[Asterisk-Dev] codecs pass-thru

[Jonne Kodu]

I wonder how * handles the frames in the following case, when a sip-session 
between a client, my * and a sip-pstn-gw, is using the same codec, e.g. 
ULAW, on both call legs.

Does asterisk receive the ULAW frames and decode them to an internal PCM, to 
code the stream to ULAW again, at the other side, OR does it pass thru the 
RTP frames without touching the audio?

I don't use any t or T parameters, so I don't see any purpose for * to 
touch(decode and code) the audio in this scenario.

Looking forward to get an explanation
/J

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