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

[Chris Stone]

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