Maybe the other endpoint is doing asymmetric RTP http://www.ietf.org/rfc/rfc4961.txt
On Thu, May 27, 2010 at 4:37 PM, Alex Bakker <[email protected]> wrote: > Hello, > > I have made a dump (pcap) of an outbound sip call with a Siemens HG1500 and > although it seems to work fine, there is something in the signalling that I > don't quite understand. > > The SIP call setup itself seems pretty normal to me. The thing is that the > audio is NOT being streamed to the expected ports... > > [step 1] Siemens on LAN sends INVITE to proxy on internet (SDP: m=audio > 29100 RTP/AVP 8 0 18 4 98 99) > [step 2] Proxy on internet sends "100 Trying" > [step 3] Proxy on internet sends "183 Session Progress" (SDP: m=audio 13524 > RTP/AVP 8 98) > [step 4] Proxy on internet sends "200 OK" (SDP: m=audio 13524 RTP/AVP 8 98) > > >From this point on the 2 parties start to stream audio. > > The Siemens sends audio to the remote proxyto port 13524 (as expected) ! > The Siemens receives audio from remote proxy on port 2418 (and not > 29100!!!!) and this is very strange! > > I can understand that sometimes wrong IP-addresses are included in headers > due to NAT issues (local IP vs WAN IP). But this is new for me. Nevertheless > for some reason the Siemens picks up the audio from the remote proxy so it > seems to work! > > Does anyone have a clue why this can work? > > Best regards, > Alex > > > > > > > > > The complete INVITE header of the Siemens [step 1] looks like this: > > INVITE sip:[email protected]:5060 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.51:5060;branch=z9hG4bKa32fa2bc37531d1d0;rport > Proxy-Authorization: Digest > username="040200001",realm="gntel",nonce="36cf6af4",uri=" > sip:[email protected]:5060 > ",response="29628c3c18f1fff6f8021db247d06c3e",algorithm=MD5 > Max-Forwards: 70 > From: Anonymous > <sip:[email protected]<sip%[email protected]> >>;tag=7c486354b6 > To: <sip:[email protected] <sip%[email protected]>> > Call-ID: f1251cdd0675d86c > CSeq: 16249 INVITE > Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, INFO > Contact: <sip:[email protected]:5060> > Min-SE: 90 > Session-Expires: 90 > Supported: timer > User-Agent: HiPath 3000 V8 M5T SIP Stack/4.0.26.26 > Content-Type: application/sdp > Content-Length: 381 > > v=0 > o=MxSIP 0 393044530 IN IP4 192.168.1.51 > s=SIP Call > c=IN IP4 192.168.1.51 > t=0 0 > m=audio 29100 RTP/AVP 8 0 18 4 98 99 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=rtpmap:4 G723/8000 > a=rtpmap:98 telephone-event/8000 > a=rtpmap:99 red/8000 > a=silenceSupp:off - - - - > a=fmtp:18 annexb=no > a=fmtp:4 annexa=no > a=fmtp:98 0-15 > a=fmtp:99 98 > a=sendrecv > > > > > The complete "200 OK" from the SIP proxy on internet: > > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.1.51:5060 > ;branch=z9hG4bK6f6e89cababcefe3b;received=77.61.86.150;rport=5060 > From: Anonymous > <sip:[email protected]<sip%[email protected]> >>;tag=a5b5dfd4a6 > To: <sip:[email protected] <sip%[email protected]> >>;tag=as472bf900 > Call-ID: f556dc6cafb76f09 > CSeq: 28063 INVITE > User-Agent: gnTel VoIP Interconnect > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Contact: <sip:[email protected] <sip%[email protected]>> > Content-Type: application/sdp > Content-Length: 215 > > v=0 > o=root 6060 6061 IN IP4 194.140.246.34 > s=session > c=IN IP4 194.140.246.34 > t=0 0 > m=audio 13524 RTP/AVP 8 98 > a=rtpmap:8 PCMA/8000 > a=rtpmap:98 telephone-event/8000 > a=fmtp:98 0-16 > a=silenceSupp:off - - - - > _______________________________________________ > Sip-implementors mailing list > [email protected] > https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors > _______________________________________________ Sip-implementors mailing list [email protected] https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
