Bugs item #3584714, was opened at 2012-11-06 03:13 Message generated for change (Comment added) made by bogdan_iancu You can respond by visiting: https://sourceforge.net/tracker/?func=detail&atid=1086410&aid=3584714&group_id=232389
Please note that this message will contain a full copy of the comment thread, including the initial issue submission, for this request, not just the latest update. Category: core Group: 1.8.x >Status: Closed >Resolution: Invalid Priority: 5 Private: No Submitted By: apsaras (apsaras) Assigned to: Nobody/Anonymous (nobody) Summary: RTP Traffic Sent to wrong destination Initial Comment: During the negotiation with a carrier SBC where SIP and RTP are running on different IPs, I found out that in some cases - unknown when - the RTP traffic is sent to SIP IP and not to RTP IP. We run 4 RTP Proxies one for each core in order to maximize performance but even if we use just one of them, the problem remains. That is the initial INVITE sent from our Proxy to the Carrier INVITE sip:[email protected] SIP/2.0 Record-Route: <sip:91.XXX.XXX.90;lr;nat=yes;did=716.b4738932> Via: SIP/2.0/UDP 91.XXX.XXX.90;branch=z9hG4bK7e46.ec664a53.0 Max-Forwards: 69 From: "asterisk" <sip:[email protected]>;tag=as0b2df024 To: <sip:[email protected]> Contact: <sip:91.XXX.XXX.90;did=716.b4738932> Call-ID: [email protected] CSeq: 102 INVITE User-Agent: OpenSIPs Privacy: id;header;user P-Asserted-Identity: <sip:[email protected]> Date: Tue, 06 Nov 2012 10:52:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 287 v=0 o=root 2009131870 2009131870 IN IP4 91.XXX.XXX.90 s=OpenSIPs c=IN IP4 91.XXX.XXX.90 t=0 0 m=audio 14034 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=direction:active a=sdpmangled:yes and this is the OK sent from Carrier SIP/2.0 200 OK Session-Expires: 3600;refresher=uas Require: timer Via: SIP/2.0/UDP 91.XXX.XXX.90;branch=z9hG4bK7e46.ec664a53.0 Record-Route: <sip:91.XXX.XXX.90;lr;nat=yes;did=716.b4738932> To: <sip:[email protected]>;tag=3561187924-714596 From: "asterisk" <sip:[email protected]>;tag=as0b2df024 Call-ID: [email protected] CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH Contact: <sip:[email protected]:5060> Call-Info: <sip:194.XXX.XXX.66>;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 333 v=0 o=root 188 1 IN IP4 194.XXX.XXX.66 s=sip call c=IN IP4 194.XXX.XXX.76 t=0 0 m=audio 44924 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=X-vrzcap:vbd Ver=0 Mode=FaxPr ModemRtpRed=0 a=X-vrzcap:identification bin=DSR1e814 Prot=mgcp App=MG But RTP traffic sent to .66 instead of .76 ---------------------------------------------------------------------- >Comment By: Bogdan-Andrei Iancu (bogdan_iancu) Date: 2012-11-06 08:09 Message: Ah..I see your update on flags :) Closing this ticket. ---------------------------------------------------------------------- Comment By: Bogdan-Andrei Iancu (bogdan_iancu) Date: 2012-11-06 08:08 Message: Hi, Hard to tell without knowing: what flags do you pass to rtpproxy_xxxx() a full capture of RTP received on that call - rtpproxy learns destination based on the incoming traffic. Regards, Bogdan ---------------------------------------------------------------------- Comment By: apsaras (apsaras) Date: 2012-11-06 08:02 Message: Problem solved. Need to put r flag to engage_rtp_proxy command. ---------------------------------------------------------------------- You can respond by visiting: https://sourceforge.net/tracker/?func=detail&atid=1086410&aid=3584714&group_id=232389 _______________________________________________ Devel mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
