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

[Kevin P. Fleming]

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.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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