Hi All, David Rowe has been working on an Asterisk codec2 module. I believe this will give wider ability to play with realtime codec2 usage and also to hear examples of real transcoding between various other codecs and codec2. Setting up asterisk is fairly easy these days and there are plenty of auto-install distros that can give you a drop in installation (or most linux distros have an asterisk package). Effectively this means that you can use a normal voip client (which doesn't support codec2), but have transcoding to codec2 forced in the middle and hear the effect this will cause
My interest is to test variable sized frames with a view to usage as a voip codec across the internet... However, it may also be very useful as a development tool just to be able to increase the range of clients that can effectively access codec2 (note Asterisk has built-in channel recording, so it's also easy to archive all audio through it). I believe that we are getting close to a testable release real soon, so this is a more of a heads up. If there are people interested and capable of testing the module then it would be interesting to get some feedback after that point? Thanks all Ed W ------------------------------------------------------------------------------ This SF email is sponsosred by: Try Windows Azure free for 90 days Click Here http://p.sf.net/sfu/sfd2d-msazure _______________________________________________ Freetel-codec2 mailing list [email protected] https://lists.sourceforge.net/lists/listinfo/freetel-codec2
