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RFC 5359

BCP 144
Pages: 170
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Session Initiation Protocol Service Examples

Part 1 of 6, p. 1 to 18
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Network Working Group                                   A. Johnston, Ed.
Request for Comments: 5359                                         Avaya
BCP: 144                                                       R. Sparks
Category: Best Current Practice                                  Tekelec
                                                           C. Cunningham
                                                              S. Donovan
                                                           Cisco Systems
                                                              K. Summers
                                                                   Sonus
                                                            October 2008


              Session Initiation Protocol Service Examples

Status of This Memo

   This document specifies an Internet Best Current Practices for the
   Internet Community, and requests discussion and suggestions for
   improvements.  Distribution of this memo is unlimited.

Abstract

   This document gives examples of Session Initiation Protocol (SIP)
   services.  This covers most features offered in so-called IP Centrex
   offerings from local exchange carriers and PBX (Private Branch
   Exchange) features.  Most of the services shown in this document are
   implemented in the SIP user agents, although some require the
   assistance of a SIP proxy.  Some require some extensions to SIP
   including the REFER, SUBSCRIBE, and NOTIFY methods and the Replaces
   and Join header fields.  These features are not intended to be an
   exhaustive set, but rather show implementations of common features
   likely to be implemented on SIP IP telephones in a business
   environment.

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Table of Contents

   1. Overview ........................................................3
      1.1. Legend for Message Flows ...................................4
   2. Service Examples ................................................6
      2.1. Call Hold ..................................................6
      2.2. Consultation Hold .........................................19
      2.3. Music on Hold .............................................38
      2.4. Transfer - Unattended .....................................50
      2.5. Transfer - Attended .......................................58
      2.6. Transfer - Instant Messaging ..............................71
      2.7. Call Forwarding Unconditional .............................77
      2.8. Call Forwarding - Busy ....................................84
      2.9. Call Forwarding - No Answer ...............................92
      2.10. 3-Way Conference - Third Party Is Added .................101
      2.11. 3-Way Conference - Third Party Joins ....................107
      2.12. Find-Me .................................................113
      2.13. Call Management (Incoming Call Screening) ...............125
      2.14. Call Management (Outgoing Call Screening) ...............132
      2.15. Call Park ...............................................135
      2.16. Call Pickup .............................................147
      2.17. Automatic Redial ........................................154
      2.18. Click to Dial ...........................................163
   3. Security Considerations .......................................166
   4. Acknowledgements ..............................................166
   5. References ....................................................167
      5.1. Normative References .....................................167
      5.2. Informative References ...................................168

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1.  Overview

   This document provides example call flows detailing a SIP
   implementation of the following traditional telephony services:

           Call Hold                        3-Way Conference
           Consultation Hold                Find-Me
           Music on Hold                    Incoming Call Screening
           Unattended Transfer              Outgoing Call Screening
           Attended Transfer                Call Park
           Instant Messaging Transfer       Call Pickup
           Unconditional Call Forwarding    Automatic Redial
           Call Forwarding on Busy          Click to Dial
           Call Forwarding on No Answer

   Note that the Single Line Extension call flow has been removed from
   this document and will be covered in a separate document.

   The call flows shown in this document were developed in the design of
   a SIP IP communications network.  They represent an example set of
   so-called IP Centrex services or PBX services.

   It is the hope of the authors that this document will be useful for
   SIP implementers, designers, and protocol researchers alike and will
   help further the goal of a standard implementation of RFC 3261
   [RFC3261] and some of its extensions.

   These flows represent carefully checked and working group reviewed
   scenarios of SIP service examples as a companion to the
   specifications.

   These call flows are based on the current version 2.0 of SIP in RFC
   3261 [RFC3261] with Session Description Protocol (SDP) usage
   described in RFC 3264 [RFC3264].  Other RFCs also form part of the
   SIP standard and are used and referenced in these call flows.

   The SIP specification and the other referenced documents are
   definitive as far as protocol issues are concerned.  Also, these
   flows do not represent the only way to implement these services --
   other approaches such as 3pcc (Third Party Call Control) [RFC3725] or
   Back-to-Back User Agents (B2BUAs) can be used.  This specification
   does not preclude these or other approaches for implementing such
   services.  The peer-to-peer design and principles of these service
   examples are described in the Multiparty Framework document
   [FRAMEWORK].

