[SR-Users] No audio when calling from SIP to WEBRTC

[Chirag Desai]

Hi all,

I have configured Kamailio for WebSockets following this guide as an
example:
https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamailio/kamailio.cfg

With sip.js and jssip I'm able to initiate a call from WebRTC to SIP and
establish a call successfully.

The issue arises when I try to receive a call from a SIP device. In this
case the call establishes but there is no audio in either direction.

I *think* the issue is with RTP Engine and I've raised a bug there, but I'm
not sure why it is misbehaving
https://github.com/sipwise/rtpengine/issues/983. There are some logs from
RTP engine posted here.

The sip device communicates with Kamailio over UDP / RTP, nothing is
encrypted.

I would appreciate any guidance.

Thanks in advance,

C
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20200504/e814b2c5/attachment.html>