[asterisk-dev] Opus and VP8

[Andrea Suisani]

Hi Lorenzo,

Firstly let me thank you for the work you're doing!

On 05/27/2013 04:09 PM, Lorenzo Miniero wrote:
> Dear all,
>
> I've just published the patch on github:
>
> https://github.com/meetecho/asterisk-opus
>
> The README should be quite self explainatory, but if you need any additional info feel free to ask me.
> Any feedback will be more than welcome!

I've applied the patch to asterisk 11.3.0
and it applies cleanly (modulo a few hunks here
and there).

Then I've tried to place a phone call from
a chrome browser session (webrtc + jssip)
through asterisk. the end point was a mobile
phone. the callee leg of the call was terminated
through a sip provider that use g729 or gms codecs.

I've tested both type of transcoding.

The sound was very choppy on callee side.
whereas the sound on caller side was good.

let me know if you are interested in any kind
of call logs (opus set debug ...).

Andrea