[asterisk-users] Improving the voice Quality,

[Al Baker]

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.
>
>