Session Initiation Protocol (SIP Tutorial: SIP to ISDN Q.

931 Call Flow (Brief)) SIP Subscriber Network SIP Client VOIP Network Company Network Alice Proxy 1 GW 1 PBX C

EventHelix.com/EventStudio 2.5 10-Jun-05 22:14 (Page 1)

This call flow diagram was generated with EventStudio Sequence Diagram Designer 2.5 (http://www.EventHelix.com/EventStudio).

LEG: Brief
This article is based on the call flow presented in http://www.iptel.org/info/players/ietf/callflows/draft-ietf-sipping-pstn-call-flows-02.txt and is reproduced here as per the copyright statement at the end of this document. Alice is a SIP device while Carol is connected via a Gateway (GW 1) to a PBX. The PBX connection is via a ISDN trunk group.

allocate Port 49172

Alice's PC allocates a port for receiving RTP data. This port number will be included in the SIP Invite. Alice dials Carol's telephone number (918-555-3333) which is globalized and put into a SIP URI. The message contains information about the RTP port number and the supported voice codecs. Proxy 1 indicates to the SIP client that it is trying to establish the call. Proxy 1 looks up the telephone number and locates the gateway that serves Carol. Carol is identified by her extension (444-3333) in the Request-URI sent to GW 1. The host portion of the Request-URI in the INVITE is used to identify the context (customer, trunk group, or line) in which the private number 444-3333 is valid. Otherwise, this INVITE message could get forwarded by GW 1 and the context of the digits could become lost and the call unroutable. GW 1 indicates to the Proxy that it is trying to establish the call. The GW routes the call. Since Carol is served by an ISDN PBX, the Gateway initiates a Q.931 call setup with the PBX. The ISDN PBX responds with Call Proceeding. This message indicates that the call is in the process of being setup. The ISDN PBX passes call progress information to the Gateway. This message indicates that the called subscriber is being rung. The Gateway sends the Ringing indication back to the proxy. The proxy forwards the ringing indication to Alice's PC. Carol has answered the call. This results in Q.931 CONNECT message being sent to the Gateway. The Gateway replies with Connect Ack. The Gateway allocates a port for receiving RTP data from Alice's PC. The port information will be passed to originating subscriber via the "SIP 200 OK" response. The Gateway indicates to the Proxy that the call is successful. The RTP audio receive port information is also passed in this message. The Proxy forwards the message to Alice's PC. Alice's PC acknowledges the message. SIP ACK The Proxy forwards the ack to the Gateway.

SIP INVITE
Calling #, Called #, Contact, Media Information

SIP 100 Trying

Identify the Gateway that servers Carol

SIP INVITE

SIP 100 Trying Q.931 SETUP Q.931 CALL PROCEEDING Q.931 PROGRESS

SIP 180 Ringing SIP 180 Ringing Q.931 CONNECT Q.931 CONNECT ACK
allocate Port 3456

SIP 200 OK
Media information

SIP 200 OK SIP ACK

Two way voice is active at this time. Alice and Carol are talking. Alice Hangs Up with Carol.

Session Initiation Protocol (SIP Tutorial: SIP to ISDN Q.931 Call Flow (Brief)) SIP Subscriber Network SIP Client VOIP Network Company Network Alice Proxy 1 GW 1 PBX C SIP BYE SIP BYE Q.931 DISCONNECT SIP 200 OK SIP 200 OK
free Port 49172

EventHelix.com/EventStudio 2.5 10-Jun-05 22:14 (Page 2)

SIP BYE signals the release of the call. The Bye is forwarded to the Gateway. The Gateway initiates the call release on SS7 side. The Gateway acknowledges the BYE to the Proxy with an 200 OK respponse code. The Proxy forwards the ack to Alice's PC.

Q.931 RELEASE
free Port 3456

The ISDN PBX indicates to the Gateway that it is releasing the call.

Q.931 RELEASE COMPLETE

The Gateway acknowledges the call release of the call with the Release Complete message.

