Re: Speex and low bandwidth communication
Tristan Matthews <[email protected]> Sun, 20 Sep 2015 15:55:59 +0200
| Newsgroups | gmane.comp.audio.compression.speex.devel |
|---|---|
| Message-ID | <CAN8HRDnnuhARWZPotw3wDMzJKEofqxr-eoce664Q-xP=FgwLGA@mail.gmail.com> |
Hi, On Fri, Sep 18, 2015 at 2:30 PM, <[email protected]> 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