On 13/04/2012 16:41, Anthony Cutler wrote:
What about a Codec2 conference call? Setup a VOIP server and we all
dial in and test out the latest tweaks. With this latest Astrisk
venture this can be setup relatively simply.
A regular conference call offers several things for David and developers:
1) A chance to demo and beta-test the latest tweaks to Codec2
2) A sampling of voices of all ages and accents, from both genders.
3) An **Active Demo** (in real-time and with other live people) of
Codec2 to people who are interested in Codec2's performance.
#3 offers a stupendous marketing opportunity at absolutely no cost,
except one's internet connection and a little time. Practically
everyone on the internet has access to a computer with a mic input
and an audio output. Offering a centrally located, easy to use,
interactive demo of Codec2 with other live people would push the reach
of Codec2 to a much broader audience.
This is an excellent idea, but having investigated the asterisk manual
I'm a little unsure that it would be possible to achieve without a
patched asterisk (which I guess is also feasible)
The issue is that Asterisk a little too smart and will transcode as
little as possible (basically it's going to just bridge calls). So to
force the transcode you need to have a leg in the middle where both
sides are Codec2. I think the main ways to set this up are to have a
"trunk" either with two asterisk machines (or it should be possible to
trunk out and back into the same box). The trunk is set to use only
Codec2, then each user is set to use their preferred codec (say G711)
and now as long as each user rings each other across the trunk you force
the transcoding (simples...)
If someone smarter than me knows how to force an intermediate transcode
then please chime in? It seems possible that astconf or whatever the
conference app is now named, could also be patched to achieve this, but
not investigated
Please would as many folks as possible give this new code a test -
whilst I think it will not be widely used around the asterisk community,
for the people interested in Codec2 performance it probably represents a
great way to test and demonstrate (also asterisk makes it simple to
record calls, so you can easily archive voice samples)
Thanks
Ed W
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