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   These flows assume the functionality described in the SIP Call Flow
   Examples document [RFC3665], which explores basic SIP behavior.  Some
   of the scenarios described herein make use of the SIP method
   extension REFER [RFC3515], the SIP header extension Replaces
   [RFC3891], and the SIP header extension Join [RFC3911].  The SIP
   Events document [RFC3265] describes the use of SUBSCRIBE and NOTIFY,
   while the SIP Dialog Event Package document [RFC4235] describes the
   dialog event package.  Some examples make use of the GRUU (Globally
   Routable User Agent URI) extension [GRUU].

   These flows were prepared assuming a network of proxies, registrars,
   and other SIP servers.  The use of Secure SIP URIs (sips) is shown
   throughout this document, implying TLS transport on each hop with
   assumed certificate validation.  However, other security approaches
   can be used.  The use of Digest authentication is shown in some
   examples.

   The emphasis in these call flows is the SIP signaling exchange.  As a
   result, only very simple SDP offer/answer exchanges are shown with
   audio media.  These flows apply equally well for other media and
   multimedia sessions.  For more advanced examples of SDP offer/answer
   exchanges, refer to [RFC4317].

   Each call flow is presented with a textual description of the
   scenario, a message flow diagram showing the messages exchanged
   between separate network elements, and the detailed contents of each
   message shown in the diagram.

   For simplicity in reading and editing the document, there are a
   number of differences between some of the examples and actual SIP
   messages.  For example, the HTTP Digest responses are not actual MD5
   encodings.  Call-IDs are often repeated, and CSeq counts often begin
   at 1.  Header fields are usually shown in the same order.  Usually
   only the minimum required header field set is shown.  Also, message
   body content lengths are often not calculated, but instead shown as
   "..." where the actual octet count would be.

1.1.  Legend for Message Flows

   Dashed lines (---) represent control messages that are mandatory to
   the call scenario.  These control messages can be SIP signaling.

   Double dashed lines (===) represent media paths between network
   elements.

   Messages with parentheses around the name represent optional control
   messages.

Top      ToC       Page 5 
   Messages are identified in the figures as F1, F2, etc.  This
   references the message details in the table that follows the figure.

   Lines longer than 72 characters are handled using the <allOneLine>
   convention defined in Section 2.1 of RFC 4475 [RFC4475].

   Comments in the message details are shown in the following form:

   /* Comments. */

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2.  Service Examples

2.1.  Call Hold

            Alice           Proxy            Bob
             |    INVITE F1   |              |
             |--------------->|              |
             |                |   INVITE F2  |
             |(100 Trying) F3 |------------->|
             |<---------------|              |
             |                |180 Ringing F4|
             | 180 Ringing F5 |<-------------|
             |<---------------|              |
             |                |  200 OK F6   |
             |    200 OK F7   |<-------------|
             |<---------------|              |
             |     ACK F8     |              |
             |--------------->|    ACK F9    |
             |                |------------->|
             |    Both way RTP Established   |
             |<=============================>|
             |                |INVITE(hold) F10
             |INVITE(hold) F11|<-------------|
             |<---------------|              |
             |    200 OK F12  |              |
             |--------------->|   200 OK F13 |
             |                |------------->|
             |                |     ACK F14  |
             |     ACK F15    |<-------------|
             |<---------------|              |
             |           No RTP Sent!        |
             |                |  INVITE F16  |
             |   INVITE F17   |<-------------|
             |<---------------|              |
             |   200 OK F18   |              |
             |--------------->|  200 OK F19  |
             |                |------------->|
             |                |    ACK F20   |
             |     ACK F21    |<-------------|
             |<---------------|              |
             |    Both way RTP Established   |
             |<=============================>|
             |     BYE F22    |              |
             |--------------->|   BYE F23    |
             |                |------------->|
             |                |   200 OK F24 |
             |    200 OK F25  |<-------------|
             |<---------------|              |

Top      ToC       Page 7 
   In this scenario, Alice calls Bob, then Bob places the call on hold.
   Bob then takes the call off hold, then Alice hangs up the call.  Note
   that hold is unidirectional in nature.  However, a UA that places the
   other party on hold will generally also stop sending media, resulting
   in no media exchange between the UAs.  Older UAs may set the
   connection address to 0.0.0.0 when initiating hold.  However, this
   behavior has been deprecated in favor or using the a=inactive SDP
   attribute if no media is sent, or the a=sendonly attribute if media
   is still sent.

