[Asterisk-Dev] Transcoding processing flow

[Gregory Junker]

Just want to make sure I am getting this correct in terms of the 
processing flow from SIP channels to VM files:

PSTN --> SPA3K (G711u) --> SIP --> * (chan_sip) --> u-Law to Slin --> 
(VM app) --> Slin to wav49/gsm/wav --> filesystem

- Coding is done frame by frame, frame size (in samples) defined in 
rtp.c (for streams coming off of SIP channels anyway)
- Everything in the PBX is translated into signed linear before being 
transcoded into some other format (unless the src and dest formats are 
the same, I assume...is this a safe assumption? I haven't checked as of 
this email)
- The GSM codec is the libgsm made available by Jutta via tu-berlin.de
- If a frame is 65 bytes long then it is assumed to be MS-GSM, and is 
shifted before being broken into standard GSM frames (but not recoded). 
If it's set to 33 bytes then it's assumed to be standard GSM and just 
written
- The PCM data in the .wav files is direct translation from the s-lin 
frames, at 8kHz/16bit

Am I on track so far, those that know for sure?

Thanks
Greg