[asterisk-users] G.722: ast_channel_make_compatible failure

[Brian West]

I would like to point out that G.722 is a really awesome codec for  
wideband.  Asterisk has some changes that will need to be made to  
support variable audio rates.  We did this in FreeSWITCH from the  
start.  I think Asterisk will be doing similar things to bridge an 8k  
to 16k channel via resample.  FreeSWITCH can already do this so you  
could use FreeSWITCH in conjunction with Asterisk to solve this for  
now.  FreeSWITCH can also do Wideband conferencing.  In addition you  
can mix and match 8k and 16k conference participants.  Just thought I  
would throw that out there as a way to bridge the gap.

/b

On Oct 5, 2007, at 1:13 AM, Ondrej Valousek wrote:

> Hi Kevin,
>
> Thanks for the answer - Hopefully this feature will be available some
> day. My opinion is, look for a transcoder only if the two (or more)
> parties does not offer any matching codec.
> Good to hear it is being worked on....
> Best regards,
>
> Ondrej
>
> Kevin P. Fleming wrote:
>> Ondrej Valousek wrote:
>>
>>
>>> My problem is, that the phone offering g722 could do alaw as well.
>>> I expected asterisk should just chose alaw for the communication  
>>> - no
>>> transcoding is necessary then...
>>>
>>
>> That is not how Asterisk works, and is well known in the community as
>> something that users would like to see changed, but has not yet been
>> done. Asterisk negotiates the codecs (formats) for each call leg  
>> pretty
>> much independently of the others, so if a G.722 endpoint initiates  
>> the
>> first call leg, and the destination call leg cannot accept G.722, and
>> there is no transcoder available, then the call will fail. If the
>> non-G.722 endpoint initiates the first call leg then the call will
>> likely go through, which is somewhat unfortunate :-)
>>
>>

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