Sip协议(三)- 通话接听流程

本文主要是介绍Sip协议(三)- 通话接听流程，希望对大家解决编程问题提供一定的参考价值，需要的开发者们随着小编来一起学习吧！

Sip协议(三)- 通话接听流程

本文主要记录下sip通信下接听的流程.

一: 接听流程

1. agent接听电话
2. 远端在未接听情况下主动挂断电话.

接听流程涉及到的请求有: INVITE,CANCEL,ACK,BYE

具体的过程如下:

二: sip过程

1. agent收到INVITE

   INVITE sip:1000@10.0.0.1:10000;transport=tcp SIP/2.0
   Via: SIP/2.0/TCP 10.0.0.1:5060;branch=z9hG4bKxxxx;rport=5060
   Route: <sip:1000@124.64.0.1:10988>;transport=tcp
   Max-Forwards: 70
   From: "Extension 1001" <sip:1001@172.17.0.1>;tag=Q79QQcg41UBpe
   To: <sip:1000@10.0.0.1:10000;transport=tcp>
   Call-ID: e040d830-a8aa-123d-0a8e-11111111111b
   CSeq: 84836083 INVITE
   Contact: <sip:mod_sofia@10.0.0.1:5060;transport=tcp>
   User-Agent: FreeSWITCH-mod_sofia/1.10.10-release~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: path, replaces
   Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 426
   X-voice_call_number: 18300000000
   X-empolyee_id: 1001
   X-FS-Support: update_display,send_info
   Remote-Party-ID: "Extension 1001" <sip:1001@172.17.0.1>;party=calling;screen=yes;privacy=offv=0
   o=FreeSWITCH 1718762975 1718762976 IN IP4 10.0.0.1
   s=FreeSWITCH
   c=IN IP4 10.0.0.1
   t=0 0
   m=audio 16776 RTP/AVP 9 0 8 101
   a=rtpmap:9 G722/8000
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   a=ptime:20
   m=video 16862 RTP/AVP 102
   b=AS:3072
   a=rtpmap:102 VP8/90000
   a=rtcp-fb:102 ccm fir
   a=rtcp-fb:102 ccm tmmbr
   a=rtcp-fb:102 nack
   a=rtcp-fb:102 nack pli
   

2. agent发起 Trying 100

   SIP/2.0 100 Trying
   Via: SIP/2.0/TCP 10.0.0.1:5060;branch=z9hG4bK1111;rport=5060
   To: <sip:1000@10.0.0.1:10000;transport=tcp>
   From: "Extension 1001" <sip:1001@172.17.0.1>;tag=027KZHpcBFDXH
   Call-ID: 42ed1f48-a8ac-123d-0a8e-11111111111a
   CSeq: 84836381 INVITE
   Server: AgentTest/1.0.0/V2318A
   Content-Length: 0
   

3. agent发起请求 Ringing 180

   180: 振铃中.

    SIP/2.0 180 RingingVia: SIP/2.0/TCP 10.0.0.1:5060;branch=z9hG4bKgvtvc29HgQZ5e;rport=5060To: <sip:1000@10.0.0.1:10000;transport=tcp>;tag=ec6c9cb5b93df28eFrom: "Extension 1001" <sip:1001@172.17.0.1>;tag=027KZHpcBFDXHCall-ID: 42ed1f48-a8ac-123d-0a8e-11111111111aCSeq: 84836381 INVITEServer: AgentTest/1.0.0/V2318AContent-Length: 204Content-Type: application/sdpv=0o=1000@10.0.0.1 0 0 IN IP4 10.239.19.29s=Session SIP/SDPc=IN IP4 10.239.19.29t=0 0m=audio 21000 RTP/AVP 8 101a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15
   

4. agent接收到CANCEL请求

   CANCEL sip:1000@10.0.0.1:10000;transport=tcp SIP/2.0
   Via: SIP/2.0/TCP 10.0.0.1:5060;branch=z9hG4bKgvtvc29HgQZ5e;rport=5060
   Route: <sip:1000@124.64.22.13:10988>;transport=tcp
   Max-Forwards: 70
   From: "Extension 1001" <sip:1001@172.17.0.1>;tag=027KZHpcBFDXH
   To: <sip:1000@10.0.0.1:10000;transport=tcp>
   Call-ID: 42ed1f48-a8ac-123d-0a8e-11111111111a
   CSeq: 84836381 CANCEL
   Reason: SIP;cause=487;text="ORIGINATOR_CANCEL"
   Content-Length: 0
   

5. agent回执 200

   SIP/2.0 200 OK
   Via: SIP/2.0/TCP 10.0.0.1:5060;branch=z9hG4bKgvtvc29HgQZ5e;rport=5060
   To: <sip:1000@10.0.0.1:10000;transport=tcp>
   From: "Extension 1001" <sip:1001@172.17.0.1>;tag=027KZHpcBFDXH
   Call-ID: 42ed1f48-a8ac-123d-0a8e-11111111111a
   CSeq: 84836381 CANCEL
   Server: AgentTest/1.0.0/V2318A
   Content-Length: 0
   

6. agent回执487

   487: Request Terminated 请求被BYE或者CANCEL所终止.

    SIP/2.0 487 Request TerminatedVia: SIP/2.0/TCP 10.0.0.1:5060;branch=z9hG4bKgvtvc29HgQZ5e;rport=5060To: <sip:1000@10.0.0.1:10000;transport=tcp>From: "Extension 1001" <sip:1001@172.17.0.1>;tag=027KZHpcBFDXHCall-ID: 42ed1f48-a8ac-123d-0a8e-11111111111aCSeq: 84836381 INVITEServer: AgentTest/1.0.0/V2318AContent-Length: 0
   

这篇关于Sip协议(三)- 通话接听流程的文章就介绍到这儿，希望我们推荐的文章对编程师们有所帮助！

---