Session Initiation Protocol (SIP Tutorial: SIP to ISDN Q.931 Call Flow (Detailed)) SIP Subscriber Network SIP Client VOIP Network Company Network Alice Proxy 1 GW 1 PBX C

EventHelix.com/EventStudio 2.5 10-Jun-05 22:14 (Page 3)

This call flow diagram was generated with EventStudio Sequence Diagram Designer 2.5 (http://www.EventHelix.com/EventStudio).

LEG: Detailed
Alice is a SIP device while Carol is connected via a Gateway (GW 1) to a PBX. The PBX connection is via a ISDN trunk group.

allocate Port 49172

Alice's PC allocates a port for receiving RTP data. This port number will be included in the SIP Invite. Alice dials Carol's telephone number (918-555-3333) which is globalized and put into a SIP URI. The message contains information about the RTP port number and the supported voice codecs.

SIP INVITE
Calling #, Called #, Contact, Media Information
INVITE sips:[email protected];user=phone SIP/2.0 Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9 Max-Forwards: 70 From: Alice <sips:[email protected];user=phone> ;tag=9fxced76sl To: Carol <sips:[email protected];user=phone> Call-ID: [email protected] CSeq: 2 INVITE Contact: <sips:[email protected]> Proxy-Authorization: Digest username="alice", realm="a.example.com", nonce="qo0dc3a5ab22aa931904badfa1cf5j9h", opaque="", uri="sips:[email protected];user=phone", response="6c792f5c9fa360358b93c7fb826bf550" Content-Type: application/sdp Content-Length: 154 v=0 o=alice 2890844526 2890844526 IN IP4 client.a.example.com s=c=IN IP4 client.a.example.com t=0 0 m=audio 49172 RTP/AVP 0 a=rtpmap:0 PCMU/8000

SIP 100 Trying

Proxy 1 indicates to the SIP client that it is trying to establish the call.

SIP/2.0 100 Trying Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9;received=192.0.2.101 From: Alice <sips:[email protected];user=phone>;tag=9fxced76sl To: Carol <sips:[email protected];user=phone> Call-ID: [email protected] CSeq: 2 INVITE Content-Length: 0

Identify the Gateway that servers Carol

SIP INVITE

Proxy 1 looks up the telephone number and locates the gateway that serves Carol. Carol is identified by her extension (444-3333) in the Request-URI sent to GW 1. The host portion of the Request-URI in the INVITE is used to identify the context (customer, trunk group, or line) in which the private number 444-3333 is valid. Otherwise, this INVITE message could get forwarded by GW 1 and the context of the digits could become lost and the call unroutable.

INVITE sips:[email protected] SIP/2.0 Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1 Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9;received=192.0.2.101 Max-Forwards: 69 Record-Route: <sips:ss1.a.example.com;lr> From: Alice <sips:[email protected];user=phone>;tag=9fxced76sl To: Carol <sips:[email protected];user=phone> Call-ID: [email protected] CSeq: 2 INVITE Contact: <sips:[email protected]> Content-Type: application/sdp Content-Length: 154 v=0 o=alice 2890844526 2890844526 IN IP4 client.a.example.com s=c=IN IP4 client.a.example.com t=0 0 m=audio 49172 RTP/AVP 0 a=rtpmap:0 PCMU/8000

SIP 100 Trying

GW 1 indicates to the Proxy that it is trying to establish the call.

Session Initiation Protocol (SIP Tutorial: SIP to ISDN Q.931 Call Flow (Detailed)) SIP Subscriber Network SIP Client VOIP Network Company Network Alice Proxy 1 GW 1 PBX C
SIP/2.0 100 Trying Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1;received=192.0.2.111 From: Alice <sips:[email protected];user=phone>;tag=9fxced76sl To: Carol <sips:[email protected];user=phone> Call-ID: [email protected] CSeq: 2 INVITE Content-Length: 0

EventHelix.com/EventStudio 2.5 10-Jun-05 22:14 (Page 4)

Q.931 SETUP

The GW routes the call. Since Carol is served by an ISDN PBX, the Gateway initiates a Q.931 call setup with the PBX.

