On Wed, Nov 28, 2012 at 6:26 AM, Klaus Darilion <klaus.mailingli...@pernau.at> wrote: > Am 27.11.2012 19:41, schrieb Peter Dunkley: > >>> We can also use jssip library. They have some demo to try. >> >> That won't fix his testing with non-WebSocket/browser client problems. >> >> Peter >> > I just tried jssip in Chrome with Asterisk (directly and via Kamailio): > signaling work (no intensive testing) but audio does not work due to a bug > in Chrome. I also tried Opera 12.11 and Firefox nightly 2012-11-27 but it > sems that both do not support webrtc at all. Seems like we can only test > Chrome vs. Chrome.
You can try doubango's patch for asterisk, the instruction is on their wiki. It works for me with chrome 23. > > regards > Klaus > > > _______________________________________________ > 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 -- Iwan Budi Kusnanto _______________________________________________ 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
