[asterisk-users] Improving the voice Quality,

[Jai Rangi]

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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