Oh yes, how could I forgot about that? Thank you, -Jai
On Fri, Oct 3, 2008 at 1:08 PM, Alex Balashov <[EMAIL PROTECTED]>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 <[EMAIL PROTECTED] > > <mailto:[EMAIL PROTECTED]>> 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 > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > -- 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 > > > -- > Alex Balashov > Evariste Systems > Web : http://www.evaristesys.com/ > Tel : (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 > Mobile : (+1) (706) 338-8599 > > _______________________________________________ > -- 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 >
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