- 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---
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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