[asterisk-dev] RTP/SDP for using unknown codecs

[Beñat Urteaga]

2012/2/23  Burteaga :



>    Hi,

>    This is my first message here, and I'd like to get some help:

>    First of all, I'm using asterisk version 1.8.4.4.



>    For music on hold, I'm trying to send an specific audio format (pcm,  mono, samplerate=32000) to some devices which accept SIP and that audio format.

>    I'm using VLC to capture the audio streamed from a media server and it sends the audio to stdout.
So, I'd like to know:
When asterisk gets that stream from stdout, how does it treat it? I guess it depends on the "format=whatever" you write at the musiconhold.conf file, but not sure...

>    Should I add that format somewhere in the code? Or is it configurable in a specific file?
How does this affect to the SDP payload type negotiation? Would I need to use a dynamic payload type? How/where to configure it?



>    Sorry for such an amount of questions but I've been looking at the review board and got even more confused... :S



>    Thanks a lot!



>    Beñat.



So, none no idea? I don't get where to start. Could anyone please send at least a link or something to help me find out where to start?

Please some light on this dark way...



Thank you!





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