[asterisk-users] Improving the voice Quality,

[Alex Balashov]

sipp can simulate RTP traffic.

Jai Rangi wrote:

> Al and Alex,
> Thank you for your input,
> Sorry TDM is not the option at this time :( .
> Everything has been great until last 2-3 days. Machine loads is not the 
> issue, we have multiple asterisk server to share the load. Not much 
> change in traffic.
> 
> Now it been narrowed down to networking and we are trying to find out 
> where the issue is?  In our Firewall or SP's router. Does any one know 
> of any tool to simulate RTP traffic. Its pain to find out the bad calls 
> out of hundreds of calls.
> BTW, What should be right value for tos in sip.conf.
> We have
> tos=0x68
> Dont remember how did I come up with this value.
> 
> I found this
> http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+tos
> 
> tos=0x10 	low delay
> tos=0x08 	high throughput
> tos=0x04 	high reliability
> tos=0x02 	ECT bit set
> tos=0x01 	CE bit set
> 
> 
> -Jai
> 
> 
> On Fri, Oct 3, 2008 at 4:58 AM, Al Baker <bwentdg at pipeline.com 
> <mailto:bwentdg at pipeline.com>> wrote:
> 
>     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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-- 
Alex Balashov
Evariste Systems
Web    : http://www.evaristesys.com/
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