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
