Yes. The technology need to be used on LAN switches is "port mirroring" or 
"line tapping"



-----Original Message-----
From: "Sherwood McGowan" <sherwood.mcgo...@gmail.com>
Sent: Tuesday, February 8, 2011 7:34am
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] Call Recording audio file quality query


On Tue, Feb 8, 2011 at 6:01 AM, <[mailto:fai...@vopium.com] fai...@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...


--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to