[asterisk-users] How can I improve call quality?

[Alain Degreffe]

Why do you use Ulaw as codec ?

Try another codec ( g729 is by far the best but isn't free ).
The overhead + the 64 kbps in each direction big if you try a conference
call.
With 3 members, the bandwith is near 300 Kpbs / second

The QOS is handled by which kind of router ? Cisco have a netflow feature
that can detect sip traffic and make a priority but don't forget that the
QOS is only tuned for outbound traffic, never inbound....

Alain



-----Message d'origine-----
De : asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] De la part de Adrian Marsh
Envoyé : Saturday, April 21, 2007 7:06 PM
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : RE: [asterisk-users] How can I improve call quality?

Thanks Tim,

I'd turned it on when I was at a site that had bad internet access...
I'll try turning it off for a while, but I thought it was supposed to
help..

Thanks,
 
Adrian 

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Tim Panton
Sent: 20 April 2007 19:33
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How can I improve call quality?

Try turning the jitterbuffer off, I found that often the endpoints can
do better on their own.


On 20 Apr 2007, at 19:01, Adrian Marsh wrote:

> Hi All,
>
> I've a single 1.2.17 Asterisk system.  Gradwell here in the UK is used
> for PSTN calls via IAX2.
> Our 'net link is a dedicated 2Mb fibre connection (of which we have  
> ever
> used 50% max bandwidth).  We've no E1/T1 links, everything is IP  
> based.
>
> My boss complains that many of the calls he holds with others has a  
> bad
> quality.  He also says that its not just him.
>
> My iax.conf file has:
>
> disallow=all
> allow=ulaw
> allow=alaw
> bandwidth=high
> jitterbuffer=yes
> dropcount=2
> maxjitterbuffer=1000
> maxjitterinterps=10
> resyncthreshold=1000
> maxexcessbuffer=80
> minexcessbuffer=10
> jittershrinkrate=1
> tos=lowdelay
> autokill=yes
>
> He complains of broken audio, muffled audio, and says compared to  
> Skype
> its very poor, particularly during conference calls (zaptel meetme).
> Most of these would be SIP based within our server though, rather than
> IAX/PSTN based (X-lite/SJphone).
>
>
> Obviously I can't do much about the far end IP connections/Mobiles  
> etc,
> but what can I do to tweak/improve the call quality on the A*k box
> itself?
>
> The CPU stays at a constant 10% usage, mainly due to a few other
> monitoring apps on there (with these turned off, its < 2%, but  
> still the
> same issues).
>
>
> Also - are there any useful stats/logs that I can examine to "see" the
> quality of calls?
>
> Thanks,
>
> Adrian
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Tim Panton

www.mexuar.net
www.westhawk.co.uk/



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