[Speex-dev] Speex and low bandwidth communication

[Tristan Matthews]

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