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

[Tzafrir Cohen]

On Wed, Oct 07, 2009 at 07:14:11PM -0400, Moises Silva wrote:
> Seem like my past msg did not go thru .... second attempt ....
> 
> At the moment the code basically works and now I want to figure out how
> >> to best fit it in Asterisk.
> >>
> >>
> Hi Tzafrir,
> 
> This might be a silly question and I'm sure you had contemplated this option
> before, but I'd like to know what made you discard it (notice I am not
> familiar with res_monitor and this suggestion is more oriented to mix
> monitor/audio hooks).

I recall that audiohooks was an obvious candidate. I'm trying to recall
why it wasn't used...

> 
> >From the very first time I saw your branch seemed like a cool project to me,
> something I did not understand though, is why not create a regular SIP call
> instead of a dummy call (that is calling ast_request(), ast_call() etc to a
> configured SIP peer), then anything read from the monitored channel would be
> sent using ast_write() and audio coming back from the SIP call would be
> silently discarded ( may be you decided is a waste? ).

1. There is no real SIP server on the other side. The recording server
is passive.

2. Even if there were someone who responded, maintaining a proper
session would be a waste of time and resources. I need to start pushing
recorded data ASAP.

-- 
               Tzafrir Cohen
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