
User B SIP Server
| |
| REGISTER F1 |
|------------------------------>|
| |
| 200 OK F2 |
|<------------------------------|
| |
User B wishes to cancel her/his registration with the SIP
registrar/redirect server. User B sends a SIP REGISTER request to the
SIP server. The request has an expiration period of 0 and applies to
all existing contact locations. Since the user already has
authenticated with the server, the user supplies authentication
credentials with the request and is not challenged by the server.
The SIP server validates the user's credentials. It clears the
user's contact list, and returns a response (200 OK) to User B's SIP
client.
Message Details
F1 REGISTER B -> SIP Server
REGISTER sip:ss2.wcom.com SIP/2.0
Via: SIP/2.0/UDP there.com:5060
From: LittleGuy
To: LittleGuy
Call-ID: 123456789@here.com
CSeq: 1 REGISTER
Expires: 0
Contact: *
Authorization:Digest username="UserB", realm="MCI WorldCom SIP",
nonce="88df84f1cac4341aea9c8ee6cbe5a359", opaque="",
uri="sip:ss2.wcom.com", response="ff0437c51696f9a76244f0cf1dbabbea"
Content-Length: 0
F2 200 OK SIP Server -> B
SIP/2.0 200 OK
Via: SIP/2.0/UDP there.com:5060
From: LittleGuy
To: LittleGuy
Call-ID: 123456789@here.com
CSeq: 1 REGISTER
Content-Length: 0
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2.2 Failure Scenarios
2.2.1 Unsuccessful SIP registration
User B SIP Server
| |
| REGISTER F1 |
|------------------------------>|
| |
| 401 Unauthorized F2 |
|<------------------------------|
| |
| REGISTER F3 |
|------------------------------>|
| |
| 401 Unauthorized F4 |
|<------------------------------|
| |
User B sends a SIP REGISTER request to the SIP Server. The SIP
server provides a challenge to User B. User B enters her/his user ID
and password. User B's SIP client encrypts the user information
according to the challenge issued by the SIP server and sends the
response to the SIP server. The SIP server attempts to validate the
user's credentials, but they are not valid (the user's password does
not match the password established for the user's account). The
server returns a response (401 Unauthorized) to User B's SIP client.
Message Details
F1 REGISTER B -> SIP Server
REGISTER sip:ss2.wcom.com SIP/2.0
Via: SIP/2.0/UDP there.com:5060
From: LittleGuy
To: LittleGuy
Call-ID: 123456789@here.com
CSeq: 1 REGISTER
Contact: LittleGuy
Contact: sip:+1-972-555-2222@gw1.wcom.com;user=phone
Contact: tel:+1-972-555-2222
Content-Length: 0
F2 Unauthorized SIP Server -> User B
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SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP there.com:5060
From: LittleGuy
To: LittleGuy
Call-ID: 123456789@here.com
CSeq: 1 REGISTER
WWW-Authenticate: Digest realm="MCI WorldCom SIP", domain="wcom.com",
nonce="f1cec4341ae6ca9c8e88df84be55a359", opaque="", stale="FALSE",
algorithm="MD5"
Content-Length: 0
F3 REGISTER B -> SIP Server
REGISTER sip:ss2.wcom.com SIP/2.0
Via: SIP/2.0/UDP there.com:5060
From: LittleGuy
To: LittleGuy
Call-ID: 123456789@here.com
CSeq: 1 REGISTER
Contact: LittleGuy
Contact: sip:+1-972-555-2222@gw1.wcom.com;user=phone
Contact: tel:+1-972-555-2222
Authorization:Digest username="UserB", realm="MCI WorldCom SIP",
nonce="f1cec4341ae6ca9c8e88df84be55a359", opaque="",
uri="sip:ss2.wcom.com", response="61f8470ceb87d7ebf508220214ed438b"
Content-Length: 0
/* The response above encodes the incorrect password _IForgotIt_ */
F4 401 Unauthorized SIP Server -> User B
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP there.com:5060
From: LittleGuy
To: LittleGuy
Call-ID: 123456789@here.com
CSeq: 1 REGISTER
WWW-Authenticate: Digest realm="MCI WorldCom SIP", domain="wcom.com",
nonce="84f1c1ae6cbe5ua9c8e88dfa3ecm3459", opaque="", stale="FALSE",
algorithm="MD5"
Content-Length: 0
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3 SIP to SIP Dialing
3.1 Success Scenarios
This section details calls between two SIP User Agent Clients (UACs)
- User A and User B. User A (LittleGuy sip:UserA@here.com) and User
B (BigGuy sip:UserB@there.com) are assumed to be SIP phones or SIP-
enabled devices. Calls route using at least one SIP Proxy server.
The successful calls show the initial signaling, the exchange of
media information in the form of SDP payloads, the establishment of
the media session, then finally the termination of the call.
SIP digest authentication is used by the first Proxy Server to
authenticate the caller User A. It is assumed that User B has
registered with Proxy Server Proxy 2 as per Section 2.1 to be able to
receive the calls.
Johnston, et al. Informational [Page 18]
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3.1.1 Successful Simple SIP to SIP
User A User B
| |
| INVITE F1 |
|----------------------->|
| (100 Trying) F2 |
|<-----------------------|
| 180 Ringing F3 |
|<-----------------------|
| |
| 200 OK F4 |
|<-----------------------|
| ACK F5 |
|----------------------->|
| Both Way RTP Media |
|<======================>|
| |
| BYE F6 |
|<-----------------------|
| 200 OK F7 |
|----------------------->|
| |
In this scenario, User A completes a call to User B directly.
