[asterisk-dev] Opus and VP8

[Andrea Suisani]

On 06/05/2013 09:42 AM, Olle E. Johansson wrote:
> 5 jun 2013 kl. 09:37 skrev Andrea Suisani <sickpig at opinioni.net>:
>> Hi all
>>
>> sorry for being late. I've managed to focus again on this subject only
>> this morning and luckily enough Lorenzo have already fix the issue!
>>
>> as James I've tested those scenarios:
>>
>>         (sip+opus)             (iax2+gsm)
>> WebRTC ----------->  Asterisk ------------> pstn
>>
>>
>>         (sip+opus)             (sip+gsm)
>> WebRTC ----------->  Asterisk ------------> pstn
>>
>>
>>         (sip+opus)             (iax2+g729)
>> WebRTC ----------->  Asterisk ------------> pstn
>>
>>
>>         (sip+opus)             (sip+g729)
>> WebRTC ----------->  Asterisk ------------> pstn
>>
>>
>> and everything seems to work flawlessly.
> Thanks for testing.
>
> Now, it would be interesting to see if we can get this call flow going
>
> SIP client with Opus   -----> Asterisk with Opus passthrough (no codec) ------> Sip client with Opus
> (Like WebRTC SIP UAs)

In the next few days I will try to test this conf:

         (sip+opus)             (sip+opus)
WebRTC ----------->  Asterisk ------------> WebRTC

and I'll report back

> Like Matt and Tzafrir mentioned - passthrough support of Opus and VP8 is something that can be
> included in Asterisk now.

yep make sense


Andrea