[asterisk-users] audio glitches in conference

[Jeff Brower]

Jonathan-

>> How did you measure the gaps?  Using signal or speech analysis
>> software to display the recording?  If you measure number of samples
>> between the gaps, does it correspond to multiples of RTP packet
>> payload length (for example, for 8 kHz G711 multiples of 80 samples
>> between gaps) ?
>
> I just loaded the file into audacity and measured the gaps by looking at
> the wave form. I just went through some of the samples I recorded
> yesterday, and found that all the gaps are multiples of 8 samples - from
> 8 up to 32. I guess that's because dahdi/zaptel ticks are 8 samples
> each, so if it misses one, 8 samples get lost. I can't imagine that RTP
> is involved, since this is happening with purely local channels (just
> the Playback application and the eagi script)

Did you measure the distance between gaps?  If those distances are multiples of RTP payload length, then possibly a
network latency issue is involved, but otherwise I agree with other posters, it sounds more like a timing and/or
sampling synchronization problem.

Are you handling TDM data, for example T1/E1 or ISDN?  If so, then normally the TDM card inputs the external T1 line
clock and then provides it as a master for software handling TDM data.  If somehow the TDM clock is not configured
this way, or the software (Asterisk) is using another clock, then effectively you have two (2) different TDM clocks
and they are "drifting" with respect to each other.  In this way you could miss bursts of samples (or get incorrect
samples) as the clocks drift to be 180 out.  Then you'd be Ok again for a while until it happens again, and so on...

-Jeff