[asterisk-users] Tracing audio problems

[Erik]

Avi,

We need more info,
Through what means are both sides connected, 1:1 xDSL?
What bandwidth, are you using tunnels (pptp/gre/ipsec), how many concurrent calls etc.
You could try analysing network delay/jitter/packetloss using Smokeping.
Note that on DSL 1 g729 calls uses about 45 kbit/s, alaw uses about 108 kbit on DSL

Erik


Avi Miller wrote:
> Hey guys,
> 
> I need some assistance in tracking down the cause of audio problems that 
> are occurring at two of my sites:
> 
> Both sites run Asterisk 1.2.10 and use Sangoma A101u PRI cards. Both 
> sites are reporting that audio in calls is "dropping out" during words, 
> so that the other caller (i.e. the remote user) can only hear bits of 
> the words.
> 
> This used to only happen on Asterisk-to-Asterisk calls via IAX2 (using 
> g729) so I assumed it was latency or bandwidth problems on the 
> inter-office network. However, the network is hardly used and my 
> round-trip times are sub 100ms according to iax2 show peers (with 
> qualify=yes).
> 
> Then, thinking it might be g729 issues, I changed the entire system to 
> only use alaw and the problem persists.
> 
> Does anyone have any suggestions on where to look next? My users are 
> getting increasingly annoyed and I'm quickly running out of ideas.
> 
> Thanks,
> Avi
>