Message Details
F1 INVITE A -> Proxy 1
INVITE sip:UserB@there.com SIP/2.0
Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy
Call-ID: 12345600@here.com
CSeq: 1 INVITE
Contact: BigGuy
Content-Type: application/sdp
Content-Length: 147
v=0
o=UserA 2890844526 2890844526 IN IP4 here.com
s=Session SDP
c=IN IP4 100.101.102.103
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000
Johnston, et al. Informational [Page 19]
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F2 (100 Trying) User B -> User A
SIP/2.0 100 Trying
Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy
Call-ID: 12345601@here.com
CSeq: 1 INVITE
Content-Length: 0
F3 180 Ringing User B -> User A
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy
Call-ID: 12345601@here.com
CSeq: 1 INVITE
Content-Length: 0
F4 200 OK User B -> User A
SIP/2.0 200 OK
Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy
Call-ID: 12345601@here.com
CSeq: 1 INVITE
Contact: LittleGuy
Content-Type: application/sdp
Content-Length: 134
v=0
o=UserB 2890844527 2890844527 IN IP4 there.com
s=Session SDP
c=IN IP4 110.111.112.113
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F5 ACK User A -> User B
ACK sip:UserB@there.com SIP/2.0
Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy
Call-ID: 12345601@here.com
CSeq: 1 ACK
Johnston, et al. Informational [Page 20]
Internet Draft SIP Telephony Call Flow Examples July 2000
Content-Length: 0
/* RTP streams are established between A and B */
/* User B Hangs Up with User A. */
F6 BYE User B -> User A
BYE sip: UserA@here.com SIP/2.0
Via: SIP/2.0/UDP there.com:5060
From: LittleGuy
To: BigGuy
Call-ID: 12345601@here.com
CSeq: 1 BYE
Content-Length: 0
F7 200 OK User A -> User B
SIP/2.0 200 OK
Via: SIP/2.0/UDP there.com:5060
From: LittleGuy
To: BigGuy
Call-ID: 12345601@here.com
CSeq: 1 BYE
Content-Length: 0
Johnston, et al. Informational [Page 21]
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3.1.2 Successful SIP to SIP through two proxies
User A Proxy 1 Proxy 2 User B
| | | |
| INVITE F1 | | |
|--------------->| | |
| 407 F2 | | |
|<---------------| | |
| ACK F3 | | |
|--------------->| | |
| INVITE F4 | | |
|--------------->| INVITE F5 | |
| (100) F6 |--------------->| INVITE F7 |
|<---------------| (100) F8 |--------------->|
| |<---------------| |
| | | 180 F9 |
| | 180 F10 |<---------------|
| 180 F11 |<---------------| |
|<---------------| | 200 F12 |
| | 200 F13 |<---------------|
| 200 F14 |<---------------| |
|<---------------| | |
| ACK F15 | | |
|--------------->| ACK F16 | |
| |--------------->| ACK F17 |
| | |--------------->|
| Both Way RTP Media |
|<================================================>|
| | | BYE F18 |
| | BYE F19 |<---------------|
| BYE F20 |<---------------| |
|<---------------| | |
| 200 F21 | | |
|--------------->| 200 F22 | |
| |--------------->| 200 F23 |
| | |--------------->|
| | | |
In this scenario, User A completes a call to User B using two proxies
Proxy 1 and Proxy 2. The initial INVITE (F1) does not contain the
Authorization credentials Proxy 1 requires, so a 407 Proxy
Authorization response is sent containing the challenge information.
A new INVITE (F4) is then sent containing the correct credentials and
the call proceeds. The call terminates when User B disconnects by
initiating a BYE message.
Proxy 1 inserts a Record-Route header into the INVITE message to
ensure that it is present in all subsequent message exchanges. Proxy
2 also inserts itself into the Record-Route header. The ACK (F15)
and BYE (F18) both have a Route header.
Johnston, et al. Informational [Page 22]
Internet Draft SIP Telephony Call Flow Examples July 2000
A tag is inserted by User B in message F9 since the initial INVITE
message contains more than one Via header and may have been forked.
