[asterisk-users] Improving the voice Quality,
Steve Totaro
stotaro at totarotechnologies.com
Fri Oct 3 21:36:44 CDT 2008
BT3 (BackTrack) LiveCD is one of the best things out there, even has sipp
built right in, as well as other great apps, utilities, and security
"auditing".
I suggest everyone have a copy in their arsenal, and it is free of course.
Thanks,
Steve Totaro
On Fri, Oct 3, 2008 at 10:09 PM, Jai Rangi <jprangi at gmail.com> wrote:
> 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.
>>> > >
>>> > >
>>> >
>>> > _______________________________________________
>>> > -- Bandwidth and Colocation Provided by http://www.api-digital.com--
>>> >
>>> > AstriCon 2008 - September 22 - 25 Phoenix, Arizona
>>> > Register Now: http://www.astricon.net
>>> >
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>>> >
>>> >
>>> >
>>> >
>>> ------------------------------------------------------------------------
>>> >
>>> > _______________________________________________
>>> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>> >
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>>> > Register Now: http://www.astricon.net
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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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>>>
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>>> Register Now: http://www.astricon.net
>>>
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>>>
>>
>>
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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