Hi, We have been trying to do an interop between WebRTC clients provided by Google Chrome and our SIP client built over PJSIP stack. It has been found that the WebRTC provided by Google Chrome mandates the following technologies in the SDP: 1. Secured RTP - Already supported by PJSIP 2. ICE - Can be used using PJNATH library 3. Secured RTCP with feedback i.e. RTP/SAVPF format in the SDP [RFC 5124 http://tools.ietf.org/html/rfc5124] - Currently not supported by PJSIP We tried a SIP call between our WebRTC client and the iOS client build over PJSIP. PJSIP stack responded with 488 Not Acceptable for the INVITE. Attached are the SIP packets for your reference. Also find attached a sample outgoing INVITE generated by our WebRTC client application. Please let know whether there is any plan to add the support for RFC 5124 i.e. SRTCP with feedback. Thanks, Yogesh -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20130403/0f17cbcd/attachment-0001.html> -------------- next part -------------- INVITE sip:agnity07 at genfuzion.com SIP/2.0 Via: SIP/2.0/UDP 10.32.4.83:5060;branch=z9hG4bKeYkUGqUfQeCExTDt5kbwYQ~~43 Max-Forwards: 70 Route: <sip:agnity07 at 64.129.37.90:5060;lr> To: <sip:agnity07 at genfuzion.com> From: <sip:agnity08@xxxxxxxxxxxxx>;tag=ds98c541ea Call-ID: ASE_1357653026369_30_null_192.168.0.1 CSeq: 1 INVITE Content-Length: 2287 Contact: <sip:agnity08 at 10.32.4.83:5060;transport=udp> Content-Type: application/sdp v=0 o=- 737416374 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE audio video m=audio 29452 RTP/SAVP 103 104 0 8 106 105 13 126 c=IN IP4 125.23.217.66 a=rtcp:29452 IN IP4 125.23.217.66 a=candidate:220237935 1 udp 2113937151 10.32.4.77 4487 typ host generation 0 a=candidate:220237935 2 udp 2113937151 10.32.4.77 4487 typ host generation 0 a=candidate:3993048707 1 udp 1677729535 125.23.217.66 29452 typ srflx generation 0 a=candidate:3993048707 2 udp 1677729535 125.23.217.66 29452 typ srflx generation 0 a=candidate:1134783647 1 tcp 1509957375 10.32.4.77 4489 typ host generation 0 a=candidate:1134783647 2 tcp 1509957375 10.32.4.77 4489 typ host generation 0 a=ice-ufrag:QDd4MKl1Md7l6JBr a=ice-pwd:YLxZ1GkaLaPW6IqGnkq/Wtnu a=ice-options:google-ice a=sendrecv a=mid:audio a=rtcp-mux a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:EuRDKWWS4FjExJamHGT9D3UEN1ZB5o3CKmfVGeQj a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=ssrc:1609497791 cname:kojJRdf++ESvQF4h a=ssrc:1609497791 mslabel:HxnllOWwENgj6kpgpjJidRf3OOFzqx0YZkcb a=ssrc:1609497791 label:HxnllOWwENgj6kpgpjJidRf3OOFzqx0YZkcb00 m=video 29452 RTP/SAVP 100 101 102 c=IN IP4 125.23.217.66 a=rtcp:29452 IN IP4 125.23.217.66 a=candidate:220237935 1 udp 2113937151 10.32.4.77 4487 typ host generation 0 a=candidate:220237935 2 udp 2113937151 10.32.4.77 4487 typ host generation 0 a=candidate:3993048707 1 udp 1677729535 125.23.217.66 29452 typ srflx generation 0 a=candidate:3993048707 2 udp 1677729535 125.23.217.66 29452 typ srflx generation 0 a=candidate:1134783647 1 tcp 1509957375 10.32.4.77 4489 typ host generation 0 a=candidate:1134783647 2 tcp 1509957375 10.32.4.77 4489 typ host generation 0 a=ice-ufrag:QDd4MKl1Md7l6JBr a=ice-pwd:YLxZ1GkaLaPW6IqGnkq/Wtnu a=ice-options:google-ice a=sendrecv a=mid:video a=rtcp-mux a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:EuRDKWWS4FjExJamHGT9D3UEN1ZB5o3CKmfVGeQj a=rtpmap:100 VP8/90000 a=rtpmap:101 red/90000 a=rtpmap:102 ulpfec/90000 a=ssrc:3814166398 cname:kojJRdf++ESvQF4h a=ssrc:3814166398 mslabel:HxnllOWwENgj6kpgpjJidRf3OOFzqx0YZkcb a=ssrc:3814166398 label:HxnllOWwENgj6kpgpjJidRf3OOFzqx0YZkcb10 -------------- next part -------------- A non-text attachment was scrubbed... Name: WebRTC_to_PJSIP_callFail.pcap Type: application/octet-stream Size: 3468 bytes Desc: not available URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20130403/0f17cbcd/attachment-0001.pcap>
