[asterisk-users] Improving the voice Quality,

[Jai Rangi]

Al and Alex,
Thank you for your input,
Sorry TDM is not the option at this time :( .
Everything has been great until last 2-3 days. Machine loads is not the
issue, we have multiple asterisk server to share the load. Not much change
in traffic.

Now it been narrowed down to networking and we are trying to find out where
the issue is?  In our Firewall or SP's router. Does any one know of any tool
to simulate RTP traffic. Its pain to find out the bad calls out of hundreds
of calls.
BTW, What should be right value for tos in sip.conf.
We have
tos=0x68
Dont remember how did I come up with this value.

I found this
http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+tos

tos=0x10 low delaytos=0x08 high throughput tos=0x04 high
reliabilitytos=0x02 ECT
bit set tos=0x01 CE bit set
-Jai


On Fri, Oct 3, 2008 at 4:58 AM, Al Baker <bwentdg at pipeline.com> wrote:

> USE TDM Circuits - Voice Quality Good
>
> Alex Balashov wrote:
> > Jai Rangi wrote:
> >
> >
> >> All,
> >>
> >> I am having audio quality problem in some calls (1-2%) on asterisk. I
> >> captured RTP traffic using ethereal and this is what I found with the
> >> problematic calls. (Worst cases)
> >> Drop by Jitter buff: 25-75%
> >> Out of Seq: 50-100% (100 % means very very poor call quality).
> >>
> >> Has anyone had similar problem? If yes, can you please share your
> >> experience on how did you fix this?
> >>
> >
> > Such poor performance is not fixable.  The network, connectivity issues,
> > machine load, etc. needs to be addressed - the underlying cause, in
> > other words.
> >
> > BTW, 100% out-of-sequence RTP packets leads to a lot more than just
> > "very very poor call quality."  I don't see how the conversation could
> > even be coherent in that situation.
> >
> > What is more likely is that Wireshark's RTP stats are giving you some
> > distorted information.  I've found its stream analysis to be somewhat
> > buggy in that regard.
> >
> >
> >> I was wondering if I can decrease the MTU size to 250-500 on the network
> >> card and use that card only for VoIP traffic. Will this have any bad
> >> effect on sip traffic/packets?
> >>
> >
> > No.  RTP packets are very small - much smaller than that MTU, or any
> > reasonable MTU you could set.
> >
> >
>
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