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

Ovidiu Sas osas at voipembedded.com
Mon Feb 7 18:35:29 CET 2011

You are re-posting the same question again without providing any
additional info:

Ovidiu Sas

On Mon, Feb 7, 2011 at 12:20 PM, Chris Stone <axisml at gmail.com> wrote:
> We have an Opensips 1.4 installation that routes calls to multiple
> Asterisk servers. We have a perl module that Opensips runs that does
> an SQL query to find the Asterisk server that the call should be sent
> to. All works great and Opensips handles only the SIP traffic - all
> the SDP/RTP traffic is between the UAs and the Asterisk servers.
> Getting a new Opensips server ready to go online. Using the same
> config (with minor changes such as the addition of loading signal.so,
> removing xlog.so, etc) and Opensips 1.6.3. In testing, I was finding
> there was no audio (either direction) for calls. Did a packet capture
> on the Asterisk server and Opensips server and found that the outgoing
> SDP/RTP packets were also being routed by Asterisk back to the
> Opensips server and the incoming packets were also going to Opensips.
> This is not what I want - would like the same behavior as we have with
> 1.4 where only the SIP traffic goes through the Opensips server.
> Have done a good amount of research to resolve this and I am not
> finding anything helpful.....
> Can anyone tell me why I am seeing this change in 1.6 v.s. 1.4 and how
> I can get 1.6 to behave the same as with 1.4 with regards to the audio
> traffic?
> Chris
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