Greetings. I test the own implementation of softphone. It is not based on pjsip; instead it uses combination of resiprocate/jrtplib/own ice & media stacks. I try to make ice traversal working for rtcp component too (rtp component works good). The test peer is pjsua from 2.1.0 on Ubuntu 12.04 in virtualbox. The sent offer is: <---INVITE sip:dbogovych1 at voipobjects.com SIP/2.0 <---Via: SIP/2.0/TCP 192.168.1.102:5060 ;branch=z9hG4bK-524287-1---a27da3758f4a7f37;rport <---Max-Forwards: 70 <---Contact: <sip:dbogovych at 95.132.162.61 :5060;transport=tcp>;+sip.instance="8078730" <---To: <sip:dbogovych1 at voipobjects.com> <---From: <sip:dbogovych@xxxxxxxxxxxxxxxxxxxxx>;tag=ee352f5f <---Call-ID: KcLpGZhqZzfz033okcroYg.. <---CSeq: 1 INVITE <---Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, INFO, MESSAGE, REFER, NOTIFY, SUBSCRIBE, REGISTER <---Content-Type: application/sdp <---Supported: timer, norefersub, replaces, eventlist <---User-Agent: IntTalk_2.2.10 <---Content-Length: 991 <--- <---v=0 <---o=ITS_user 0 1 IN IP4 8046 <---s=ITS_session <---c=IN IP4 95.132.162.61 <---t=0 0 <---a=ice-pwd:sxknaaiewaqajgizkrkfaz <---a=ice-ufrag:itab <---m=audio 8046 RTP/AVP 106 0 8 3 100 99 9 97 103 104 101 <---a=rtpmap:106 opus/16000 <---a=rtpmap:0 pcmu/8000 <---a=rtpmap:8 pcma/8000 <---a=rtpmap:3 gsm/8000 <---a=rtpmap:100 ilbc/8000 <---a=rtpmap:99 ilbc/8000 <---a=fmtp:99 mode=20 <---a=rtpmap:9 g722/16000 <---a=rtpmap:97 isac/16000 <---a=rtpmap:103 speex/8000 <---a=rtpmap:104 speex/16000 <---a=rtpmap:101 telephone-event/8000 <---a=silenceSupp:off - - - - <---a=RS:0 <---a=RR:0 <---a=candidate:20490432 1 UDP 2113929471 192.168.56.1 8046 typ host <---a=candidate:1711384768 1 UDP 2113929471 192.168.1.102 8046 typ host <---a=candidate:1728161984 1 UDP 1677721855 95.132.162.61 8046 typ srflx raddr 192.168.1.102 rport 8046 <---a=candidate:20490432 2 UDP 2113929470 192.168.56.1 8047 typ host <---a=candidate:1711384768 2 UDP 2113929470 192.168.1.102 8047 typ host <---a=candidate:1728161984 2 UDP 1677721854 95.132.162.61 8047 typ srflx raddr 192.168.1.102 rport 8047 The answer is: --->SIP/2.0 200 OK --->Via: SIP/2.0/TCP 192.168.1.102:5060 ;rport=2712;received=95.132.162.61;branch=z9hG4bK-524287-1---a27da3758f4a7f37 --->Record-Route: <sip:0.0.0.0;lr;r2=on> --->Record-Route: <sip:0.0.0.0;transport=tcp;lr;r2=on> --->Contact: <sip:dbogovych1 at 95.132.162.61:59679;ob>;+sip.ice --->To: <sip:dbogovych1 at voipobjects.com >;tag=JVWGJPZQrdfCP4-Xf0.t41FCQP1lbC1X --->From: <sip:dbogovych@xxxxxxxxxxxxxxxxxxxxx>;tag=ee352f5f --->Call-ID: KcLpGZhqZzfz033okcroYg.. --->CSeq: 1 INVITE --->Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS --->Content-Type: application/sdp --->Supported: replaces, 100rel, timer, norefersub --->Content-Length: 461 ---> --->v=0 --->o=- 3595921123 3595921124 IN IP4 95.132.162.61 --->s=pjmedia --->b=AS:84 --->t=0 0 --->a=X-nat:8 --->m=audio 50889 RTP/AVP 0 101 --->c=IN IP4 95.132.162.61 --->b=TIAS:64000 --->b=RS:0 --->b=RR:0 --->a=sendrecv --->a=rtpmap:0 PCMU/8000 --->a=ice-ufrag:761e30d2 --->a=ice-pwd:350aa3bd --->a=candidate:Sa00020f 1 UDP 1862270975 95.132.162.61 50889 typ srflx raddr 10.0.2.15 rport 4037 --->a=candidate:Ha00020f 1 UDP 1694498815 10.0.2.15 4037 typ host --->a=rtpmap:101 telephone-event/8000 --->a=fmtp:101 0-15 Why pjsua does not insert RTCP component information into candidate list? Does it support RTCP in ICE? Thank you! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20131213/6b005d61/attachment-0001.html>
