|
Hi, explain SIP - ISUP call flow? michaeldavid23 11-July-2008 11:11:50 AMComments SIP UAC (1)-> MGC/MG () SSP SIP UAC (2)<- MGC/MG ->(3) SSP SIP UAC (5)<-MGC/MG <-(4) SSP SIP UAC () <-MGC/MG <-(6) SSP SIP UAC (8)<-MGC/MG <-(7) SSP SIP UAC (9)-> MGC/MG () SSP SIP UAC (10)-> MGC/MG () SSP SIP UAC (11)<-> MGC/MG <->(11) SSP SIP UAC (12)-> MGC/MG (13)-> SSP SIP UAC (14)<- MGC/MG (15) <- SSP 1 Invite include URI of called UAS (PSTN phone) and SDP 2 and 3. If the proxy (here it is in the MGC) can authenticate the UAS, it sends 100 Trying and passes IAM on to the PSTN switch 4 PSTN Switch returns ACM that it has accepted call and will ring the called UAC 5 The ACM is interpreted into 183 Session in Progress and UAS SDP information is added 6 In this case Ring Tone and Call Progress tones are delivered via a temporary one-way connection. The MG must translate from the SSP codec to the UAC codec. Note that the UAC could accept a 180 Ringback response. Some MGCs might do this instead of establishing a session 7 When the called party goes off hook, the SSP generates an ANM which is interpreted into 200 OK. 8, 9, 10, and 11 When UAC gets OK, it ACKs the one-way invite and issues another Invite for a two-way conversation. (There is an OK and Ack involved, not shown to fit the page.) 12. Whoever hangs up first will send a BYE and 13 REL to SSP. The SS7 network will respond with REL and RLC (15) and MSGC will send 200OK to UAC. Posted by yogendra The Session Initiation Protocol (SIP) is an Internet Engineering Task Force (IETF) standard call control protocol, based on research at Columbia University by Henning Schulzrinne and his team. SIP is designed to provide signaling and session management for voice and multimedia connections over packet-based networks. It is a peer-to-peer protocol with intelligent endpoints and distributed call control, such as H.323. Gateways that use SIP do not depend on a call agent, although the protocol does define several functional entities that help SIP endpoints locate each other and establish a session. ISUP Call Flow: ISUP call flows illustrate the order of messages in typical success and error cases when setting up a call initiated from the SIP network. "100 Trying" acknowledgments to INVITE requests are not displayed, since their presence is optional. In ISUP call flow, all call signaling (SIP, ISUP) is going to and from the MGC; media handling (e.g. audio cut-through, trunk freeing) is being performed by the MG, under the control of the MGC. For the purpose of simplicity, these are shown as a single node, labeled "MGC/MG." Posted by sagitraz |
Posted: 12-July-2008 03:50:28 AM By: sagitraz The Session Initiation Protocol (SIP) is an Internet Engineering Task Force (IETF) standard call control protocol, based on research at Columbia University by Henning Schulzrinne and his team. SIP is designed to provide signaling and session management for voice and multimedia connections over packet-based networks. It is a peer-to-peer protocol with intelligent endpoints and distributed call control, such as H.323. Gateways that use SIP do not depend on a call agent, although the protocol does define several functional entities that help SIP endpoints locate each other and establish a session. ISUP Call Flow: ISUP call flows illustrate the order of messages in typical success and error cases when setting up a call initiated from the SIP network. "100 Trying" acknowledgments to INVITE requests are not displayed, since their presence is optional. In ISUP call flow, all call signaling (SIP, ISUP) is going to and from the MGC; media handling (e.g. audio cut-through, trunk freeing) is being performed by the MG, under the control of the MGC. For the purpose of simplicity, these are shown as a single node, labeled "MGC/MG." | |
Posted: 14-July-2008 12:00:52 PM By: yogendra SIP UAC (1)-> MGC/MG () SSP SIP UAC (2)<- MGC/MG ->(3) SSP SIP UAC (5)<-MGC/MG <-(4) SSP SIP UAC () <-MGC/MG <-(6) SSP SIP UAC (8)<-MGC/MG <-(7) SSP SIP UAC (9)-> MGC/MG () SSP SIP UAC (10)-> MGC/MG () SSP SIP UAC (11)<-> MGC/MG <->(11) SSP SIP UAC (12)-> MGC/MG (13)-> SSP SIP UAC (14)<- MGC/MG (15) <- SSP 1 Invite include URI of called UAS (PSTN phone) and SDP 2 and 3. If the proxy (here it is in the MGC) can authenticate the UAS, it sends 100 Trying and passes IAM on to the PSTN switch 4 PSTN Switch returns ACM that it has accepted call and will ring the called UAC 5 The ACM is interpreted into 183 Session in Progress and UAS SDP information is added 6 In this case Ring Tone and Call Progress tones are delivered via a temporary one-way connection. The MG must translate from the SSP codec to the UAC codec. Note that the UAC could accept a 180 Ringback response. Some MGCs might do this instead of establishing a session 7 When the called party goes off hook, the SSP generates an ANM which is interpreted into 200 OK. 8, 9, 10, and 11 When UAC gets OK, it ACKs the one-way invite and issues another Invite for a two-way conversation. (There is an OK and Ack involved, not shown to fit the page.) 12. Whoever hangs up first will send a BYE and 13 REL to SSP. The SS7 network will respond with REL and RLC (15) and MSGC will send 200OK to UAC. |