[asterisk-users] audio glitches in conference

[Jonathan Addleman]

Jeff Brower wrote:
> 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.

I just did this, and oddly enough, the distance between gaps *is* 
multiples of 80 - 640, 7360, 8480 samples on the recordings that I just 
tested. I say oddly, because there is no network latency involved - 
everything is local. It's playing a local audio file into a conference 
and recording the audio through the eagi script - nothing else.

> Are you handling TDM data, for example T1/E1 or ISDN?  

I'm not sure what TDM data is, but I'm pretty sure I'm not using it. The 
only telephony thing used in my application is a handful of sip 
connections, but these problems show up even if they aren't used, as 
mentioned above.

Now I'm getting rather confused... why would I be seeing the RTP payload 
length show up up anywhere at all? Maybe asterisk uses it internally for 
something else?

-- 
Jon-o Addleman - http://www.redowl.ca