Message Details
F1 INVITE A -> Proxy 1
INVITE sip:UserB@there.com SIP/2.0
Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy
Call-ID: 12345600@here.com
CSeq: 1 INVITE
Contact: BigGuy
Content-Type: application/sdp
Content-Length: 147
v=0
o=UserA 2890844526 2890844526 IN IP4 here.com
s=Session SDP
c=IN IP4 100.101.102.103
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000
/* Proxy 1 challenges User A for authentication */
F2 407 Proxy Authorization Required Proxy 1 -> User A
SIP/2.0 407 Proxy Authorization Required
Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy
Call-ID: 12345600@here.com
CSeq: 1 INVITE
Proxy-Authenticate: Digest realm="MCI WorldCom SIP",
domain=_wcom.com_, nonce="wf84f1ceczx41ae6cbe5aea9c8e88d359",
opaque="", stale="FALSE", algorithm="MD5"
Content-Length: 0
F3 ACK A -> Proxy 1
ACK sip:UserB@there.com SIP/2.0
Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy
Johnston, et al. Informational [Page 23]
Internet Draft SIP Telephony Call Flow Examples July 2000
Call-ID: 12345600@here.com
CSeq: 1 INVITE
Content-Length: 0
/* User A responds be re-sending the INVITE with authentication
credentials in it. */
F4 INVITE A -> Proxy 1
INVITE sip:UserB@there.com SIP/2.0
Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy
Call-ID: 12345601@here.com
CSeq: 1 INVITE
Contact: BigGuy
Authorization:Digest username="UserA", realm="MCI WorldCom SIP",
nonce="wf84f1ceczx41ae6cbe5aea9c8e88d359", opaque="",
uri="sip:ss1.wcom.com", response="42ce3cef44b22f50c6a6071bc8"
Content-Type: application/sdp
Content-Length: 147
v=0
o=UserA 2890844526 2890844526 IN IP4 here.com
s=Session SDP
c=IN IP4 100.101.102.103
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000
/* Proxy 1 accepts the credentials and forwards the INVITE to Proxy
2. Proxy 1 is assumed to have been authenticated by Proxy 2 using
IPSec. Client for A prepares to receive data on port 49172 from the
network. */
F5 INVITE Proxy 1 -> Proxy 2
INVITE sip:UserB@there.com SIP/2.0
Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
Via: SIP/2.0/UDP here.com:5060
Record-Route:
From: BigGuy
To: LittleGuy
Call-ID: 12345601@here.com
CSeq: 1 INVITE
Contact: BigGuy
Content-Type: application/sdp
Content-Length: 132
Johnston, et al. Informational [Page 24]
Internet Draft SIP Telephony Call Flow Examples July 2000
v=0
o=UserA 2890844526 2890844526 IN IP4 here.com
s=Session SDP
c=IN IP4 100.101.102.103
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F6 (100 Trying) Proxy 1 -> User A
SIP/2.0 100 Trying
Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy
Call-ID: 12345601@here.com
CSeq: 1 INVITE
Content-Length: 0
F7 INVITE Proxy 2 -> B
INVITE sip:UserB@there.com SIP/2.0
Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
Via: SIP/2.0/UDP here.com:5060
Record-Route: ,
From: BigGuy
To: LittleGuy
Call-ID: 12345601@here.com
CSeq: 1 INVITE
Contact: BigGuy
Content-Type: application/sdp
Content-Length: 132
v=0
o=UserA 2890844526 2890844526 IN IP4 here.com
s=Session SDP
c=IN IP4 100.101.102.103
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F8 (100 Trying) Proxy 2 -> Proxy 1
SIP/2.0 100 Trying
Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
Via: SIP/2.0/UDP here.com:5060
From: BigGuy
Johnston, et al. Informational [Page 25]
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To: LittleGuy
Call-ID: 12345601@here.com
CSeq: 1 INVITE
Content-Length: 0
F9 180 Ringing B -> Proxy 2
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy ;tag=314159
Call-ID: 12345601@here.com
CSeq: 1 INVITE
Content-Length: 0
F10 180 Ringing Proxy 2 -> Proxy 1
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy ;tag=314159
Call-ID: 12345601@here.com
CSeq: 1 INVITE
Content-Length: 0
F11 180 Ringing Proxy 1 -> A
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy ;tag=314159
Call-ID: 12345601@here.com
CSeq: 1 INVITE
Content-Length: 0
F12 200 OK B -> Proxy 2
SIP/2.0 200 OK
Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
Via: SIP/2.0/UDP here.com:5060
Record-Route: ,
From: BigGuy
Johnston, et al. Informational [Page 26]
Internet Draft SIP Telephony Call Flow Examples July 2000
To: LittleGuy ;tag=314159
Call-ID: 12345601@here.com
CSeq: 1 INVITE
Contact: LittleGuy
Content-Type: application/sdp
Content-Length: 134
v=0
o=UserB 2890844527 2890844527 IN IP4 there.com
s=Session SDP
c=IN IP4 110.111.112.113
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F13 200 OK Proxy 2 -> Proxy 1
SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
Via: SIP/2.0/UDP here.com:5060
Record-Route: ,
From: BigGuy
To: LittleGuy ;tag=314159
Call-ID: 12345601@here.com
CSeq: 1 INVITE
Contact: LittleGuy
Content-Type: application/sdp
Content-Length: 134
v=0
o=UserB 2890844527 2890844527 IN IP4 there.com
s=Session SDP
c=IN IP4 110.111.112.113
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F14 200 OK Proxy 1 -> A
SIP/2.0 200 OK
Via: SIP/2.0/UDP here.com:5060
Record-Route: ,
From: BigGuy
To: LittleGuy ;tag=314159
Call-ID: 12345601@here.com
CSeq: 1 INVITE
Contact: LittleGuy
Johnston, et al. Informational [Page 27]
Internet Draft SIP Telephony Call Flow Examples July 2000
Content-Type: application/sdp
Content-Length: 134
v=0
o=UserB 2890844527 2890844527 IN IP4 there.com
s=Session SDP
c=IN IP4 110.111.112.113
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F15 ACK A -> Proxy 1
ACK sip:UserB@there.com SIP/2.0
Via: SIP/2.0/UDP here.com:5060
Route: ,
From: BigGuy
To: LittleGuy ;tag=314159
Call-ID: 12345601@here.com
CSeq: 1 ACK
Content-Length: 0
F16 ACK Proxy 1 -> Proxy 2
ACK sip:UserB@there.com SIP/2.0
Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
Via: SIP/2.0/UDP here.com:5060
Route:
From: BigGuy
To: LittleGuy ;tag=314159
Call-ID: 12345601@here.com
CSeq: 1 ACK
Content-Length: 0
F17 ACK Proxy 2 -> B
ACK sip: UserB@there.com SIP/2.0
Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy ;tag=314159
Call-ID: 12345601@here.com
CSeq: 1 ACK
Content-Length: 0
Johnston, et al. Informational [Page 28]
Internet Draft SIP Telephony Call Flow Examples July 2000
/* RTP streams are established between A and B */
/* User B Hangs Up with User A. */
F18 BYE User B -> Proxy 2
BYE sip: UserA@here.com SIP/2.0
Via: SIP/2.0/UDP there.com:5060
Route:
.
