[Asterisk-Users] SIP voice recorder

[Leo Ann Boon]

Rich Adamson wrote:

>I believe that Cisco does the monitoring/recording that way. We've been
>working with a company that has implemented Cisco's approach and they
>are having problems with the recording due to network design (eg, high-
>availability dual-everything. Port mirroring is only picking up half the
>conversation).
>  
>
That's the problem with VoIP voice logging, either your connect
everything to a hub/repeater or you need a SPAN/mirror port. The problem
with SPAN/mirror is the bandwidth. Imagine trying to mirror all traffic
from 24x100Mbps ports into one port. There's an open source logger
www.oreka.org that supports SIP. But, I'm not convinced the system can
scale to large number of concurrent sessions. Victor is looking at 150+
sessions, that's quite large. IMHO, any setup with more than 1 network
switch will require extensive architecture planning. You will either
require cascaded SPAN ports, remote SPAN (Cisco only?), or use 1 NIC (on
your logger) for each network switch.

Other possibilities for logging without SPAN/mirroring:
- Record from a conference server. When the call starts, the CTI layer
will 'invite' the logger into the call and trigger the start of recording.
- Make the endpoints send an extra RTP stream to the logger directly. No
mirroring is required. Nice and clean, but does require custom firmware
for the phones. So far, I only know of 1 PBX vendor that supports this.
- Record from the TDM trunk. Just bypass the IP layer. Drawback - this
approach is unable to record internal calls.
- Record from the switch directly - requires the PBX to sit in the media
path like the Asterisk native monitoring method. At least 1 commercial
PBX supports this method of integrating with a logger.

Leo