To give a precise description - here is the call flow:
SIP OUT: INVITE sip:[email protected]:20972 SIP/2.0 ... Content-Type: application/sdp Content-Length: 345 v=0 o=TLS_SIP 12858 12858 IN IP4 172.16.1.64 s=SIP_TLS c=IN IP4 172.16.1.64 t=0 0 m=audio 30000 RTP/SAVP 8 0 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:cmxxxxxxxxxxxxxx a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=encryption:optional a=sendrecv SIP IN: SIP/2.0 100 Trying ... Content-Length: 0 SIP IN: SIP/2.0 183 Session Progress ... Content-Type: application/sdp Content-Length: 297 v=0 o=SIP_TLS 6314 6314 IN IP4 172.16.1.63 s=TLS_SIP c=IN IP4 172.16.1.63 t=0 0 m=audio 40006 RTP/SAVP 8 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:Qixxxxxxxxxxxxxxx a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <--------- t = 0: RTP IN ------------------ -------------------- sRTP OUT ------------> SIP IN: SIP/2.0 200 OK ... Content-Type: application/sdp Content-Length: 297 v=0 o=SIP_TLS 6314 6314 IN IP4 172.16.1.63 s=TLS_SIP c=IN IP4 172.16.1.63 t=0 0 m=audio 40006 RTP/SAVP 8 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:Qixxxxxxxxxxxxxxx a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv SIP OUT: ACK sip:[email protected]:20972;transport=tls SIP/2.0 ... Content-Length: 0 <----- t = 1.5sec: switch to sRTP IN ------ SIP IN: BYE sip:[email protected]:5061;transport=tls SIP/2.0 ... _______________________________________________ Sip-implementors mailing list [email protected] https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