From: LittleGuy ;tag=314159
To: BigGuy
Call-ID: 12345601@here.com
CSeq: 1 BYE
Content-Length: 0
F19 BYE Proxy 2 -> Proxy 1
BYE sip: UserA@here.com SIP/2.0
Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
Via: SIP/2.0/UDP there.com:5060
Route:
From: LittleGuy ;tag=314159
To: BigGuy
Call-ID: 12345601@here.com
CSeq: 1 BYE
Content-Length: 0
F20 BYE Proxy 1 -> User A
BYE sip: UserA@here.com SIP/2.0
Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
Via: SIP/2.0/UDP there.com:5060
From: LittleGuy ;tag=314159
To: BigGuy
Call-ID: 12345601@here.com
CSeq: 1 BYE
Content-Length: 0
F21 200 OK User A -> Proxy 1
SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
Via: SIP/2.0/UDP there.com:5060
From: LittleGuy ;tag=314159
Johnston, et al. Informational [Page 29]
Internet Draft SIP Telephony Call Flow Examples July 2000
To: BigGuy
Call-ID: 12345601@here.com
CSeq: 1 BYE
Content-Length: 0
F22 200 OK Proxy 1 -> Proxy 2
SIP/2.0 200 OK
Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
Via: SIP/2.0/UDP there.com:5060
From: LittleGuy ;tag=314159
To: BigGuy
Call-ID: 12345601@here.com
CSeq: 1 BYE
Content-Length: 0
F23 200 OK Proxy 2 -> User B
SIP/2.0 200 OK
Via: SIP/2.0/UDP there.com:5060
From: LittleGuy ;tag=314159
To: BigGuy
Call-ID: 12345601@here.com
CSeq: 1 BYE
Content-Length: 0
Johnston, et al. Informational [Page 30]
Internet Draft SIP Telephony Call Flow Examples July 2000
3.1.3 Successful SIP to SIP with Proxy failure
User A Proxy 1 Proxy 2 User B
| | | |
| INVITE F1 | | |
|--------------->| | |
| INVITE F2 | | |
|--------------->| | |
| INVITE F3 | | |
|--------------->| | |
| INVITE F4 | | |
|--------------->| | |
| INVITE F5 | | |
|--------------->| | |
| INVITE F6 | | |
|--------------->| | |
| INVITE F7 | | |
|--------------->| | |
| CANCEL F8 | | |
|--------------->| | |
| INVITE F9 | |
|-------------------------------->| |
| 407 F10 | |
|<--------------------------------| |
| ACK F11 | |
|-------------------------------->| |
| INVITE F12 | |
|-------------------------------->| INVITE F13 |
| (100) F14 |--------------->|
|<--------------------------------| |
| | 180 F15 |
| 180 F16 |<---------------|
|<--------------------------------| |
| | 200 F17 |
| 200 F18 |<---------------|
|<--------------------------------| |
| ACK F19 | |
|-------------------------------->| ACK F20 |
| |--------------->|
| Both Way RTP Media |
|<================================================>|
| | BYE F21 |
| BYE F22 |<---------------|
|<--------------------------------| |
| 200 F23 | |
|-------------------------------->| 200 F24 |
| |--------------->|
| | |
Johnston, et al. Informational [Page 31]
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In this scenario, User A completes a call to User B via a Proxy
Server. User A is configured for a primary SIP Proxy Server Proxy 1
and a secondary SIP Proxy Server Proxy 2 (Or is able to use DNS SRV
records to locate Proxy 1 and Proxy 2). Proxy 1 is out of service and
does not respond to INVITEs (it is reachable, but unresponsive).
After sending a CANCEL to Proxy 1, User A then completes the call to
User B using Proxy 2.
