Jonne Kodu wrote:
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?
It doesn't decode/recode, but even if it did, the conversions are lossless. It doesn't "pass through the rtp frames", though, just the audio payload from the frames; the frames are unpacked and repacked.
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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