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

[Ovidiu Sas]

Have you checked if the your config is altering the SDP?
Check the SDP for the same message before coming to your opensips
server and after leaving the opensips server.
If the SDP is altered and the IP in SDP is pointing to your opensips
server, then there is your problem.


Regards,
Ovidiu Sas

On Fri, Feb 4, 2011 at 7:53 PM, Chris Stone <axisml at gmail.com> wrote:
> Hello - We have an Opensips 1.4 server that routes incoming calls to a
> couple of different Asterisk servers and to upstream providers. All
> working great and with the current config, the Opensips server only
> handles the SIP traffic - all of the audio is between the UAs and
> Asterisk servers.
>
> Am building another Opensips server and decided to do it with the
> 1.6.3 release. With virtually the same config (really only had to
> change a couple of things at the top like loading signal.so, dropping
> the loading of xlog.so, etc), now Opensips is in the picture for all
> of the audio as well as SIP traffic. Was troubleshooting an issue with
> no audio in either direction when calling in - so was capturing
> traffic on the Opensips server, saw the SIP traffic, call relayed
> correctly to the Asterisk server where and IVR was played and all the
> audio was going back to the Opensips server. Did the same test on the
> Opensips 1.4 server and no audio packets - which is what I want.
>
> Done a days searching and am not finding the fix. Someone know what
> changed and how I can get the same behavior I have with 1.4 with the
> 1.6 release?
>
>
> Thanks!
>
>
>
> Chris
>
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