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
