[OpenSIPS-Users] RTPProxy nortpproxy_str issue

Fri Feb 15 08:20:50 CET 2013

Yes, you can use this method,

http://www.opensips.org/html/docs/modules/devel/textops.html#id250333

e.g. something like,

if (has_body("application/sdp") && replace_body_atonce("a=schipmangle:yes",
""))
   xlog("Removed a=schipmangle:yes from carrier xxx");

Thank you.

On Fri, Feb 15, 2013 at 2:53 AM, Seth Schultz <sschultz at scholarchip.com>wrote:

>  Muhammad,
>
> I don't know what the remote carrier is using for their RTP.  I set a
> custom nortpproxy_str to try and avoid this (instead of leaving it as the
> default a=nortpproxy:yes).  Is it correct for them to leave our custom
> a=schipmangled:yes record in the SDP?  I have had problems with the "f"
> flag and failover routing (basically rewrites the IP in the SDP twice like
> this yyy.yyy.yyy.yyyyyy.yyy.yyy.yyy).  Is there an easy way for me to just
> remove the a=schipmangle:yes in my onreply_route?
>
> Thanks,
> Seth
>
>
> On 2/14/2013 8:28 PM, Muhammad Shahzad wrote:
>
> 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><sip:19999999999 at yyy.yyy.yyy.yyy>
>> ;tag=33XjNy6SQZrQS
>> To: <sip:19999999999 at yyy.yyy.yyy.yyy> <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><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><sip:19999999999 at yyy.yyy.yyy.yyy>
>> ;tag=33XjNy6SQZrQS
>> To: <sip:19999999999 at yyy.yyy.yyy.yyy> <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
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> 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: shari_786pk at hotmail.com
> Email: shaheryarkh at googlemail.com
>
> _______________________________________________
> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
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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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