Protocol discriminator=Q.931 Message type=SETUP Bearer capability: Information transfer capability=0 (Speech) or 16 (3.1 kHz audio) Channel identification=Preferred or exclusive B-channel Progress indicator=1 (Call is not end-to-end ISDN;further call progress information may be available inband) Called party number: Type of number unknown Digits=444-3333

Q.931 CALL PROCEEDING

The ISDN PBX responds with Call Proceeding. This message indicates that the call is in the process of being setup.

Protocol discriminator=Q.931 Message type=CALL PROC Channel identification=Exclusive B-channel

Q.931 PROGRESS

The ISDN PBX passes call progress information to the Gateway. This message indicates that the called subscriber is being rung.

Protocol discriminator=Q.931 Message type=PROG Progress indicator=1 (Call is not end-to-end ISDN;further call progress information may be available inband)

SIP 180 Ringing

The Gateway sends the Ringing indication back to the proxy.

SIP/2.0 180 Ringing Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1;received=192.0.2.111 Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9;received=192.0.2.101 Record-Route: <sips:ss1.a.example.com;lr> From: Alice <sips:[email protected];user=phone>;tag=9fxced76sl To: Carol <sips:[email protected];user=phone>;tag=314159 Call-ID: [email protected] CSeq: 2 INVITE Contact: <sips:[email protected]> Content-Length: 0

SIP 180 Ringing

The proxy forwards the ringing indication to Alice's PC.

SIP/2.0 180 Ringing Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9;received=192.0.2.101 Record-Route: <sips:ss1.a.example.com;lr> From: Alice <sips:[email protected];user=phone>;tag=9fxced76sl To: Carol <sips:[email protected];user=phone>;tag=314159 Call-ID: [email protected] CSeq: 2 INVITE Contact: <sips:[email protected]> Content-Length: 0

Q.931 CONNECT
Protocol discriminator=Q.931 Message type=CONN

Carol has answered the call. This results in Q.931 CONNECT message being sent to the Gateway.

Q.931 CONNECT ACK

Protocol discriminator=Q.931 Message type=CONN ACK

The Gateway replies with Connect Ack.

allocate Port 3456

SIP 200 OK
Media information

The Gateway allocates a port for receiving RTP data from Alice's PC. The port information will be passed to originating subscriber via the "SIP 200 OK" response. The Gateway indicates to the Proxy that the call is successful. The RTP audio receive port information is also passed in this message.

Session Initiation Protocol (SIP Tutorial: SIP to ISDN Q.931 Call Flow (Detailed)) SIP Subscriber Network SIP Client VOIP Network Company Network Alice Proxy 1 GW 1 PBX C
SIP/2.0 200 OK Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1;received=192.0.2.111 Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9;received=192.0.2.101 Record-Route: <sips:ss1.a.example.com;lr> From: Alice <sips:[email protected];user=phone>;tag=9fxced76sl To: Carol <sips:[email protected];user=phone>;tag=314159 Call-ID: [email protected] CSeq: 2 INVITE Contact: <sips:[email protected]> Content-Type: application/sdp Content-Length: 144 v=0 o=GW 2890844527 2890844527 IN IP4 gw1.a.example.com s=c=IN IP4 gw1.a.example.com t=0 0 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000

EventHelix.com/EventStudio 2.5 10-Jun-05 22:14 (Page 5)

SIP 200 OK

The Proxy forwards the message to Alice's PC.

SIP/2.0 200 OK Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9;received=192.0.2.101 Record-Route: <sips:ss1.a.example.com;lr> From: Alice <sips:[email protected];user=phone>;tag=9fxced76sl To: Carol <sips:[email protected];user=phone>;tag=314159 Call-ID: [email protected] CSeq: 2 INVITE Contact: <sips:[email protected]> Content-Type: application/sdp Content-Length: 144 v=0 o=GW 2890844527 2890844527 IN IP4 gw1.a.example.com s=c=IN IP4 gw1.a.example.com t=0 0 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000

SIP ACK
ACK sips:[email protected] SIP/2.0 Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9 Max-Forwards: 70 Route: <sips:ss1.a.example.com;lr> From: Alice <sips:[email protected];user=phone>;tag=9fxced76sl To: Carol <sips:[email protected];user=phone>;tag=314159 Call-ID: [email protected] CSeq: 2 ACK Content-Length: 0

Alice's PC acknowledges the message.