   Also note the use of the rendering feature tag defined in RFC 4235
   [RFC4235] used in F10 and F11 to indicate that Bob's UA is no longer
   rendering media to Bob, i.e., that Bob has placed the call on hold.

   Message Details

      F1 INVITE Alice -> Proxy 1

      INVITE sips:bob@biloxi.example.com SIP/2.0
      Via: SIP/2.0/TLS client.atlanta.example.com:5061
       ;branch=z9hG4bK74bf9
      Max-Forwards: 70
      From: Alice <sips:alice@atlanta.example.com>;tag=1234567
      To: Bob <sips:bob@biloxi.example.com>
      Call-ID: 12345601@atlanta.example.com
      CSeq: 1 INVITE
      Contact: <sips:alice@client.atlanta.example.com>
      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
      Supported: replaces
      Content-Type: application/sdp
      Content-Length: ...

      v=0
      o=alice 2890844526 2890844526 IN IP4 client.atlanta.example.com
      s=
      c=IN IP4 client.atlanta.example.com
      t=0 0
      m=audio 49170 RTP/AVP 0
      a=rtpmap:0 PCMU/8000


      F2 INVITE Proxy 1 -> Bob

      INVITE sips:bob@client.biloxi.example.com SIP/2.0
      Via: SIP/2.0/TLS ss1.example.com:5061
       ;branch=z9hG4bK83749.1
      Via: SIP/2.0/TLS client.atlanta.example.com:5061

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       ;branch=z9hG4bK74bf9
       ;received=192.0.2.103
      Record-Route: <sips:ss1.example.com;lr>
      Max-Forwards: 69
      From: Alice <sips:alice@atlanta.example.com>;tag=1234567
      To: Bob <sips:bob@biloxi.example.com>
      Call-ID: 12345601@atlanta.example.com
      CSeq: 1 INVITE
      Contact: <sips:alice@client.atlanta.example.com>
      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
      Supported: replaces
      Content-Type: application/sdp
      Content-Length: ...

      v=0
      o=alice 2890844526 2890844526 IN IP4 client.atlanta.example.com
      s=
      c=IN IP4 client.atlanta.example.com
      t=0 0
      m=audio 49170 RTP/AVP 0
      a=rtpmap:0 PCMU/8000


      F3 (100 Trying) Proxy 1 -> Alice

      SIP/2.0 100 Trying
      Via: SIP/2.0/TLS client.atlanta.example.com:5061
       ;branch=z9hG4bK74bf9
       ;received=192.0.2.103
      From: Alice <sips:alice@atlanta.example.com>;tag=1234567
      To: Bob <sips:bob@biloxi.example.com>
      Call-ID: 12345601@atlanta.example.com
      CSeq: 1 INVITE
      Content-Length: 0


      F4 180 Ringing Bob -> Proxy 1

      SIP/2.0 180 Ringing
      Via: SIP/2.0/TLS ss1.example.com:5061
       ;branch=z9hG4bK83749.1
       ;received=192.0.2.54
      Via: SIP/2.0/TLS client.atlanta.example.com:5061
       ;branch=z9hG4bK74bf9
       ;received=192.0.2.103
      Record-Route: <sips:ss1.example.com;lr>
      From: Alice <sips:alice@atlanta.example.com>;tag=1234567
      To: Bob <sips:bob@biloxi.example.com>;tag=314159

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      Call-ID: 12345601@atlanta.example.com
      CSeq: 1 INVITE
      Contact: <sips:bob@client.biloxi.example.com>
      Content Length:0


      F5 180 Ringing Proxy 1 -> Alice

      SIP/2.0 180 Ringing
      Via: SIP/2.0/TLS client.atlanta.example.com:5061
       ;branch=z9hG4bK74bf9
       ;received=192.0.2.103
      Record-Route: <sips:ss1.example.com;lr>
      From: Alice <sips:alice@atlanta.example.com>;tag=1234567
      To: Bob <sips:bob@biloxi.example.com>;tag=314159
      Call-ID: 12345601@atlanta.example.com
      CSeq: 1 INVITE
      Contact: <sips:bob@client.biloxi.example.com>
      Content Length: 0


