[Asterisk-Users] VoIP bad voice quality

[Andrew Kohlsmith]

On December 16, 2004 12:40 am, Ashish Shinde wrote:
>      We have Asterisk, running on a  P4 box running Suse 9.1, making
> calls using IAX through SimpleTelecom and Nufone. What we are looking
> for is toll quality voice.

Make sure your *routes* to your VOIP providers are stable.

We use Nufone.  Our calls were excellent quality but then one day they started 
acting funny.  Turns out that our route to switch-2.nufone.net was (is 
still?) going through Cogent Networks, the Wal-Mart of carriers.  The route 
to switch-1.nufone.net is still good so we started routing our calls to that 
box instead and our call quality came back to what we expected.

Another thing to make very sure of is that you're uplink (SimpleTelecom) is 
not getting flooded.  I use (what I think is) a most-excellent traffic 
shaping and QoS-aware tc script located at 
http://www.mixdown.ca/~andrew/dump/rc.tc.  It's more robust than WonderShaper 
and combined with an internal PCI ADSL card (Sangoma S518) I am able to 
sustain transfers at my full uplink speed (800kbps) without any degradation 
in my outgoing audio.  Received data I have no (direct) control over but the 
script attempts to control that as well.  With our particular setup we try to 
ensure that nobody can flood the downlink (4032kbps), which would kill our 
received audio.

The reason for an internal ADSL modem is that with every external 
(ethernet-connected) modem we tried (Speedstream, Flowpoint, Hitachi) the 
modem would start buffering like crazy at about 1/2 the uplink speed, so we 
could not sustain more than about 400kbps of upstream traffic without having 
audio break up very badly.

>      The problem is that voice over calls routed through SimpleTelecom
> and nNufone occassionally breaks. We also have a digium card and the
> calls over the digium card using the Zaptel Interface have a very good
> quality.

define "occasionally breaks" ?

>      We have enough bandwidth, the latency to the servers is 100-150ms
> and the packet loss is around 1%. We tried using the G711 and G729
> codec and also have the jitterbuffer enabled.

I really don't think you're going to notice 1% packet loss on g.729 (even with 
lost packet concealment disabled, which it currently is in Asteirsk) -- I use 
the GSM codec exclusively and as I've said before, the audio quality on our 
calls is excellent.

>    How can I solve this problem of voice quality? Can a better
> implementation of jitterbuffer with packet loss concealment  help? If
> so how do I get the newer implementation. I would really like to help
> out in the development of the new jitterbuffer if it has not yet been
> implemented.

Packet Loss Concealment is coming to Asterisk.  The hooks have been placed in 
the latest code (CVS HEAD) but nothing is quite wired up yet.

My first suggestion would be to traceroute out to switch-1 and 
switch-2.nufone.net and see what the routes look like -- how many hops, WHO 
you're going through, etc.  Perhaps you're being routed through poor carriers 
like we were.  Unfortunately I don't think there's much you can do about it 
if you are, aside from trying another VOIP provider.

-A.