<div dir="ltr">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".<br><br>I suggest everyone have a copy in their arsenal, and it is free of course.<br>
<br>Thanks,<br>Steve Totaro<br><br><div class="gmail_quote">On Fri, Oct 3, 2008 at 10:09 PM, Jai Rangi <span dir="ltr"><<a href="mailto:jprangi@gmail.com">jprangi@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div dir="ltr">All,<br><br>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. <br><br>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. <br>
<font color="#888888">
<br>-Jai</font><div><div></div><div class="Wj3C7c"><br><br><br><div class="gmail_quote">On Fri, Oct 3, 2008 at 1:21 PM, Jai Rangi <span dir="ltr"><<a href="mailto:jprangi@gmail.com" target="_blank">jprangi@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div dir="ltr">Oh yes, how could I forgot about that? <br>Thank you,<br><font color="#888888"><br>-Jai</font><div><div></div><div><br><br><br><div class="gmail_quote">On Fri, Oct 3, 2008 at 1:08 PM, Alex Balashov <span dir="ltr"><<a href="mailto:abalashov@evaristesys.com" target="_blank">abalashov@evaristesys.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">sipp can simulate RTP traffic.<br>
<div><br>
Jai Rangi wrote:<br>
<br>
> Al and Alex,<br>
> Thank you for your input,<br>
> Sorry TDM is not the option at this time :( .<br>
> Everything has been great until last 2-3 days. Machine loads is not the<br>
> issue, we have multiple asterisk server to share the load. Not much<br>
> change in traffic.<br>
><br>
> Now it been narrowed down to networking and we are trying to find out<br>
> where the issue is? In our Firewall or SP's router. Does any one know<br>
> of any tool to simulate RTP traffic. Its pain to find out the bad calls<br>
> out of hundreds of calls.<br>
> BTW, What should be right value for tos in sip.conf.<br>
> We have<br>
> tos=0x68<br>
> Dont remember how did I come up with this value.<br>
><br>
> I found this<br>
> <a href="http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+tos" target="_blank">http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+tos</a><br>
><br>
> tos=0x10 low delay<br>
> tos=0x08 high throughput<br>
> tos=0x04 high reliability<br>
> tos=0x02 ECT bit set<br>
> tos=0x01 CE bit set<br>
><br>
><br>
> -Jai<br>
><br>
><br>
> On Fri, Oct 3, 2008 at 4:58 AM, Al Baker <<a href="mailto:bwentdg@pipeline.com" target="_blank">bwentdg@pipeline.com</a><br>
</div><div><div></div><div>> <mailto:<a href="mailto:bwentdg@pipeline.com" target="_blank">bwentdg@pipeline.com</a>>> wrote:<br>
><br>
> USE TDM Circuits - Voice Quality Good<br>
><br>
> Alex Balashov wrote:<br>
> > Jai Rangi wrote:<br>
> ><br>
> ><br>
> >> All,<br>
> >><br>
> >> I am having audio quality problem in some calls (1-2%) on<br>
> asterisk. I<br>
> >> captured RTP traffic using ethereal and this is what I found<br>
> with the<br>
> >> problematic calls. (Worst cases)<br>
> >> Drop by Jitter buff: 25-75%<br>
> >> Out of Seq: 50-100% (100 % means very very poor call quality).<br>
> >><br>
> >> Has anyone had similar problem? If yes, can you please share your<br>
> >> experience on how did you fix this?<br>
> >><br>
> ><br>
> > Such poor performance is not fixable. The network, connectivity<br>
> issues,<br>
> > machine load, etc. needs to be addressed - the underlying cause, in<br>
> > other words.<br>
> ><br>
> > BTW, 100% out-of-sequence RTP packets leads to a lot more than just<br>
> > "very very poor call quality." I don't see how the conversation<br>
> could<br>
> > even be coherent in that situation.<br>
> ><br>
> > What is more likely is that Wireshark's RTP stats are giving you some<br>
> > distorted information. I've found its stream analysis to be somewhat<br>
> > buggy in that regard.<br>
> ><br>
> ><br>
> >> I was wondering if I can decrease the MTU size to 250-500 on the<br>
> network<br>
> >> card and use that card only for VoIP traffic. Will this have any bad<br>
> >> effect on sip traffic/packets?<br>
> >><br>
> ><br>
> > No. RTP packets are very small - much smaller than that MTU, or any<br>
> > reasonable MTU you could set.<br>
> ><br>
> ><br>
><br>
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</div></div>> ------------------------------------------------------------------------<br>
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<br>
</div><div>--<br>
Alex Balashov<br>
Evariste Systems<br>
Web : <a href="http://www.evaristesys.com/" target="_blank">http://www.evaristesys.com/</a><br>
Tel : (+1) (678) 954-0670<br>
Direct : (+1) (678) 954-0671<br>
Mobile : (+1) (706) 338-8599<br>
<br>
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</blockquote></div><br></div></div></div>
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+18887771888 (Toll Free)<br>+12409381212 (Cell)<br>+12024369784 (Skype)<br>
</div>