Alex,
A) without proxy : Source (sip a-side)---------> Destination (sip b-side) 192.168.1.51:29100 -------- > 194.140.246.34:13524 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ B) with proxy ? Regards! Vp -----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of ext Alex Bakker Sent: Thursday, May 27, 2010 7:08 AM To: [email protected] Subject: [Sip-implementors] RTP streaming to other port than speficied inSDP data (but client picks up audio) 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
