[asterisk-dev] Regarding SIP direct media (new topic)

[Brian Candler]

Kevin P. Fleming wrote:
> > ... and in particular, at this point don't offer G729 if you don't
> > have a
> > G729 licence available ...
> 
> At this time there is not method to know whether a G.729 license is
> available or not, and if there was, it would require some method of
> 'reserving' one so that it didn't go away until this call attempt released
> it.

Ah, that's more sophisticated than I was thinking, which was simply that you
wouldn't offer G.729 in the second INVITE when _no_ G.729 licences have been
installed at all (i.e. the common out-of-the-box situation)

Then end-to-end G.729 would work, but would be avoided in situations where
Asterisk knows it would need to transcode the call could not possibly decode
G.729.

e.g.

phone 1                      asterisk                    phone 2

    ----> INVITE (g729, g726-32)
                                    ---> INVITE (g729, g726-32)
                                    <--- 606 Not Acceptable

                                    ---> INVITE (g711, g726-32, gsm...)
                                    <--- 180 Ringing
                                    <--- 200 OK (g711)
    <---- 200 OK (g726-32)

Now, you could probably get the same result by setting disallow=g729 in
phone 2's channel, if you knew that phone 2 didn't implement g729.

The advantage of the above approach is that it's handled automatically;
end-to-end g729 works when it can, and transcoding avoids offering g729 when
Asterisk has no g729 capability.

Running out of licences is a different issue which I'm less worried about,
as presumably g729 users monitor their licence utilisation and buy enough to
cover the peaks.

Regards,

Brian.