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).
This article is based on thecall 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 theProxy 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 beingrung. 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 "SIP200 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.
Calling #, Called #, Contact, Media Information
SIP 100 Trying
Identify the Gateway thatservers Carol
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
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 CallFlow (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 tothe Proxy with an 200 OK respponse code. The Proxy forwards the ack to Alice's PC.
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...