      F6 200 OK Bob -> Proxy 1

      SIP/2.0 200 OK
      Via: SIP/2.0/TLS ss1.example.com:5061
       ;branch=z9hG4bK83749.1
       ;received=192.0.2.54
      Via: SIP/2.0/TLS client.atlanta.example.com:5061
       ;branch=z9hG4bK74bf9
       ;received=192.0.2.103
      Record-Route: <sips:ss1.example.com;lr>
      From: Alice <sips:alice@atlanta.example.com>;tag=1234567
      To: Bob <sips:bob@biloxi.example.com>;tag=314159
      Call-ID: 12345601@atlanta.example.com
      CSeq: 1 INVITE
      Contact: <sips:bob@client.biloxi.example.com>
      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
      Supported: replaces
      Content-Type: application/sdp
      Content-Length: ...

      v=0
      o=bob 2890844527 2890844527 IN IP4 client.biloxi.example.com
      s=
      c=IN IP4 client.biloxi.example.com
      t=0 0
      m=audio 3456 RTP/AVP 0
      a=rtpmap:0 PCMU/8000

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      F7 200 OK Proxy 1 -> Alice

      SIP/2.0 200 OK
      Via: SIP/2.0/TLS client.atlanta.example.com:5061
       ;branch=z9hG4bK74bf9
       ;received=192.0.2.103
      Record-Route: <sips:ss1.example.com;lr>
      From: Alice <sips:alice@atlanta.example.com>;tag=1234567
      To: Bob <sips:bob@biloxi.example.com>;tag=314159
      Call-ID: 12345601@atlanta.example.com
      CSeq: 1 INVITE
      Contact: <sips:bob@client.biloxi.example.com>
      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
      Supported: replaces
      Content-Type: application/sdp
      Content-Length: ...

      v=0
      o=bob 2890844527 2890844527 IN IP4 client.biloxi.example.com
      s=
      c=IN IP4 client.biloxi.example.com
      t=0 0
      m=audio 3456 RTP/AVP 0
      a=rtpmap:0 PCMU/8000


      F8 ACK Alice -> Proxy 1

      ACK sips:bob@client.biloxi.example.com SIP/2.0
      Via: SIP/2.0/TLS client.atlanta.example.com:5061
       ;branch=z9hG4bK74bf92
      Route: <sips:ss1.example.com;lr>
      Max-Forwards: 70
      From: Alice <sips:alice@atlanta.example.com>;tag=1234567
      To: Bob <sips:bob@biloxi.example.com>;tag=314159
      Call-ID: 12345601@atlanta.example.com
      CSeq: 1 ACK
      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
      Supported: replaces
      Content-Length: 0


      F9 ACK Proxy 1 -> Bob

      ACK sips:bob@client.biloxi.example.com SIP/2.0
      Via: SIP/2.0/TLS ss1.example.com:5061
       ;branch=z9hG4bK837492.1
      Via: SIP/2.0/TLS client.atlanta.example.com:5061

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       ;branch=z9hG4bK74bf92
       ;received=192.0.2.103
      Max-Forwards: 69
      From: Alice <sips:alice@atlanta.example.com>;tag=1234567
      To: Bob <sips:bob@biloxi.example.com>;tag=314159
      Call-ID: 12345601@atlanta.example.com
      CSeq: 1 ACK
      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
      Supported: replaces
      Content-Length: 0


      /* Bob places Alice on hold.  Note that the version is
         incremented in the o= field of the SDP. */

      F10 INVITE Bob -> Proxy 1

      INVITE sips:alice@client.atlanta.example.com SIP/2.0
      Via: SIP/2.0/TLS client.biloxi.example.com:5061
       ;branch=z9hG4bKnashds7
      Route: <sips:ss1.example.com;lr>
      Max-Forwards: 70
      From: Bob <sips:bob@biloxi.example.com>;tag=314159
      To: Alice <sips:alice@atlanta.example.com>;tag=1234567
      Call-ID: 12345601@atlanta.example.com
      CSeq: 1 INVITE
      Contact: <sips:bob@client.biloxi.example.com>;+sip.rendering="no"
      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
      Supported: replaces
      Content-Type: application/sdp
      Content-Length: ...

