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.
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
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].
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.
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. */
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 |<-------------| |<---------------| |
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
;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
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
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
;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
;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: ...
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
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
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
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
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
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