[OpenSIPS-Users] RTPProxy nortpproxy_str issue

[Muhammad Shahzad]

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 <sschultz at scholarchip.com>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: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
>
>
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>



-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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