      v=0
      o=bob 2890844527 2890844528 IN IP4 client.biloxi.example.com
      s=
      c=IN IP4 client.biloxi.example.com
      t=0 0
      m=audio 3456 RTP/AVP 0
      a=rtpmap:0 PCMU/8000
      a=sendonly


      F11 INVITE Proxy 1 -> Alice

      INVITE sips:alice@client.atlanta.example.com SIP/2.0
      Via: SIP/2.0/TLS ss1.example.com:5061
       ;branch=z9hG4bK83749.1
      Via: SIP/2.0/TLS client.biloxi.example.com:5061

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       ;branch=z9hG4bKnashds7
       ;received=192.0.2.105
      Record-Route: <sips:ss1.example.com;lr>
      Max-Forwards: 69
      From: Bob <sips:bob@biloxi.example.com>;tag=314159
      To: Alice <sips:alice@atlanta.example.com>;tag=1234567
      Call-ID: 12345601@atlanta.example.com
      CSeq: 1 INVITE
      Contact: <sips:bob@client.biloxi.example.com>;+sip.rendering="no"
      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
      Supported: replaces
      Content-Type: application/sdp
      Content-Length: ...

      v=0
      o=bob 2890844527 2890844528 IN IP4 client.biloxi.example.com
      s=
      c=IN IP4 client.biloxi.example.com
      t=0 0
      m=audio 3456 RTP/AVP 0
      a=rtpmap:0 PCMU/8000
      a=sendonly


      /* Alice replies to hold. */

      F12 200 OK Alice -> Proxy 1

      SIP/2.0 200 OK
      Via: SIP/2.0/TLS ss1.example.com:5061
       ;branch=z9hG4bK83749.1
       ;received=192.0.2.54
      Via: SIP/2.0/TLS client.biloxi.example.com:5061
       ;branch=z9hG4bKnashds7
       ;received=192.0.2.105
      Record-Route: <sips:ss1.example.com;lr>
      From: Bob <sips:bob@biloxi.example.com>;tag=314159
      To: Alice <sips:alice@atlanta.example.com>;tag=1234567
      Call-ID: 12345601@atlanta.example.com
      CSeq: 1 INVITE
      Contact: <sips:alice@client.atlanta.example.com>
      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
      Supported: replaces
      Content-Type: application/sdp
      Content-Length: ...

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      v=0
      o=alice 2890844526 2890844527 IN IP4 client.atlanta.example.com
      s=
      c=IN IP4 client.atlanta.example.com
      t=0 0
      m=audio 49170 RTP/AVP 0
      a=rtpmap:0 PCMU/8000
      a=recvonly


      F13 200 OK Proxy 1 -> Bob

      SIP/2.0 200 OK
      Via: SIP/2.0/TLS client.biloxi.example.com:5061
       ;branch=z9hG4bKnashds7
       ;received=192.0.2.105
      Record-Route: <sips:ss1.example.com;lr>
      From: Bob <sips:bob@biloxi.example.com>;tag=314159
      To: Alice <sips:alice@atlanta.example.com>;tag=1234567
      Call-ID: 12345601@atlanta.example.com
      CSeq: 1 INVITE
      Contact: <sips:alice@client.atlanta.example.com>
      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
      Supported: replaces
      Content-Type: application/sdp
      Content-Length: ...

      v=0
      o=alice 2890844526 2890844527 IN IP4 client.atlanta.example.com
      s=
      c=IN IP4 client.atlanta.example.com
      t=0 0
      m=audio 49170 RTP/AVP 0
      a=rtpmap:0 PCMU/8000
      a=recvonly


      F14 ACK Bob -> Proxy 1

      ACK sips:alice@client.atlanta.example.com SIP/2.0
      Via: SIP/2.0/TLS client.biloxi.example.com:5061
       ;branch=z9hG4bKnashds72
      Route: <sips:ss1.example.com;lr>
      Max-Forwards: 70
      From: Bob <sips:bob@biloxi.example.com>;tag=314159
      To: Alice <sips:alice@atlanta.example.com>;tag=1234567
      Call-ID: 12345601@atlanta.example.com
      CSeq: 1 ACK

Top      ToC       Page 14 
      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
      Supported: replaces
      Content-Length: 0


