[asterisk-dev] [Code Review] RTP monitoring branch: initial review

[Moises Silva]

>
> Also, it would require *two* SIP sessions, since there are two audio
> streams being sent to the recording server for a single call.
>

ah, I was thinking in terms of sending the mixed audio of the call therefore
needing just 1 sip session.

-- 
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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