Message Details
F1 INVITE A -> Proxy 1
INVITE sip:UserB@there.com SIP/2.0
Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy
Call-ID: 12345600@here.com
CSeq: 1 INVITE
Contact: BigGuy
Content-Type: application/sdp
Content-Length: 132
v=0
o=UserA 2890844526 2890844526 IN IP4 here.com
s=Session SDP
c=IN IP4 100.101.102.103
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F2 INVITE A -> Proxy 1
Same as Message F1
F3 INVITE A -> Proxy 1
Same as Message F1
F4 INVITE A -> Proxy 1
Same as Message F1
F5 INVITE A -> Proxy 1
Same as Message F1
Johnston, et al. Informational [Page 32]
Internet Draft SIP Telephony Call Flow Examples July 2000
F6 INVITE A -> Proxy 1
Same as Message F1
F7 INVITE A -> Proxy 1
Same as Message F1
/* User A gives up on the unresponsive proxy and sends a CANCEL. If
any 200 OK responses come back to the INVITE, User A sends an ACK,
then a BYE. */
F8 CANCEL A -> Proxy 1
CANCEL sip:UserB@there.com SIP/2.0
Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy
Call-ID: 12345600@here.com
CSeq: 1 CANCEL
F9 INVITE A -> Proxy 2
INVITE sip:UserB@there.com SIP/2.0
Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy
Call-ID: 12345601@here.com
CSeq: 1 INVITE
Contact: BigGuy
Content-Type: application/sdp
Content-Length: 132
v=0
o=UserA 2890844526 2890844526 IN IP4 here.com
s=Session SDP
c=IN IP4 100.101.102.103
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000
/* Proxy 2 challenges User A for authentication */
F10 407 Proxy Authorization Required Proxy 2 -> User A
Johnston, et al. Informational [Page 33]
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SIP/2.0 407 Proxy Authorization Required
Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy
Call-ID: 12345601@here.com
CSeq: 1 INVITE
Proxy-Authenticate: Digest realm="MCI WorldCom SIP",
domain=_wcom.com_, nonce="1ae6cbe5ea9c8e8df84fqnlec434a359",
opaque="", stale="FALSE", algorithm="MD5"
Content-Length: 0
F11 ACK A -> Proxy 2
ACK sip:UserB@there.com SIP/2.0
Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy
Call-ID: 12345601@here.com
CSeq: 1 INVITE
Content-Length: 0
/* User A responds be re-sending the INVITE with authentication
credentials in it. */
F12 INVITE A -> Proxy 2
INVITE sip:UserB@there.com SIP/2.0
Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy
Call-ID: 12345602@here.com
CSeq: 1 INVITE
Contact: BigGuy
Authorization:Digest username="UserA", realm="MCI WorldCom SIP",
nonce="1ae6cbe5ea9c8e8df84fqnlec434a359", opaque="",
uri="sip:ss2.wcom.com", response="8a880c919d1a52f20a1593e228adf599"
Content-Type: application/sdp
Content-Length: 132
v=0
o=UserA 2890844526 2890844526 IN IP4 here.com
s=Session SDP
c=IN IP4 100.101.102.103
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000
/* Proxy 2 accepts the credentials and forwards the INVITE to User B.
Johnston, et al. Informational [Page 34]
Internet Draft SIP Telephony Call Flow Examples July 2000
Client for A prepares to receive data on port 49172 from the network.
*/
F13 INVITE Proxy 2 -> B
INVITE sip:UserB@there.com SIP/2.0
Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
Via: SIP/2.0/UDP here.com:5060
Record-Route:
From: BigGuy
To: LittleGuy
Call-ID: 12345602@here.com
CSeq: 1 INVITE
Contact: BigGuy
Content-Type: application/sdp
Content-Length: 132
v=0
o=UserA 2890844526 2890844526 IN IP4 here.com
s=Session SDP
c=IN IP4 100.101.102.103
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F14 (100 Trying) Proxy 2 -> User A
SIP/2.0 100 Trying
Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy
Call-ID: 12345602@here.com
CSeq: 1 INVITE
Content-Length: 0
F15 180 Ringing B -> Proxy 2
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy ;tag=314159
Call-ID: 12345602@here.com
CSeq: 1 INVITE
Content-Length: 0
F16 180 Ringing Proxy 2 -> A
Johnston, et al. Informational [Page 35]
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SIP/2.0 180 Ringing
Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy ;tag=314159
Call-ID: 12345602@here.com
CSeq: 1 INVITE
Content-Length: 0
F17 200 OK B -> Proxy 2
SIP/2.0 200 OK
Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
Via: SIP/2.0/UDP here.com:5060
Record-Route:
From: BigGuy
To: LittleGuy ;tag=314159
Call-ID: 12345602@here.com
CSeq: 1 INVITE
Contact: LittleGuy
Content-Type: application/sdp
Content-Length: 134
v=0
o=UserB 2890844527 2890844527 IN IP4 there.com
s=Session SDP
c=IN IP4 110.111.112.113
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F18 200 OK Proxy 2 -> A
SIP/2.0 200 OK
Via: SIP/2.0/UDP here.com:5060
Record-Route:
From: BigGuy
To: LittleGuy ;tag=314159
Call-ID: 12345602@here.com
CSeq: 1 INVITE
Contact: LittleGuy
Content-Type: application/sdp
Content-Length: 134
v=0
o=UserB 2890844527 2890844527 IN IP4 there.com
s=Session SDP
c=IN IP4 110.111.112.113
t=0 0
Johnston, et al. Informational [Page 36]
Internet Draft SIP Telephony Call Flow Examples July 2000
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F19 ACK A -> Proxy 2
ACK sip:UserB@there.com SIP/2.0