      F15 ACK Proxy 1 -> Alice

      ACK sips:alice@client.atlanta.example.com SIP/2.0
      Via: SIP/2.0/TLS ss1.example.com:5061
       ;branch=z9hG4bK83749.1
      Via: SIP/2.0/TLS client.biloxi.example.com:5061
       ;branch=z9hG4bKnashds72
       ;received=192.0.2.105
      Max-Forwards: 69
      From: Bob <sips:bob@biloxi.example.com>;tag=314159
      To: Alice <sips:alice@atlanta.example.com>;tag=1234567
      Call-ID: 12345601@atlanta.example.com
      CSeq: 1 ACK
      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
      Supported: replaces
      Content-Length: 0


      /* Bob takes the call off hold. */

      F16 INVITE Bob -> Proxy 1

      INVITE sips:alice@client.atlanta.example.com SIP/2.0
      Via: SIP/2.0/TLS client.biloxi.example.com:5061
       ;branch=z9hG4bKnashds73
      Route: <sips:ss1.example.com;lr>
      Max-Forwards: 70
      From: Bob <sips:bob@biloxi.example.com>;tag=314159
      To: Alice <sips:alice@atlanta.example.com>;tag=1234567
      Call-ID: 12345601@atlanta.example.com
      CSeq: 2 INVITE
      Contact: <sips:bob@client.biloxi.example.com>
      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
      Supported: replaces
      Content-Type: application/sdp
      Content-Length: ...

      v=0
      o=bob 2890844527 2890844529 IN IP4 client.biloxi.example.com
      s=
      c=IN IP4 client.biloxi.example.com

Top      ToC       Page 15 
      t=0 0
      m=audio 3456 RTP/AVP 0
      a=rtpmap:0 PCMU/8000


      F17 INVITE Proxy 1 -> Alice

      INVITE sips:alice@client.atlanta.example.com SIP/2.0
      Via: SIP/2.0/TLS ss1.example.com:5061
       ;branch=z9hG4bK837493.1
      Via: SIP/2.0/TLS client.biloxi.example.com:5061
       ;branch=z9hG4bKnashds73
       ;received=192.0.2.105
      Record-Route: <sips:ss1.example.com;lr>
      Max-Forwards: 69
      From: Bob <sips:bob@biloxi.example.com>;tag=314159
      To: Alice <sips:alice@atlanta.example.com>;tag=1234567
      Call-ID: 12345601@atlanta.example.com
      CSeq: 2 INVITE
      Contact: <sips:bob@client.biloxi.example.com>
      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
      Supported: replaces
      Content-Type: application/sdp
      Content-Length: ...

      v=0
      o=bob 2890844527 2890844529 IN IP4 client.biloxi.example.com
      s=
      c=IN IP4 client.biloxi.example.com
      t=0 0
      m=audio 3456 RTP/AVP 0
      a=rtpmap:0 PCMU/8000


      F18 200 OK Alice -> Proxy 1

      SIP/2.0 200 OK
      Via: SIP/2.0/TLS ss1.example.com:5061
       ;branch=z9hG4bK837493.1
       ;received=192.0.2.54
      Via: SIP/2.0/TLS client.biloxi.example.com:5061
       ;branch=z9hG4bKnashds73
       ;received=192.0.2.105
      Record-Route: <sips:ss1.example.com;lr>
      From: Bob <sips:bob@biloxi.example.com>;tag=314159
      To: Alice <sips:alice@atlanta.example.com>;tag=1234567
      Call-ID: 12345601@atlanta.example.com
      CSeq: 2 INVITE

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      Contact: <sips:alice@client.atlanta.example.com>
      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
      Supported: replaces
      Content-Type: application/sdp
      Content-Length: ...

      v=0
      o=alice 2890844526 2890844528 IN IP4 client.atlanta.example.com
      s=
      c=IN IP4 client.atlanta.example.com
      t=0 0
      m=audio 49170 RTP/AVP 0
      a=rtpmap:0 PCMU/8000


      F19 200 OK Proxy 1 -> Bob

      SIP/2.0 200 OK
      Via: SIP/2.0/TLS client.biloxi.example.com:5061
       ;branch=z9hG4bKnashds73
       ;received=192.0.2.105
      Record-Route: <sips:ss1.example.com;lr>
      From: Bob <sips:bob@biloxi.example.com>;tag=314159
      To: Alice <sips:alice@atlanta.example.com>;tag=1234567
      Call-ID: 12345601@atlanta.example.com
      CSeq: 2 INVITE
      Contact: <sips:alice@client.atlanta.example.com>
      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
      Supported: replaces
      Content-Type: application/sdp
      Content-Length: ...

