[asterisk-dev] Opus and VP8

[Andrea Suisani]

On 06/06/2013 09:49 AM, Andrea Suisani wrote:
> On 06/05/2013 09:08 PM, Andrea Suisani wrote:
>> On Wed, June 5, 2013 5:53 pm, Lorenzo Miniero wrote:
>>> Hi Andrea,
>>
>>>> [cut explanation of a maybe false theory]
>
> [cut]
>
>>> The same scenario, in my setup, works fine: a browser calling a softphone
>>> using a narrowband codec (e.g., u-Law) is capped to 8kHz, opus<->8000.
>>>
>>> I guess the only difference between our scenarios (except for the several
>>> MACROS that are not in my test extension) is the protocol: in my
>>> extensions, only SIP is involved, and not IAX2. This may be what is
>>> causing
>>> the issue, as recently someone posted a similar problem on github:
>>>
>>> https://github.com/meetecho/asterisk-opus/issues/1#issuecomment-18926261
>>>
>>> Unfortunately I'm unfamiliar with IAX, so I don't know how codecs are
>>> negotiated and the related translation paths put in place for a call.
>>
>>
>> many thanks Lorenzo tomorrow I will try to test using SIP protocol
>> on both call legs.
>
> done. still 48kHz encoder/decoder. Unfortunately today I've no time
> to dig it, hope to get a few hours tomorrow. this is the console
> log (opus set debug huge)
>

did another two tests before switching to another task:

1) force the second leg (asterisk ---> pstn/mobile) to use
    a gsm codec, still 48kHz

2) all in the same lan (asterisk and webrtc clients)


            sip+opus                sip+ulaw
   webrtc <---------->  asterisk <------------> webrtc


   but still 48kHz

3)
            sip+opus                sip+opus
   webrtc <---------->  asterisk <------------> webrtc

   no transcoding happening but from a quick iftop check
   it seems to take 50/55kbps (both up/downstream)


maybe it's chrome fault. my version is Version 29.0.1521.3 dev.
what's yours?

Andrea