[OpenSIPS-Users] Change in SDP/RTP routing with Opensips 1.4 and 1.6

[Henk Hesselink]

Hi Chris,

That config should't touch the Contact header, and yet that's also been 
modified:

In:  Contact:<sip:+13038382386 at 208.94.157.10 ...
Out: Contact:<sip:+13038382386 at 67.212.153.178 ...

Are you sure nothing else is touching the message?

Regards,

Henk Hesselink


On 08-02-11 02:33, Chris Stone wrote:
> Ovidiu,
>
> On Mon, Feb 7, 2011 at 4:19 PM, Ovidiu Sas<osas at voipembedded.com>  wrote:
>> By default, opensips does not modify the SDP.
>> Double check your config.  If you don't need to touch SDP, make sure
>> that you are not loading nathelper or mediaproxy.  Those are the two
>> modules that are changing SDP.
>
> Made sure neither of these were being loaded and used - mediaproxy
> was, but nathelper was not. I need neither, so removed, restarted
> opensips, tested a call. No change - problem persisted. So, dropped
> down to a bare config:
>
> #-----------------------------------------------------------------------
> debug=9          # debug level (cmd line: -dddddddddd)
> fork=yes
> log_stderror=no  # (cmd line: -E)
>
> children=25
> check_via=no      # (cmd. line: -v)
> dns=off           # (cmd. line: -r)
> rev_dns=off       # (cmd. line: -R)
> port=5060
>
> # for more info: sip_router -h
>
> # ------------------ module loading ----------------------------------
> mpath="/usr/lib64/opensips/modules"
>
> # ----------------- setting module-specific parameters ---------------
>
>
> route{
>          forward("67.212.153.179");
>          exit;
> }
> #-----------------------------------------------------------------------
>
> Restarted OpenSIPS with the above, and problem persists - SDP routing
> modified (apparently) and Opensips proxies the audio.
>
> Incoming from upstream:
>
>      INVITE sip:17204497101 at 67.212.153.178:5060;transport=udp SIP/2.0\r\n
>      From: "STONE C AND C"
> <sip:+13038382386 at 208.94.157.10:5060>;tag=a9d5ed0-13c4-4d509b92-1bc5e644-648c7598\r\n
>      To:<sip:17204497101 at 67.212.153.178:5060>\r\n
>      Call-ID: CXC-260-758763d0-a9d5ed0-13c4-4d509b92-1bc5e643-43d2bf0a at 208.94.157.10\r\n
>      CSeq: 1 INVITE\r\n
>      Via: SIP/2.0/UDP
> 208.94.157.10:5060;branch=z9hG4bK-11150e-4d509b92-1bc5e644-273291f1\r\n
>      Max-Forwards: 69\r\n
>      P-Asserted-Identity: "STONE C AND C  "
> <sip:+13038382386 at cxc.dashcs.com:5060>\r\n
>      Supported: timer,100rel\r\n
>      Content-Disposition: session;handling=required\r\n
>      Contact:<sip:+13038382386 at 208.94.157.10:5060;maddr=208.94.157.10;transport=udp>\r\n
>      Session-Expires: 1800\r\n
>      Content-Type: application/sdp\r\n
>      Content-Length: 238\r\n
>      \r\n
>      v=0\r\n
>      o=Acme_UAS 0 1 IN IP4 208.94.157.10\r\n
>      s=SIP Media Capabilities\r\n
>      c=IN IP4 208.94.157.10\r\n
>      t=0 0\r\n
>      m=audio 22684 RTP/AVP 0 18 101\r\n
>      a=rtpmap:0 PCMU/8000\r\n
>      a=rtpmap:18 G729/8000\r\n
>      a=rtpmap:101 telephone-event/8000\r\n
>      a=maxptime:20\r\n
>      a=sendrecv\r\n
>
> Outgoing to Asterisk:
>
>      INVITE sip:17204497101 at 67.212.153.178:5060;transport=udp SIP/2.0\r\n
>      From: "STONE C AND C"
> <sip:+13038382386 at 208.94.157.10:5060>;tag=a9d5ed0-13c4-4d509b92-1bc5e644-648c7598\r\n
>      To:<sip:17204497101 at 67.212.153.178:5060>\r\n
>      Call-ID: CXC-260-758763d0-a9d5ed0-13c4-4d509b92-1bc5e643-43d2bf0a at 208.94.157.10\r\n
>      CSeq: 1 INVITE\r\n
>      Via: SIP/2.0/UDP
> 67.212.153.178:5060;branch=z9hG4bK-11150e-4d509b92-1bc5e644-273291f1\r\n
>      Via: SIP/2.0/UDP
> 208.94.157.10:5060;branch=z9hG4bK-11150e-4d509b92-1bc5e644-273291f1\r\n
>      Max-Forwards: 69\r\n
>      P-Asserted-Identity: "STONE C AND C  "
> <sip:+13038382386 at cxc.dashcs.com:5060>\r\n
>      Supported: timer,100rel\r\n
>      Content-Disposition: session;handling=required\r\n
>      Contact:<sip:+13038382386 at 67.212.153.178:5060;maddr=208.94.157.10;transport=udp>\r\n
>      Session-Expires: 1800\r\n
>      Content-Type: application/sdp\r\n
>      Content-Length: 240\r\n
>      \r\n
>      v=0\r\n
>      o=Acme_UAS 0 1 IN IP4 67.212.153.178\r\n
>      s=SIP Media Capabilities\r\n
>      c=IN IP4 67.212.153.178\r\n
>      t=0 0\r\n
>      m=audio 22684 RTP/AVP 0 18 101\r\n
>      a=rtpmap:0 PCMU/8000\r\n
>      a=rtpmap:18 G729/8000\r\n
>      a=rtpmap:101 telephone-event/8000\r\n
>      a=maxptime:20\r\n
>      a=sendrecv\r\n
>
> I've got to be missing something stupid - the setup works great under
> 1.4 - would expect as well or better under 1.6 - but appears that
> there's some config option or default that I'm missing....
>
> But, with such a basic config as above, not sure what it would
> be.....Would sure seem that, by some default, OpenSIPS proxies the
> audio, no?
>
>
> Thanks!
>
>
> Chris
>
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