asterisk users - Sep 2004 - Garbled voice on long distance calls

I've been having random problems when I make long distance calls using 
either VoicePulse or Nufone.  Sometimes the calls go through clear, and 
other calls (or even just part of a call) the person on the other end 
just hears garbled voice, or really broken up voice.  Sometimes it lasts 
for only a few seconds, but other times it goes on for a few minutes 
until I give up on the call.
At first I thought it was just a problem with our ISP, but I also have a 
Packet 8 phone that works just fine even when Asterisk is having these 
problems.
I get decent ping times to the Nufone or Voicepulse servers (85-200ms) 
and only minimal (0-2%) packet loss.
I've tried connecting with Ulaw, and G726(with Nufone), but it doesn't 
seem to make a difference which codec I connect with, I still get the 
same  garbled sound.
I'm have jitterbuffer=no in iax.conf, but I've also tried it with 
jitterbuffer on as well.
Local calls through the zap card seem just fine.
What could be causing these problems?

I do know what QOS is, and the effect it can have on VOIP, although I'm 
no expert on QOS.  However, I assume that our Packet 8 phone would be 
having the same problems at the same time as *, unless Packet 8 is doing 
something to compensate for packet drops, or jitter that * doesn't do.
I did set tos=lowdelay, but that will only help if the routers between 
me and Nufone honor the request.
I don't really know enough about QOS to monitor it if it's not on my own
network though.  I'll look more into that.

Devon

Marcelo Pacheco wrote:
>Do you know what QOS is ?
>Do you know how lack of QOS can affect VOIP applications ?
>If you don't then I sugest you learn that or pay for someone to teach
you
>first.
>You could be having ocasional packet drops.
>
>Please reply to the list,
>
>Marcelo Pacheco
>
>Em Seg 20 Set 2004 13:04, voc? escreveu:
>  
>
>>I've been having random problems when I make long distance calls
using
>>either VoicePulse or Nufone.  Sometimes the calls go through clear, and
>>other calls (or even just part of a call) the person on the other end
>>just hears garbled voice, or really broken up voice.  Sometimes it lasts
>>for only a few seconds, but other times it goes on for a few minutes
>>until I give up on the call.
>>At first I thought it was just a problem with our ISP, but I also have a
>>Packet 8 phone that works just fine even when Asterisk is having these
>>problems.
>>I get decent ping times to the Nufone or Voicepulse servers (85-200ms)
>>and only minimal (0-2%) packet loss.
>>I've tried connecting with Ulaw, and G726(with Nufone), but it
doesn't
>>seem to make a difference which codec I connect with, I still get the
>>same  garbled sound.
>>I'm have jitterbuffer=no in iax.conf, but I've also tried it
with
>>jitterbuffer on as well.
>>Local calls through the zap card seem just fine.
>>What could be causing these problems?
>>    
>>
>
>
>  
>

On 20 Sep 2004 at 10:50, Devon Stephens wrote:
> I do know what QOS is, and the effect it can have on VOIP, although
> I'm no expert on QOS.  However, I assume that our Packet 8 phone would
> be having the same problems at the same time as *, unless Packet 8 is
> doing something to compensate for packet drops, or jitter that *
> doesn't do. 
The only thing I can think of here is Packet Loss Concealment, which 
is built into some codecs but not yet included in Asterisk.  There 
are developers interested in this, but it would be a major change and 
definitely won't happen till at least after 1.0
>I did set tos=lowdelay, but that will only help if the
> routers between me and Nufone honor the request. I don't really know
> enough about QOS to monitor it if it's not on my own network though. 
> I'll look more into that.
It's unlikely to be running without you setting it up.  There is a 
good resource on the wiki at:

http://www.voip-info.org/tiki-index.php?page=QoS

It really makes a big difference when you download a 100Mb file at 
the same time as multiple voice calls.  TC will put any packets with 
the flag you set (i.e. tos=lowdelay) into the first bin and the rest 
of the traffic should fall into the later bins.  You can also set it 
up so that any traffic coming from asterisk ports/ip is put into the 
first bin.  TC will then only send the latter bins if the first bin 
is empty.  Because your voip traffic is not using the full bandwidth, 
there will be times when the first bin is empty and TC will send the 
contents of bin 2 etc.

Don't forget to set your maximum connection speeds to be slightly 
lower than your actual maximum so as to stop your ISP from buffering 
the data.

Cheers,

Matt Riddell
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