[asterisk-users] G.722: ast_channel_make_compatible failure

[Ondrej Valousek]

Hi Brian,

No I do not need g722 that seriously - I just thought - we all in the
company have phones that support it, we are all on switched LAN (so
bandwidth is not a problem) - so why not use it?
At this point I would like to know why you think it is awesome? I know
the are some extensions/improvements to this codec but these are
unfortunately not free so no use for asterisk.

Thanks,
Ondrej

Brian West wrote:
> 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 :-)
>>>
>>>
>>>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users


The information contained in this e-mail and in any attachments is confidential and is designated solely for the attention of the intended recipient(s). If you are not an intended recipient, you must not use, disclose, copy, distribute or retain this e-mail or any part thereof. If you have received this e-mail in error, please notify the sender by return e-mail and delete all copies of this e-mail from your computer system(s).
Please direct any additional queries to: communications at s3group.com.
Thank You.
Silicon and Software Systems Limited. Registered in Ireland no. 378073.
Registered Office: Whelan House, South County Business Park, Leopardstown, Dublin 18