|ASTERISK-08121: Close Codec Translation Feature
I am a new user of Asterisk. In my platform, I don?t have any hardware DSP for transcoding functions. Besides, I don?t want to use the software resource to do the transcoding process while the caller and callee use different codec voice.
If the provided codecs of caller are not suitable for callee, the call should be not setup successfully. May I ask a question; is there any possible way to disable the transcoding process? I wish there is only one codec used between caller and callee, and the Asterisk server would not do the transcoding job.
For example, my Asterisk can not support that a dialogue whose caller's codec is g729 and callee?s one is ulaw (or alaw).
We observe that the way to decide the RTP codec for caller and callee in asterisk is that asterisk would create two channels when setting up a dialogue, and than using its own codec respectively to communicate with each other.
The method that the caller's channel to chooses its codec is the first content priority including in (1) user's (or admin's) prefer order in sip.conf and (2) the compatible codec recording in SDP of caller's incoming INVITE.
Then, the callee's channel to decide codec depends on the response ?200 ok? that the callee sends to asterisk server.
However, the designate codec is selected from (1) the caller's channel's codec (Have decided) and (2) user's (or admin's) prefer codec in sip.conf.
Since the caller's channel has been decided its codec before asterisk sever sending INVITE to the callee, it may cause the different codec between caller and callee.
Howerver our request is list below:
When setting up a dialogue, channels of caller and callee won't need to decide their codec until callee sent its 200 ok back.
Then, we can guarantee that the codec between both channels are always the same.
Could you please give us some supports?
|By: Joshua C. Colp (jcolp) 2006-11-13 10:22:03.000-0600
The bug tracker is not a support medium. I will however tell you that not loading a codec translation module will cause Asterisk not to transcode the audio and drop the call. You also have to remember that Asterisk is a PBX, and thus negotiation happens between it and the phone because, well, you might not end up calling another phone. I would suggest going to the asterisk-users mailing list and see how others deal with this issue.