SIP ACK

The Proxy forwards the ack to the Gateway.

ACK sips:[email protected] SIP/2.0 Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1 Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9;received=192.0.2.101 Max-Forwards: 69 From: Alice <sips:[email protected];user=phone>;tag=9fxced76sl To: Carol <sips:[email protected];user=phone>;tag=314159 Call-ID: [email protected] CSeq: 2 ACK Content-Length: 0 Two way voice is active at this time. Alice and Carol are talking. Alice Hangs Up with Carol.

SIP BYE
BYE sips:[email protected] SIP/2.0 Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9 Max-Forwards: 70 Route: <sips:ss1.a.example.com;lr> From: Alice <sips:[email protected];user=phone>;tag=9fxced76sl To: Carol <sips:[email protected];user=phone>;tag=314159 Call-ID: [email protected] CSeq: 3 BYE Content-Length: 0

SIP BYE signals the release of the call.

SIP BYE

The Bye is forwarded to the Gateway.

BYE sips:[email protected] SIP/2.0 Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1 Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9;received=192.0.2.101 Max-Forwards: 69 From: Alice <sips:[email protected];user=phone>;tag=9fxced76sl To: Carol <sips:[email protected];user=phone>;tag=314159 Call-ID: [email protected] CSeq: 3 BYE

Session Initiation Protocol (SIP Tutorial: SIP to ISDN Q.931 Call Flow (Detailed)) SIP Subscriber Network SIP Client VOIP Network Company Network Alice Proxy 1 GW 1 PBX C
Content-Length: 0

EventHelix.com/EventStudio 2.5 10-Jun-05 22:14 (Page 6)

Q.931 DISCONNECT
Protocol discriminator=Q.931 Message type=DISC Cause=16 (Normal clearing)

The Gateway initiates the call release on SS7 side.

SIP 200 OK

The Gateway acknowledges the BYE to the Proxy with an 200 OK respponse code.

SIP/2.0 200 OK Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1;received=192.0.2.111 Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9;received=192.0.2.101 From: Alice <sips:[email protected];user=phone>;tag=9fxced76sl To: Carol <sips:[email protected];user=phone>;tag=314159 Call-ID: [email protected] CSeq: 3 BYE Content-Length: 0

SIP 200 OK

The Proxy forwards the ack to Alice's PC.

SIP/2.0 200 OK Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9;received=192.0.2.101 From: Alice <sips:[email protected];user=phone>;tag=9fxced76sl To: Carol <sips:[email protected];user=phone>;tag=314159 Call-ID: [email protected] CSeq: 3 BYE Content-Length: 0

free Port 49172

Q.931 RELEASE
Protocol discriminator=Q.931 Message type=REL

The ISDN PBX indicates to the Gateway that it is releasing the call.

free Port 3456

Q.931 RELEASE COMPLETE

Protocol discriminator=Q.931 Message type=REL COM Full Copyright Statement Copyright The Internet Society (2003). All Rights Reserved. This document and translations of it may be copied and furnished to others, and derivative works that comment on or otherwise explain it or assist in its implementation may be prepared, copied, published and distributed, in whole or in part, without restriction of any kind, provided that the above copyright notice and this paragraph are included on all such copies and derivative works. However, this document itself may not be modified in any way, such as by removing the copyright notice or references to the Internet Society or other Internet organizations, except as needed for the purpose of developing Internet standards in which case the procedures for copyrights defined in the Internet Standards process must be followed, or as required to translate it into languages other than English.

The Gateway acknowledges the call release of the call with the Release Complete message.