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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