I'm familiar with Freeswitch, not Asterisk. So, I don't know my comment will be applicable there.
But, could you explain your signaling path a little. Is websocket being handled by Asterisk or somebody else in between. In my case, there is Kamailio in between FS and webRTC client. So, Freeswitch was modifying the SDP to non-webRTC, so called webRTC client rejected the call. I had to set FS to media proxy mode to stop it from modifying SDP. Thanks, Dipak On Mon, Feb 10, 2014 at 8:00 AM, jaflong jaflong <jafl...@yandex.com> wrote: > I am having problems with calls from webrtc to kamailio forwarded to > Asterisk > > These are snippet of the debug logs > > Asterisk > > CSeq: 4910 BYE > Reason: SIP ;cause=488; text="Not Acceptable Here" > Supported: path, outbound, gruu > User-Agent: JsSIP 0.3.0 > Content-Length: 0 > > > Jssip > > Cause: Bad Media Description > Origin: remote > > > > Searching on google I get some indication this is to do with ice config? > Please can some one suggest if this is so. > > In my scenerio the webrt clients will only call to the asterisk server > (and not to other user agent). > Considersing this I think maybe can do without ice. > > Is it possbile to disable ice. > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > -- Thanks, Dipak
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