Hello, I see this was discussed further on rtpengine issue tracker. Did using a newer version of rtpengine made it work?
The typical hint I have is to look at javascript console in the browser, there should be logs printed when some dtls negotiation fails. Cheers, Daniel On 05.05.20 02:21, Chirag Desai wrote: > > 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 > > > _______________________________________________ > Kamailio (SER) - Users Mailing List > [email protected] > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla
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