Via: SIP/2.0/UDP here.com:5060
Route:
From: BigGuy
To: LittleGuy ;tag=314159
Call-ID: 12345602@here.com
CSeq: 1 ACK
Content-Length: 0
F20 ACK Proxy 2 -> B
ACK sip: UserB@there.com SIP/2.0
Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy ;tag=314159
Call-ID: 12345602@here.com
CSeq: 1 ACK
Content-Length: 0
/* RTP streams are established between A and B */
/* User B Hangs Up with User A. */
F21 BYE User B -> Proxy 2
BYE sip: UserA@here.com SIP/2.0
Via: SIP/2.0/UDP there.com:5060
Route:
From: LittleGuy ;tag=314159
To: BigGuy
Call-ID: 12345602@here.com
CSeq: 1 BYE
Content-Length: 0
F22 BYE Proxy 2 -> User A
BYE sip: UserA@here.com SIP/2.0
Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
Via: SIP/2.0/UDP there.com:5060
From: LittleGuy ;tag=314159
To: BigGuy
Johnston, et al. Informational [Page 37]
Internet Draft SIP Telephony Call Flow Examples July 2000
Call-ID: 12345602@here.com
CSeq: 1 BYE
Content-Length: 0
F23 200 OK User A -> Proxy 2
SIP/2.0 200 OK
Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
Via: SIP/2.0/UDP there.com:5060
From: LittleGuy ;tag=314159
To: BigGuy
Call-ID: 12345602@here.com
CSeq: 1 BYE
Content-Length: 0
F24 200 OK Proxy 2 -> User B
SIP/2.0 200 OK
Via: SIP/2.0/UDP there.com:5060
From: LittleGuy ;tag=314159
To: BigGuy
Call-ID: 12345602@here.com
CSeq: 1 BYE
Content-Length: 0
Johnston, et al. Informational [Page 38]
Internet Draft SIP Telephony Call Flow Examples July 2000
3.1.4 Successful SIP to SIP through SIP Firewall Proxy
User A Proxy 1 Proxy 2 User B
| | | |
| INVITE F1 | | |
|--------------->| INVITE F2 | |
| (100) F3 |--------------->| INVITE F4 |
|<---------------| (100) F5 |--------------->|
| |<---------------| (100) F6 |
| | |<---------------|
| | | 180 F7 |
| | 180 F8 |<---------------|
| 180 F9 |<---------------| |
|<---------------| | 200 F10 |
| | 200 F11 |<---------------|
| 200 F12 |<---------------| |
|<---------------| | |
| ACK F13 | | |
|--------------->| ACK F14 | |
| |--------------->| ACK F15 |
| | |--------------->|
| RTP Media | Both Way RTP Media |
|<==============>|<===============================>|
| BYE F16 | | |
|--------------->| BYE F17 | |
| |--------------->| BYE F18 |
| | |--------------->|
| | | 200 F19 |
| | 200 F20 |<---------------|
| 200 F21 |<---------------| |
|<---------------| | |
| | | |
User A completes a call to User B through a Firewall Proxy and a SIP
Proxy. The signaling message exchange is identical to 3.1.1 but the
media stream setup is not end-to-end - the Firewall proxy terminates
both media streams and bridges them. This is done by the Proxy
modifying the SDP in the INVITE (F1) and 200 OK (F11) messages.
In addition to firewall traversal, this back-to-back User Agent
Client and User Agent Server could be used as part of an anonymizer
service (in which all identifying information on User A would be
removed), or to perform codec media conversion, such as mu-law to A-
law conversion of PCM on an international call.
Message Details
F1 INVITE A -> SIP FW
INVITE sip:UserB@ fwp1.wcom.com SIP/2.0
Johnston, et al. Informational [Page 39]
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Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy
Call-ID: 12345600@here.com
CSeq: 1 INVITE
Contact: BigGuy
Authorization:Digest username="UserA", realm="MCI WorldCom SIP",
nonce="85b4f1cen4341ae6cbe5a3a9c8e88df9", opaque="",
uri="sip:ss1.wcom.com", response="b3f392f9218a328b9294076d708e6815"
Content-Type: application/sdp
Content-Length: 132
v=0
o=UserA 2890844526 2890844526 IN IP4 here.com
s=Session SDP
c=IN IP4 100.101.102.103
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000
/* Client for A prepares to receive data on port 49172 from the
network. */
F2 INVITE SIP FW -> Proxy 1
INVITE sip:UserB@there.com SIP/2.0
Via: SIP/2.0/UDP fwp1.wcom.com:5060;branch=9471385739578.1
Via: SIP/2.0/UDP here.com:5060
Record-Route:
From: BigGuy
To: LittleGuy
Call-ID: 12345600@here.com
CSeq: 1 INVITE
Contact: BigGuy
Authorization:Digest username="UserA", realm="MCI WorldCom SIP",
nonce="85b4f1cen4341ae6cbe5a3a9c8e88df9", opaque="",
uri="sip:ss1.wcom.com", response="b3f392f9218a328b9294076d708e6815"
Content-Type: application/sdp
Content-Length: 134
v=0
o=UserA 2890844526 2890844526 IN IP4 here.com
s=Session SDP
c=IN IP4 200.201.202.203
t=0 0
m=audio 1000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F3 (100 Trying) SIP FW -> A
Johnston, et al. Informational [Page 40]
Internet Draft SIP Telephony Call Flow Examples July 2000
SIP/2.0 100 Trying
Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy
Call-ID: 12345600@here.com
CSeq: 1 INVITE
Content-Length: 0
/* SIP FW prepares to proxy data from port 1000 to
100.101.102.103/49172. Proxy 1 uses a location manager function to
determine where B is located. Based upon location analysis the call
is forwarded to User B */
F4 INVITE Proxy 1 -> B
INVITE sip:UserB@there.com SIP/2.0
Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
