You mean both you and your carrier are using their own rtp-proxy? If so, then simply add "f" flag to rtpproxy_offer | rtpproxy_answer. Which will allow you can you carrier to create a chain of rtp-proxy together. See flags description here,
http://www.opensips.org/html/docs/modules/devel/rtpproxy.html#id292744 Thank you. On Fri, Feb 15, 2013 at 2:18 AM, Seth Schultz <[email protected]>wrote: > Hello, > > I am having a problem with RTPProxy where in the reply, the remote carrier > is sending the "nortpproxy_str" in the reply SDP (example below). I would > like to know what the best way is to detect this, and remove it from the > sip message before calling rtpproxy_answer function, because > rtpproxy_answer will fail if the nortpproxy_str already exists in the SDP. > > Thanks in advance, > Seth > > U 2013/02/14 19:32:21.142567 yyy.yyy.yyy.yyy:5060 -> xxx.xxx.xxx.xxx:5060 > INVITE sip:[email protected].**xxx SIP/2.0 > Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=**z9hG4bK2d9e.187ebf5.0 > Max-Forwards: 69 > From: "Unknown" <sip:[email protected].**yyy>;tag=33XjNy6SQZrQS > To: <sip:[email protected].**yyy> > Call-ID: 004c5840-f1aa-1230-9c93-**6320dec8e883 > CSeq: 40108106 INVITE > Contact: <sip:yyy.yyy.yyy.yyy;did=3901.**59b3bb21> > User-Agent: FS1 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, > REFER, NOTIFY > Supported: timer, precondition, path, replaces > Allow-Events: talk, hold, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 247 > P-Call-Type: Notification > X-FS-Support: update_display,send_info > Remote-Party-ID: "Unknown" <sip:[email protected].** > yyy>;party=calling;screen=yes;**privacy=off > > v=0 > o=FreeSWITCH 1360855702 1360855703 IN IP4 yyy.yyy.yyy.yyy > s=FreeSWITCH > c=IN IP4 yyy.yyy.yyy.yyy > t=0 0 > m=audio 40562 RTP/AVP 0 8 3 101 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=schipmangled:yes <--- rtpproxy added this on initial invite > > ... > > U 2013/02/14 19:32:37.425606 xxx.xxx.xxx.xxx:5060 -> yyy.yyy.yyy.yyy:5060 > SIP/2.0 200 OK > Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=**z9hG4bK2d9e.187ebf5.0 > From: "Unknown" <sip:[email protected].**yyy>;tag=33XjNy6SQZrQS > To: <sip:[email protected].**yyy>;tag=SDs07f299-gK0e9f2e8d > Call-ID: 004c5840-f1aa-1230-9c93-**6320dec8e883 > CSeq: 40108106 INVITE > Accept: application/sdp, application/isup, application/dtmf, > application/dtmf-relay, multipart/mixed > Contact: <sip:xxx.xxx.xxx.xxx;did=39.**60d51ef> > Allow: INVITE,ACK,CANCEL,BYE,**REGISTER,REFER,INFO,SUBSCRIBE,** > NOTIFY,PRACK,UPDATE,OPTIONS > Require: timer > Supported: timer > Session-Expires: 7200;refresher=uas > Content-Length: 259 > Content-Disposition: session; handling=required > Content-Type: application/sdp > > v=0 > o=Sonus_UAC 7607 20874 IN IP4 xxx.xxx.xxx.xxx > s=SIP Media Capabilities > c=IN IP4 xxx.xxx.xxx.xxx > t=0 0 > m=audio 29772 RTP/AVP 0 101 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=schipmangled:yes <--- they sent this back in the 200 OK reply > a=ptime:20 > a=sendrecv > > > ______________________________**_________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-**bin/mailman/listinfo/users<http://lists.opensips.org/cgi-bin/mailman/listinfo/users> > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: [email protected] Email: [email protected]
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