[asterisk-users] Call Recording audio file quality query

[Sherwood McGowan]

yep..that would be what i said, using the nifty slang my "peeps" use in the
datacenters....

I just wanted to be "cool" like them...*hangs head*...
great...now I gotta transfer to another school...

LOL, have a good one mate!

On Tue, Feb 8, 2011 at 7:23 AM, <faisal at vopium.com> wrote:

> Yes. The technology need to be used on LAN switches is "port mirroring" or
> "line tapping"
>
>
>
>
> -----Original Message-----
> From: "Sherwood McGowan" <sherwood.mcgowan at gmail.com>
> Sent: Tuesday, February 8, 2011 7:34am
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <
> asterisk-users at lists.digium.com>
> Subject: Re: [asterisk-users] Call Recording audio file quality query
>
> 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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