Via: SIP/2.0/UDP fwp1.wcom.com:5060;branch=9471385739578.1
Via: SIP/2.0/UDP here.com:5060
Record-Route: ,
From: BigGuy
To: LittleGuy
Call-ID: 12345600@here.com
CSeq: 1 INVITE
Contact: BigGuy
Content-Type: application/sdp
Content-Length: 134
v=0
o=UserA 2890844526 2890844526 IN IP4 here.com
s=Session SDP
c=IN IP4 200.201.202.203
t=0 0
m=audio 1000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F5 (100 Trying) Proxy 1 -> SIP FW
SIP/2.0 100 Trying
Via: SIP/2.0/UDP fwp1.wcom.com:5060;branch=9471385739578.1
Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy
Call-ID: 12345600@here.com
CSeq: 1 INVITE
Content-Length: 0
Johnston, et al. Informational [Page 41]
Internet Draft SIP Telephony Call Flow Examples July 2000
F6 (100 Trying) B -> Proxy 1
SIP/2.0 100 Trying
Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
Via: SIP/2.0/UDP fwp1.wcom.com:5060;branch=9471385739578.1
Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy
Call-ID: 12345600@here.com
CSeq: 1 INVITE
Content-Length: 0
F7 180 Ringing B -> Proxy 1
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
Via: SIP/2.0/UDP fwp1.wcom.com:5060;branch=9471385739578.1
Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy ;tag=314159
Call-ID: 12345600@here.com
CSeq: 1 INVITE
Content-Length: 0
F8 180 Ringing Proxy 1 -> SIP FW
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP fwp1.wcom.com:5060;branch=9471385739578.1
Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy ;tag=314159
Call-ID: 12345600@here.com
CSeq: 1 INVITE
Content-Length: 0
F9 180 Ringing SIP FW -> A
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy ;tag=314159
Call-ID: 12345600@here.com
CSeq: 1 INVITE
Content-Length: 0
F10 200 OK B -> Proxy 1
Johnston, et al. Informational [Page 42]
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SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
Via: SIP/2.0/UDP fwp1.wcom.com:5060;branch=9471385739578.1
Via: SIP/2.0/UDP here.com:5060
Record-Route: ,
From: BigGuy
To: LittleGuy ;tag=314159
Call-ID: 12345600@here.com
CSeq: 1 INVITE
Contact: LittleGuy
Content-Type: application/sdp
Content-Length: 133
v=0
o=UserB 2890844527 2890844527 IN IP4 there.com
s=Session SDP
c=IN IP4 110.111.112.113
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F11 200 OK Proxy 1 -> SIP FW
SIP/2.0 200 OK
Via: SIP/2.0/UDP gw1.wcom.com:5060
Via: SIP/2.0/UDP here.com:5060
Record-Route: ,
From: BigGuy
To: LittleGuy ;tag=314159
Call-ID: 12345600@here.com
CSeq: 1 INVITE
Contact: LittleGuy
Content-Type: application/sdp
Content-Length: 134
v=0
o=UserB 2890844527 2890844527 IN IP4 there.com
s=Session SDP
c=IN IP4 110.111.112.113
t=0 0
m=audio 3456 RTP/AVP 0
a =rtpmap:0 PCMU/8000
F12 200 OK SIP FW -> A
SIP/2.0 200 OK
Johnston, et al. Informational [Page 43]
Internet Draft SIP Telephony Call Flow Examples July 2000
Via: SIP/2.0/UDP here.com:5060
Record-Route: ,
From: BigGuy
To: LittleGuy ;tag=314159
Call-ID: 12345600@here.com
CSeq: 1 INVITE
Contact: LittleGuy
Content-Type: application/sdp
Content-Length: 134
v=0
o=UserB 2890844527 2890844527 IN IP4 there.com
s=Session SDP
c=IN IP4 200.201.202.203
t=0 0
m=audio 1002 RTP/AVP 0
a=rtpmap:0 PCMU/8000
/* The Firewall Proxy prepares to proxy packets from port 1002 to
110.111.112.113/3456 */
F13 ACK A -> SIP FW
ACK sip:UserB@fwp1.wcom.com SIP/2.0
Via: SIP/2.0/UDP here.com:5060
Route: ,
From: BigGuy
To: LittleGuy ;tag=314159
Call-ID: 12345600@here.com
CSeq: 1 ACK
Content-Length: 0
F14 ACK SIP FW -> Proxy 1
ACK sip:UserB@there.com SIP/2.0
Via: SIP/2.0/UDP fwp1.wcom.com:5060;branch=9471385739578.1
Via: SIP/2.0/UDP here.com:5060
Route:
From: BigGuy
To: LittleGuy ;tag=314159
Call-ID: 12345600@here.com
CSeq: 1 ACK
Content-Length: 0
F15 ACK Proxy 1 -> B
Johnston, et al. Informational [Page 44]
Internet Draft SIP Telephony Call Flow Examples July 2000
ACK sip: UserB@there.com SIP/2.0
Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
Via: SIP/2.0/UDP fwp1.wcom.com:5060;branch=9471385739578.1
Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy ;tag=314159
Call-ID: 12345600@here.com
CSeq: 1 ACK
Content-Length: 0
/* RTP streams are established between A and the Firewall Proxy and
between the Firewall Proxy and B*/
/* User A Hangs Up with User B. */
F16 BYE A -> SIP FW
BYE sip: UserB@fwp1.wcom.com SIP/2.0
Via: SIP/2.0/UDP here.com:5060
Route: ,
From: BigGuy
To: LittleGuy ;tag=314159
Call-ID: 12345600@here.com
CSeq: 2 BYE
Content-Length: 0
F17 BYE SIP FW -> Proxy 1
BYE sip: UserB@there.com SIP/2.0
Via: SIP/2.0/UDP fwp1.wcom.com:5060;branch=9471385739578.1
Via: SIP/2.0/UDP here.com:5060
Route:
From: BigGuy
To: LittleGuy ;tag=314159
Call-ID: 12345600@here.com
CSeq: 2 BYE
Content-Length: 0
F18 BYE F18 Proxy 1 -> B
BYE sip: UserB@there.com SIP/2.0
Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
Via: SIP/2.0/UDP fwp1.wcom.com:5060;branch=9471385739578.1
Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy ;tag=314159
Call-ID: 12345600@here.com
Johnston, et al. Informational [Page 45]
