[asterisk-users] Improving the voice Quality,

[Jai Rangi]

Oh yes, how could I forgot about that?
Thank you,

-Jai


On Fri, Oct 3, 2008 at 1:08 PM, Alex Balashov <abalashov at evaristesys.com>wrote:

> sipp can simulate RTP traffic.
>
> Jai Rangi wrote:
>
> > 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 delay
> > tos=0x08      high throughput
> > tos=0x04      high reliability
> > tos=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
> > <mailto: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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> >
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>
> --
> Alex Balashov
> Evariste Systems
> Web    : http://www.evaristesys.com/
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