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. > >
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