asterisk users - Nov 2009 - MixMonitor and Call Latency during conversation

Hi,

 

We are using MixMonitor to record the call. When the call is bridged, the
latency is significant. We tried to increase the internet speed and the
server RAM and processor speed and still we are having that issue.

 

We use VoiceTrading and Gafachi's Termination minutes to make calls. As we
are in US and VoiceTrading in Europe, somebody suggested to move the
termination minute provider to within USA. So, we bought the minutes from
Gafachi. Still we are having the call latency issues. 

 

$ConversationFile
$ConversationPath."conv_"."$CallQID-$ConversationID.wav";

$self->agi->answer();

$self->agi->exec("MixMonitor",
"$ConversationFile|ba");

 

Any suggestions would be greatly appreciated.

 

Thanks a lot

 

Bharath

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20091116/1c969f71/attachment.htm

On Mon, Nov 16, 2009 at 9:40 AM, Bharath B. Reddy Bynagari
<bynagari at mavensphere.com> wrote:
> We are using MixMonitor to record the call. When the call is bridged, the
> latency is significant.
> $ConversationFile >
$ConversationPath."conv_"."$CallQID-$ConversationID.wav";
>
> $self->agi->answer();
>
> $self->agi->exec("MixMonitor",
"$ConversationFile|ba");
You're obviously using SIP. I don't like to admit it, but I've seen
this problem before.

Please try modifying the voice-activity-detection sections of your SIP
settings and see if this fixes the problems.

My hunch, which is not proven, is that when SIP silence detection
thinks it should stop transmitting packets, the recording module
thinks it shouldn't record the lack of voice transmission, and then
the timing in the recording gets farther and farther from the truth
the longer the call goes on.

in asterisk.conf
transmit_silence = yes
transmit_silence_during_record = yes

in dsp.conf
silencethreshold=1000

in codecs.conf
vad => false
pp_vad => false