On 23 Apr 2007, at 10:56, Gordon Henderson wrote:
On Mon, 23 Apr 2007, Adrian Marsh wrote:
So which is the best quality?
Gradwells www site lists g711u and g729a, but we currently use
ulaw/alaw
with them too..
ulaw is g711u ...
g711 (u or a), or ulaw or alaw which are the same things
Check first using something like testmyvoip.com to get an idea of your
situation (stress the internet by opening up lots of simultaneous
downloads during the test)
Repeat: Try the above before you do anything else...
Ed W
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Subject: Re: [asterisk-users] How can I improve call quality?
Alain Degreffe wrote:
Why do you use Ulaw as codec ?
Try another codec ( g729 is by far the best but isn't free ).
G.729 isn't the best. Its just the one you need to be compatible with
the other end. G.729
On Mon, 23 Apr 2007, Adrian Marsh wrote:
So which is the best quality?
Gradwells www site lists g711u and g729a, but we currently use ulaw/alaw
with them too..
ulaw is g711u ...
g711 (u or a), or ulaw or alaw which are the same things will give you the
best audio quality, but it's not
Well g.711 (a/u) is essentially the same as PCM (raw) which is used to
traditional circuit switched voice environments.
There are such mechanisms as packet loss concealment, where a
predictive algorithm is used to determine wave forms.
But outside of this, whether you use g.711 or g.729 you
Of Steve
Underwood
Sent: 22 April 2007 09:18
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How can I improve call quality?
Alain Degreffe wrote:
Why do you use Ulaw as codec ?
Try another codec ( g729 is by far the best but isn't free ).
G.729
Steve Totaro wrote:
If I am not mistaking, g711u is ulaw. Ulaw and Alaw are the best
since they are lossless, meaning no
Lossless? Our friends at http://en.wikipedia.org/wiki/Ulaw wouldn't lie. :-)
Steve
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: 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
Alain Degreffe wrote:
Why do you use Ulaw as codec ?
Try another codec ( g729 is by far the best but isn't free ).
G.729 isn't the best. Its just the one you need to be compatible with
the other end. G.729 is the lock-in choice, not the quality choice.
Steve
On Friday 20 April 2007 20:01, 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
, that would help the Meetme function? Maybe
different codecs?
Thanks,
Adrian Marsh
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ed W
Sent: 20 April 2007 19:14
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How can I
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
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
Hi
Also - are there any useful stats/logs that I can examine to see the
quality of calls?
You didn't mention that you have any QOS on your router, so we can
basically guarantee that your problem is the internet connection.
Remember that all the research on networking has been how to
Our 'net link is a dedicated 2Mb fibre connection (of which we have ever
used 50% max bandwidth).
Remember in computer terms this means that you used 100% of the
connection, 50% of the time Your voice will loose out against the
big data packets and spoil the voice quality big time
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
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