[asterisk-dev] DTLS-SRTP SDP correction?

[Lorenzo Miniero]

2014-01-28 Joshua Colp <jcolp at 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
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