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

[Kevin P. Fleming]

On 03/01/2012 05:03 AM, Beñat Urteaga wrote:

> Is there something missing? Must I add anything else at the
> configuration files?
>
> Or the internal_sample_rate has nothing to do with the audio codec?

Correct. The internal sample rate is what ConfBridge uses to mix the 
signed linear audio streams it is handed.

> Then, how is the payload type/codec negotiation done?

Via configuration of the channel driver that you are using for the 
endpoints, in this case, sip.conf for chan_sip. You will need to enable 
'slin32' for the endpoints you are using.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
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