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.
Again, supurb work all!
Anthony
________________________________
From: Ed W <[email protected]>
To: [email protected]
Sent: Friday, 13 April 2012, 7:48
Subject: Re: [Freetel-codec2] Asterisk support for Codec 2
Just to followup on what David has done here:
You can install the asterisk VOIP server, then setup some normal
telephone extensions, eg a normal voip client or a desktop. Then, with
a bit of fiddling, you can achieve the effect that the normal handsets
(which don't support Codec2 of course), talk normally to the Asterisk
server, asterisk converts to Codec2, then back to high quality and on to
the second handset. Net effect is that you can use any handy voip
client to listen in on the effect Codec2 has to your audio.
I think this could be very interesting for tuning the codec because it
makes it relatively simple to "get someone on a call" and so hear audio
samples with a range of voices, background conditions, etc
For VOIP I think this is a very niche situation. *IF* you have a narrow
band internet connection, AND that connection supports IP header
compression, then the header overhead is only 1-4 bytes. On a normal
internet connection the overhead is perhaps 2-3x the size of the voip
packet... So for my expensive satellite link which does appear to
support header compression, plus if we increase the frame size, this is
potentially a really interesting codec that might let us offer low
bandwidth calling (Don't forget there is already G729, GSM, G723.1 and
even low rate Speex - for non niche situations these are probably
acceptable and very widely supported)
OK, first person to try for a second of audio via SMS message?
Thanks David!
Ed W
On 12/04/2012 23:37, David Rowe wrote:
> Hello,
>
> Ed W has kindly sponsored the development of Asterisk support for Codec
> 2. This involved some patching of the Asterisk core, and the
> development of an Asterisk codec module "codec_codec2.so".
>
> This has some applications where header compression and trunking
> multiple calls can make the bandwidth savings possible with Codec 2
> useful for VOIP.
>
> I have a first pass of Codec 2 support for Asterisk running, and have
> made a few calls. First time I have actually had a real-time
> conversations over Codec 2. Usually I just listen to samples, so this
> system is useful for conversational testing alone.
>
> I talked to my 6 year old to see how it handles children. I think they
> actually sound better than males, which have a clicky artefact to them.
> The levels were a bit low for me, not sure if that is due to the codec
> or my SIP phones.
>
> Its given me some ideas for tuning Codec 2 that I will look at over this
> year.
>
> Instructions for building and testing are here:
>
>
> https://freetel.svn.sourceforge.net/svnroot/freetel/codec2-dev/asterisk/README
>
> Cheers,
>
> David
>
>
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