      v=0
      o=alice 2890844526 2890844528 IN IP4 client.atlanta.example.com
      s=
      c=IN IP4 client.atlanta.example.com
      t=0 0
      m=audio 49170 RTP/AVP 0
      a=rtpmap:0 PCMU/8000


      F20 ACK Bob -> Proxy 1

      ACK sips:alice@client.atlanta.example.com SIP/2.0
      Via: SIP/2.0/TLS client.biloxi.example.com:5061
       ;branch=z9hG4bKnashds74
      Route: <sips:ss1.example.com;lr>
      Max-Forwards: 70

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      From: Bob <sips:bob@biloxi.example.com>;tag=314159
      To: Alice <sips:alice@atlanta.example.com>;tag=1234567
      Call-ID: 12345601@atlanta.example.com
      CSeq: 2 ACK
      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
      Supported: replaces
      Content-Length: 0


      F21 ACK Proxy 1 -> Alice

      ACK sips:alice@client.atlanta.example.com SIP/2.0
      Via: SIP/2.0/TLS ss1.example.com:5061
       ;branch=z9hG4bK837494.1
      Via: SIP/2.0/TLS client.biloxi.example.com:5061
       ;branch=z9hG4bKnashds74
       ;received=192.0.2.105
      Max-Forwards: 69
      From: Bob <sips:bob@biloxi.example.com>;tag=314159
      To: Alice <sips:alice@atlanta.example.com>;tag=1234567
      Call-ID: 12345601@atlanta.example.com
      CSeq: 2 ACK
      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
      Supported: replaces
      Content-Length: 0


      /* RTP Media stream re-established.  Alice disconnects. */

      F22 BYE Alice -> Proxy 1

      BYE sips:bob@client.biloxi.example.com SIP/2.0
      Via: SIP/2.0/TLS client.atlanta.example.com:5061
       ;branch=z9hG4bK74bf97
      Route: <sips:ss1.example.com;lr>
      Max-Forwards: 70
      From: Alice <sips:alice@atlanta.example.com>;tag=1234567
      To: Bob <sips:bob@biloxi.example.com>;tag=314159
      Call-ID: 12345601@atlanta.example.com
      CSeq: 2 BYE
      Content-Length: 0


      F23 BYE Proxy 1 -> Bob

      BYE sips:bob@client.biloxi.example.com SIP/2.0
      Via: SIP/2.0/TLS ss1.example.com:5061
       ;branch=z9hG4bK837497.1

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      Via: SIP/2.0/TLS client.atlanta.example.com:5061
       ;branch=z9hG4bK74bf97
       ;received=192.0.2.103
      Max-Forwards: 69
      From: Alice <sips:alice@atlanta.example.com>;tag=1234567
      To: Bob <sips:bob@biloxi.example.com>;tag=314159
      Call-ID: 12345601@atlanta.example.com
      CSeq: 2 BYE
      Content-Length: 0


      F24 200 OK Bob -> Proxy 1

      SIP/2.0 200 OK
      Via: SIP/2.0/TLS ss1.example.com:5061
       ;branch=z9hG4bK837497.1
       ;received=192.0.2.54
      Via: SIP/2.0/TLS client.atlanta.example.com:5061
       ;branch=z9hG4bK74bf97
       ;received=192.0.2.103
      From: Alice <sips:alice@atlanta.example.com>;tag=1234567
      To: Bob <sips:bob@biloxi.example.com>;tag=314159
      Call-ID: 12345601@atlanta.example.com
      CSeq: 2 BYE
      Content-Length: 0


      F25 200 OK Proxy 1 -> Alice

      SIP/2.0 200 OK
      Via: SIP/2.0/TLS client.atlanta.example.com:5061
       ;branch=z9hG4bK74bf97
       ;received=192.0.2.103
      From: Alice <sips:alice@atlanta.example.com>;tag=1234567
      To: Bob <sips:bob@biloxi.example.com>;tag=314159
      Call-ID: 12345601@atlanta.example.com
      CSeq: 2 BYE
      Content-Length: 0


Next RFC Part