[Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording

[Matt Florell]

On 9/21/05, Matt Roth <mroth at imminc.com> wrote:

> - What format are you recording to?
> - What codec are the SIP calls being placed over?
> We are recording to the PCM format and using the G711 uLaw codec. High
> voice quality is essential to our application (we are a call center) so
> we partnered with MCI to configure our network for the required
> bandwidth and chose the highest quality, zero compression codec. We
> noload all other codecs in order to avoid transcoding on the switch, so
> we must record to PCM. Later (on a separate server) the recordings are
> mixed to GSM which provides a 5 to 1 compression ratio with very little
> artifacts.


Have you tried recording directly to GSM format? It will help reduce the
bottleneck on disk IO although it will use more CPU cycles(in your case on a
RAM drive this may not help at all)

- We've run into the "Avoided deadlock" recording issues several times
> when trying to do
> - more than 50 concurrent recordings. Changing the ast_channel_lock loop
> from 10 to 20 has
> - helped somewhat reduce the warnings and reduce audio gaps on the
> recordings, but what is
> - really needed for more robust recording is a configurable recording
> buffer that wouldn't
> - freak out if a 10ms delay occurs.
> Are you saying that these messages indicate a gap in a digital
> recording? If so, what is the duration of the gap? If it's comparable
> to a CD skip, I think we can deal with it until a buffer or another
> solution is implemented.


There aren't always audio skips but they do happen more when you get more
ast_channel_walk warnings. The audio gaps are usually less than a quarter
second in our experience but can be upto a second depending on the severity
of the IO problem at that instance. It's very hard to test for until you get
into production and you have real conversations and real people listening to
them that can hear the audio skips.

We have sevaral call centers as well, and we just restrict a single server
to 50 recordings at once and then we would pass the next recording as an
IAX2 channel to another recording server. It's a scalable system for us that
is relatively cheap and works well since we can mix and gsm-encode the audio
on these multiple servers at night when not in production leaving the NSF
server just for storage and not audio processing.

MATT---
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