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 > > > ------------------------------------------------------------------------------ > For Developers, A Lot Can Happen In A Second. > Boundary is the first to Know...and Tell You. > Monitor Your Applications in Ultra-Fine Resolution. Try it FREE! > http://p.sf.net/sfu/Boundary-d2dvs2 > _______________________________________________ > Freetel-codec2 mailing list > [email protected] > https://lists.sourceforge.net/lists/listinfo/freetel-codec2 ------------------------------------------------------------------------------ For Developers, A Lot Can Happen In A Second. Boundary is the first to Know...and Tell You. Monitor Your Applications in Ultra-Fine Resolution. Try it FREE! http://p.sf.net/sfu/Boundary-d2dvs2 _______________________________________________ Freetel-codec2 mailing list [email protected] https://lists.sourceforge.net/lists/listinfo/freetel-codec2
