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

[Beñat Urteaga]

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.





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