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CSeq: 2 BYE
Content-Length: 0
F19 200 OK B -> Proxy 1
SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
Via: SIP/2.0/UDP fwp1.wcom.com:5060;branch=9471385739578.1
Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy ;tag=314159
Call-ID: 12345600@here.com
CSeq: 2 BYE
Content-Length: 0
F20 200 OK Proxy 1 -> SIP FW
SIP/2.0 200 OK
Via: SIP/2.0/UDP fwp1.wcom.com:5060;branch=9471385739578.1
Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy ;tag=314159
Call-ID: 12345600@here.com
CSeq: 2 BYE
Content-Length: 0
F21 200 OK SIP FW -> A
SIP/2.0 200 OK
Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy ;tag=314159
Call-ID: 12345600@here.com
CSeq: 2 BYE
Content-Length: 0
Johnston, et al. Informational [Page 46]
Internet Draft SIP Telephony Call Flow Examples July 2000
3.1.5 Successful SIP to SIP via Redirect and Proxy
User A Redirect Proxy Proxy 2 User B
| | | |
| INVITE F1 | | |
|--------------->| | |
| 302 F2 | | |
|<---------------| | |
| ACK F3 | | |
|--------------->| | |
| INVITE F4 | |
|-------------------------------->| INVITE F5 |
| (100) F6 |--------------->|
|<--------------------------------| (100) F7 |
| |<---------------|
| | 180 F8 |
| 180 F9 |<---------------|
|<--------------------------------| |
| | 200 F10 |
| 200 F11 |<---------------|
|<--------------------------------| |
| ACK F12 | |
|-------------------------------->| ACK F13 |
| |--------------->|
| Both Way RTP Media |
|<================================================>|
| | BYE F14 |
| BYE F15 |<---------------|
|<--------------------------------| |
| 200 F16 | |
|-------------------------------->| 200 F17 |
| |--------------->|
| | |
In this scenario, User A places a call to User B using first a
Redirect server then a Proxy Server. The INVITE message is first
sent to the Redirect Server. The Server returns a 302 Moved
Temporarily response (F2) containing a Contact header with User B's
current SIP address. User A then generates a new INVITE and sends to
User B via the Proxy Server and the call proceeds normally.
The call is terminated when User B sends a BYE message.
Message Details
F1 INVITE A -> Redirect Proxy
INVITE sip:UserB@there.com SIP/2.0
Via: SIP/2.0/UDP here.com:5060
Johnston, et al. Informational [Page 47]
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From: BigGuy
To: LittleGuy
Call-ID: 12345600@here.com
CSeq: 1 INVITE
Contact: BigGuy
Content-Type: application/sdp
Content-Length: 132
v=0
o=UserA 2890844526 2890844526 IN IP4 here.com
s=Session SDP
c=IN IP4 100.101.102.103
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000
/* Client for A prepares to receive data on port 49172 from the
network. */
F2 302 Moved Temporarily Redirect Proxy -> A
SIP/2.0 302 Moved Temporarily
Contact: sip:UserB@everywhere.com
Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy
Call-ID: 12345600@here.com
CSeq: 1 INVITE
Content-Length: 0
F3 ACK A -> Redirect Proxy
ACK sip:UserB@there.com SIP/2.0
Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy
Call-ID: 12345600@here.com
CSeq: 1 INVITE
Content-Length: 0
F4 INVITE A -> Proxy 2
INVITE sip:UserB@everywhere.com SIP/2.0
Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy
Call-ID: 12345600@here.com
CSeq: 2 INVITE
Johnston, et al. Informational [Page 48]
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Contact: BigGuy
Content-Type: application/sdp
Content-Length: 132
v=0
o=UserA 2890844526 2890844526 IN IP4 here.com
s=Session SDP
c=IN IP4 100.101.102.103
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F5 INVITE Proxy 2 -> B
INVITE sip:UserB@everywhere.com SIP/2.0
Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
Via: SIP/2.0/UDP here.com:5060
Record-Route:
From: BigGuy
To: LittleGuy
Call-ID: 12345600@here.com
CSeq: 2 INVITE
Contact: BigGuy
Content-Type: application/sdp
Content-Length: 132
v=0
o=UserA 2890844526 2890844526 IN IP4 here.com
s=Session SDP
c=IN IP4 100.101.102.103
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F6 (100 Trying) Proxy 2 -> A
SIP/2.0 100 Trying
Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy
Call-ID: 12345600@here.com
CSeq: 2 INVITE
Content-Length: 0
F7 (100 Trying) B -> Proxy 2
SIP/2.0 100 Trying
Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
Johnston, et al. Informational [Page 49]
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Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy
Call-ID: 12345600@here.com
CSeq: 2 INVITE
Content-Length: 0
F8 180 Ringing B -> Proxy 2
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy ;tag=314159
Call-ID: 12345600@here.com
CSeq: 2 INVITE
Content-Length: 0
F9 180 Ringing Proxy 2 -> A
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP here.com:5060
From: BigGuy
To: LittleGuy ;tag=314159
Call-ID: 12345600@here.com
CSeq: 2 INVITE
Content-Length: 0
F10 200 OK B -> Proxy 2
SIP/2.0 200 OK
Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
Via: SIP/2.0/UDP here.com:5060
Record-Route:
From: BigGuy