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

[Chris Stone]

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