[asterisk-dev] SIT detection in pre-answer audio - guidance for rewriting for trunk

[D Tucny]

Hi Folks,

I had a need to detect SITs, at the time, just on SIP calls in early media,
but, expected that the same requirement could be present on any channel
driver... So... I modified app_dial, added an option to enable inband
progress detection, plugged in a dsp call and during early media forced
transcoding to slin for the dsp's consumption, switching transcoding back on
answer... This was with Asterisk 1.2... It's been running pretty
successfully for the past 6 months, so I want to contribute this
functionality back.

What I could use some feedback on is that the way I've implemented this,
while fine for my needs, probably isn't the 'right way' and does cause
problems with early media being bridged elsewhere when the dsp is attached
due to forcing the transcoding, so while looking to rewrite for trunk I'm
first looking at doing it the right way ...

My thoughts on this are that perhaps duplicating the frames would make sense
and feeding one half through transcoding leaving the original audio frames
intact, but, also wondering whether I'd even need to do this with additions
since 1.2 such as transcode_via_sln and audiohooks perhaps usable to do the
main work there...

Thanks,

d
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