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
