Speex dev - Sep 2015 - Speex and low bandwidth communication

Hi,

I'm fairly new on codecs, I'm trying to implement a communication 
between two PCs.

The data rate should be of approx. 6Kbits/s, this 6Kbits/s can just be 
the useful data. Any encapsulation (for example UDP , RTP, etc ...) can 
be present even if that means to rise the overall bit rate.

Also if this rate can be achieved on Asterisk. For exemple I've tried to 
change de rate on a FrePBX distribution and on asterisk compiled by 
myself, but with no success ( I've tried to modify the codecs.conf) i 
always get stucked on IAX/SIP frames sending 21 Bytes each 20ms, so i 
get a useful data rate of 8.4Kbits/s.

Or

if it can be acheived using an method of audio streaming for example 
(Using GStreamer from a Linux machine to a Windows Machine).

Any ideas on how to achieve a communication at 6Kbits/s without counting 
the protocol layers?

Thanks for any help you can lend me.

I might be sending my question to the wrong place, If so can you suggest 
me where should I post it please? I found this addresse on the following 
PDF document : http://www.speex.org/docs/manual/speex-manual.pdf

Miguel SEGOVIANO

Hi,

On Fri, Sep 18, 2015 at 2:30 PM,  <miguel.segoviano at actiled.com>
wrote:
> Hi,
>
> I'm fairly new on codecs, I'm trying to implement a communication
> between two PCs.
>
> The data rate should be of approx. 6Kbits/s, this 6Kbits/s can just be
> the useful data. Any encapsulation (for example UDP , RTP, etc ...) can
> be present even if that means to rise the overall bit rate.
>
> Also if this rate can be achieved on Asterisk. For exemple I've tried
to
> change de rate on a FrePBX distribution and on asterisk compiled by
> myself, but with no success ( I've tried to modify the codecs.conf) i
> always get stucked on IAX/SIP frames sending 21 Bytes each 20ms, so i
> get a useful data rate of 8.4Kbits/s.
>
According to http://www.speex.org/comparison/ you should be able to
achieve rates as low as 2.15Kbits/s using narrow-band mode and
4Kbits/s for wide-band. Of course RTP headers will add some overhead
but this should be minimal.

You may find this page useful for ensuring correct Asterisk configuration:
http://www.voip-info.org/wiki/view/Asterisk+config+codecs.conf

If you only want to transmit audio and don't care about implementing a
VoIP stack, you might find that Asterisk/IAX/SIP is overkill.

Finally, for very low bitrate communication, Codec2 might better serve
your needs: http://www.rowetel.com/blog/?page_id=452

Best,
Tristan