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

[Beñat Urteaga]

>On 02/27/2012 04:58 AM, Beñat Urteaga wrote:
>>> 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...

> Asterisk 10 already supports signed linear at 32kHz sample rate.

> If you really want to support this in Asterisk 1.8, it can be done;
> adding support for a 'passthrough' codec is not terribly difficult. Do a
> search through the Asterisk source tree for 'G719' (which is supported
> in passthrough and record/playback modes in Asterisk 1.8) and you'll see
> all the places that need to be touched.

THANK YOU VERY MUCH.


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