[OpenSIPS-Users] RTPProxy nortpproxy_str issue

[Seth Schultz]

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:19999999999 at xxx.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK2d9e.187ebf5.0
Max-Forwards: 69
From: "Unknown" <sip:19999999999 at yyy.yyy.yyy.yyy>;tag=33XjNy6SQZrQS
To: <sip:19999999999 at yyy.yyy.yyy.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:19999999999 at yyy.yyy.yyy.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:19999999999 at yyy.yyy.yyy.yyy>;tag=33XjNy6SQZrQS
To: <sip:19999999999 at yyy.yyy.yyy.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