[asterisk-users] Improving the voice Quality,

[Alex Balashov]

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.

-- 
Alex Balashov
Evariste Systems
Web    : http://www.evaristesys.com/
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