[asterisk-users] Improving the voice Quality,

[Jai Rangi]

All,

Just an update on this. This turned out to be a bug in Cisco firewall. We
ended up in upgrading the Firmware on the firewall.

One thing I want to add, this was first time we used the fail over unit
during peak time. In the whole process (failover, upgrade and failover back
to active unit) was completely seamless. Did not had any down time, there
was just a pause for just 1 second in the audio. I was very impressed.

-Jai


On Fri, Oct 3, 2008 at 1:21 PM, Jai Rangi <jprangi at gmail.com> wrote:

> 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/
>> Tel    : (+1) (678) 954-0670
>> Direct : (+1) (678) 954-0671
>> Mobile : (+1) (706) 338-8599
>>
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