Delay in audio recordings, audio streams out of sync

I’m having a problem with call monitoring. The audio streams are out of sync in my recorded audio files. As we all know, on a phone conversation, there is a full duplex audio signal. That means there are two strains incoming and outgoing. Well, when you record this, you should hear the audio synchronized. When one person speaks and the other responds, the timing should be just like a real conversation. However, something is causing the two streams to be out of sync.

When listening to call recordings, I’ll hear the person on one side of the conversation speak, then the other side doesn’t speak for a few seconds. There is a lag and I’m not sure where to begin trying to figure where it’s coming from.

thanks, everyone in advance for your help.

check the load on your box when recording. If you are also doing this in a virtual machine this is common it the box does not have enough cycles to process things in a timely fashion. VM’s are great for playing and testing but are NOT good for a REAL production box as you do not have control over what the other VM’s are doing at any given time. Good quality audio is a time critical thing and slicing time up (which is what a VM does) just does not cut it in a real world situation.

Hi we have the same issue with the time lag.

On short conversations it isnt so bad but the lag gets worse as the time moves on, i.e. just reviewed a 210sec recording and the other caller was lagging by 16 seconds about 160secs in!

However this ones odd… If i turned the volume up I can faintly hear the synced audio response, in other words say at 160sec the lady actually should respond “hello Chris”, I can at a strain just about hear that, however at an audio level far below mine, then 16secs later the audio responce is reheard “hello Chris” however at an audio level equal to mine!

I can forward an example file if required?

Asterisk (Ver.
PBX-in-a-flash deployed and all sources updated etc regularly!
CentOS 5.3 (Final)
Kernel Vers 2.6.18-164.el5 (SMP)
AMD Sempron 2800+ (1.61 Ghz (256KB Cache))

Could be something codecs used?

Please help,



Anyone this is getting so frustrating now?!

Please help!

Configure specific call queues to record calls. This is done in the queues.conf file, for each individual queue. We set it thus:
. . .
monitor-format = wav
monitor-join = yes

The first line tells Asterisk to record the conversation in the .wav format. This is the best choice because it is most compatible with other operating systems. As archived conversations can be burned to CDs, compatibility is a high priority. The second line tells Asterisk to join the two files (in and out) into one file. If we do not do this, we will only hear half of the conversation.

Hi Blanchae,

We had some help from another user and after updating SOX everything is fine now.