Re: [Freetel-codec2] Voip

[Gregory M. <gma...@gm...>]

On Wed, Jan 12, 2011 at 1:14 PM, Sorin Cocorada
<sor...@gm...> wrote:
> Hello everyone,
> I have modified the old speexclient
> (http://svn.xiph.org/trunk/speex/speexclient/) to use codec2.
> Source code is available at the following address
> http://etc.unitbv.ro/~sorin.cocorada/codec2client/
> It uses speex JitterBuffer and RTP encapsulation to facilitate loss analysis
> using Wireshark. Because of the overhead bandwidth usage is about 25kbps.

One frame of delay is probably far less than you would happily
tolerate, so you could decrease the overhead significantly by packing
several codec2 frames per packet.