2014-01-28 Joshua Colp <jc...@digium.com> > On 14-01-28 08:47 AM, Daniel Pocock wrote: > > > > Is that what Firefox is trying to do with the SDP it sends on INVITE? > > I don't know. I don't know what spec they are trying to follow. > > > > > I implemented a quick hack in JSCommunicator that tweaks the SDP from > > Firefox into what Asterisk expects. This makes it work. Here is the > > commit: > > > https://github.com/opentelecoms-org/jscommunicator/commit/6980f8e1c3311c46154b3840d695f0ddc9c8c8ae > > > > However, I didn't want to keep that in my code in the long term. It is > > just to make the sip5060.net/test-calls work for as many people as > possible. > > > > Do you believe this should be fixed by some change in Firefox or is it > > something that would potentially have to be implemented in Asterisk? > > I would say Firefox, but in the time since the original code in Asterisk > was written yet another RFC could have come to fruition that they are > following... > >
Bot Chrome and Asterisk use RTP/SAVPF, I guess following what http://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-11 mandates right now. Anyway, since DTLS is involved, Asterisk is doing the right thing: the draft currently mandates DTLS but doesn't clarify that UDP/TLS/RTP/SAVPF must be used instead. That said, the workaround Daniel described is the same that others, myself included, do at the signalling level (e.g., in JavaScript or in a signalling gateway), and it's easy enough to not require any change in Asterisk IMHO. Lorenzo -- > Joshua Colp > Digium, Inc. | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at: www.digium.com & www.asterisk.org > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev >
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