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? 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? Any thoughts? -Thank you, -Jai
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