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,<br>
<br><a href="http://www.opensips.org/html/docs/modules/devel/rtpproxy.html#id292744">http://www.opensips.org/html/docs/modules/devel/rtpproxy.html#id292744</a><br><br>Thank you.<br><br><br><div class="gmail_quote">On Fri, Feb 15, 2013 at 2:18 AM, Seth Schultz <span dir="ltr"><<a href="mailto:sschultz@scholarchip.com" target="_blank">sschultz@scholarchip.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hello,<br>
<br>
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.<br>
<br>
Thanks in advance,<br>
Seth<br>
<br>
U 2013/02/14 19:32:21.142567 yyy.yyy.yyy.yyy:5060 -> xxx.xxx.xxx.xxx:5060<br>
INVITE sip:19999999999@xxx.xxx.xxx.<u></u>xxx SIP/2.0<br>
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=<u></u>z9hG4bK2d9e.187ebf5.0<br>
Max-Forwards: 69<br>
From: "Unknown" <sip:19999999999@yyy.yyy.yyy.<u></u>yyy>;tag=33XjNy6SQZrQS<br>
To: <sip:19999999999@yyy.yyy.yyy.<u></u>yyy><br>
Call-ID: 004c5840-f1aa-1230-9c93-<u></u>6320dec8e883<br>
CSeq: 40108106 INVITE<br>
Contact: <sip:yyy.yyy.yyy.yyy;did=3901.<u></u>59b3bb21><br>
User-Agent: FS1<br>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY<br>
Supported: timer, precondition, path, replaces<br>
Allow-Events: talk, hold, refer<br>
Content-Type: application/sdp<br>
Content-Disposition: session<br>
Content-Length: 247<br>
P-Call-Type: Notification<br>
X-FS-Support: update_display,send_info<br>
Remote-Party-ID: "Unknown" <sip:19999999999@yyy.yyy.yyy.<u></u>yyy>;party=calling;screen=yes;<u></u>privacy=off<br>
<br>
v=0<br>
o=FreeSWITCH 1360855702 1360855703 IN IP4 yyy.yyy.yyy.yyy<br>
s=FreeSWITCH<br>
c=IN IP4 yyy.yyy.yyy.yyy<br>
t=0 0<br>
m=audio 40562 RTP/AVP 0 8 3 101<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=silenceSupp:off - - - -<br>
a=ptime:20<br>
a=schipmangled:yes <--- rtpproxy added this on initial invite<br>
<br>
...<br>
<br>
U 2013/02/14 19:32:37.425606 xxx.xxx.xxx.xxx:5060 -> yyy.yyy.yyy.yyy:5060<br>
SIP/2.0 200 OK<br>
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=<u></u>z9hG4bK2d9e.187ebf5.0<br>
From: "Unknown" <sip:19999999999@yyy.yyy.yyy.<u></u>yyy>;tag=33XjNy6SQZrQS<br>
To: <sip:19999999999@yyy.yyy.yyy.<u></u>yyy>;tag=SDs07f299-gK0e9f2e8d<br>
Call-ID: 004c5840-f1aa-1230-9c93-<u></u>6320dec8e883<br>
CSeq: 40108106 INVITE<br>
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed<br>
Contact: <sip:xxx.xxx.xxx.xxx;did=39.<u></u>60d51ef><br>
Allow: INVITE,ACK,CANCEL,BYE,<u></u>REGISTER,REFER,INFO,SUBSCRIBE,<u></u>NOTIFY,PRACK,UPDATE,OPTIONS<br>
Require: timer<br>
Supported: timer<br>
Session-Expires: 7200;refresher=uas<br>
Content-Length: 259<br>
Content-Disposition: session; handling=required<br>
Content-Type: application/sdp<br>
<br>
v=0<br>
o=Sonus_UAC 7607 20874 IN IP4 xxx.xxx.xxx.xxx<br>
s=SIP Media Capabilities<br>
c=IN IP4 xxx.xxx.xxx.xxx<br>
t=0 0<br>
m=audio 29772 RTP/AVP 0 101<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=silenceSupp:off - - - -<br>
a=schipmangled:yes <--- they sent this back in the 200 OK reply<br>
a=ptime:20<br>
a=sendrecv<br>
<br>
<br>
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</blockquote></div><br><br clear="all"><br>-- <br><span style="color:rgb(136,136,136);font-family:arial,sans-serif;font-size:13px;background-color:rgb(255,255,255)">Muhammad Shahzad</span><br style="color:rgb(136,136,136);font-family:arial,sans-serif;font-size:13px;background-color:rgb(255,255,255)">
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