[asterisk-users] MixMonitor and Call Latency during conversation

[David Backeberg]

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