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

[Chris Stone]

On Mon, Feb 7, 2011 at 6:48 PM, Henk Hesselink <opensips-users at voipro.nl> wrote:
> 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
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>