[asterisk-users] Call Recording audio file quality query

[Sherwood McGowan]

On Tue, Feb 8, 2011 at 6:01 AM, <faisal at vopium.com> wrote:

> But if you are getting calls all the way on VoIP then you can have calls in
> HD audio using HD audio codec on all locations (Server and Client). In that
> case you either need use some available 3rd party solution which uses packet
> capturing to trace the calls and record call using packet capture and
> assembling regardless of server as asterisk still will not be able to record
> call in HD but some other switches like FreeSWITCH can do it or you need to
> write your own app like it.
>
>

It's not difficult at all to perform what you're referring to..If you have
the hardware...

A simple way is to have a port on your main network switch/router that will
"firehose" the traffic the device interacts with In case someone reading
this doesn't know, I'm talking about having a port that just makes a copy of
EVERY PACKET that the device "sees" and sends those copies out over the port
that you've set up for the purpose..It just GUSHES data over that
port...like a firehose just gushes out all the water it possibly can... LOL

Anyway, once your data is being mirrored over that firehose, send it to a
dedicated "recording" server...all it has to do is find the signaling
packets for each call and then just dump the "payload" from the RTP. It'll
come out exactly as it was transported within RTP...in the codec the call
set up....

I may be wrong, but I'm fairly sure that Asterisk can write a filetype for
almost any of it's codecs...I know it can READ audio files that are encoded
in GSM, uLaw (ul), aLaw (al), G726 and G729 formats (.g729, g.726)...etc...

If the "DECoding" portion is there, there's almost GOT to be the "enCOding"
functionality...
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