[asterisk-users] How can I improve call quality?
tim at mexuar.com
Fri Apr 20 11:33:26 MST 2007
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
> used 50% max bandwidth). We've no E1/T1 links, everything is IP
> My boss complains that many of the calls he holds with others has a
> quality. He also says that its not just him.
> My iax.conf file has:
> He complains of broken audio, muffled audio, and says compared to
> 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
> but what can I do to tweak/improve the call quality on the A*k box
> 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?
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