[asterisk-dev] DTLS-SRTP SDP correction?

[Joshua Colp]

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...

-- 
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