<div dir="ltr">Oh yes, how could I forgot about that? <br>Thank you,<br><br>-Jai<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">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 class="Ih2E3d"><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">bwentdg@pipeline.com</a><br>
</div><div><div></div><div class="Wj3C7c">> <mailto:<a href="mailto:bwentdg@pipeline.com">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>
> _______________________________________________<br>
> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
><br>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona<br>
> Register Now: <a href="http://www.astricon.net" target="_blank">http://www.astricon.net</a><br>
><br>
> asterisk-users mailing list<br>
> To UNSUBSCRIBE or update options visit:<br>
> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
><br>
><br>
><br>
</div></div>> ------------------------------------------------------------------------<br>
<div class="Ih2E3d">><br>
> _______________________________________________<br>
> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
><br>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona<br>
> Register Now: <a href="http://www.astricon.net" target="_blank">http://www.astricon.net</a><br>
><br>
> asterisk-users mailing list<br>
> To UNSUBSCRIBE or update options visit:<br>
> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
<br>
<br>
</div><div class="Ih2E3d">--<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>
_______________________________________________<br>
</div><div><div></div><div class="Wj3C7c">-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
<br>
AstriCon 2008 - September 22 - 25 Phoenix, Arizona<br>
Register Now: <a href="http://www.astricon.net" target="_blank">http://www.astricon.net</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
</div></div></blockquote></div><br></div>