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Improving X-Lite sound quality

 
 
Al
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      05-13-2005
On Thu, 12 May 2005 23:16:39 +0100, "Ian" <spam"AT"bathfordhill.co.uk>
wrote:

>
>What I mean is you need at least 128K upload but preferably 256K as with jst
>a upload speed of 64K you will not get a decent voip service.
>This is because if you call a user who only supports G711 and you only
>support GSM your call will fail as you cannot negotiate similar codecs.


I am calling people who use a standard telephone. Neither they nor I
have the faintest idea what "codec" they are using. Surely there must
be a SIMPLE explanation somewhere on what all this "codec" business is
when talking about VoIP? I don't need a full blown technical
explanation - just briefly what it means, why there are so many
versions and how do I use the information in practice.

>So basicly accept the fact that If you are serious about using VoIP you will
>have to upgrade to get an acceptable bandwidth.


I would like to know about "codec" first. Right now it is just a set
of characters which people around here keep using without explaining
exactly what it means and how to use it.

 
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Linus Surguy
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      05-13-2005
Al <> wrote:

>On Thu, 12 May 2005 23:16:39 +0100, "Ian" <spam"AT"bathfordhill.co.uk>
>wrote:
>
>>
>>What I mean is you need at least 128K upload but preferably 256K as with jst
>>a upload speed of 64K you will not get a decent voip service.
>>This is because if you call a user who only supports G711 and you only
>>support GSM your call will fail as you cannot negotiate similar codecs.

>
>I am calling people who use a standard telephone. Neither they nor I
>have the faintest idea what "codec" they are using. Surely there must
>be a SIMPLE explanation somewhere on what all this "codec" business is
>when talking about VoIP? I don't need a full blown technical
>explanation - just briefly what it means, why there are so many
>versions and how do I use the information in practice.
>
>>So basicly accept the fact that If you are serious about using VoIP you will
>>have to upgrade to get an acceptable bandwidth.

>
>I would like to know about "codec" first. Right now it is just a set
>of characters which people around here keep using without explaining
>exactly what it means and how to use it.
>


A 'Codec' is the technique used to encode your voice as a data stream. G.711
codec is basically the same one that is used to encode your voice as digital
data on the PSTN phone network. However, there are others that compress your
voice into less 'space' and therefore need a lower data rate to carry the call,
for example 'GSM' which uses about a quarter of the bandwidth of G.711.

The equipment at both ends of a link needs to support the same codec for the
call to complete.

Linus




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Al
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      05-14-2005
On Fri, 13 May 2005 20:20:18 GMT, Linus Surguy wrote:

>A 'Codec' is the technique used to encode your voice as a data stream. G.711
>codec is basically the same one that is used to encode your voice as digital
>data on the PSTN phone network.


So if I'm calling someone who is using a standard wired phone
connected to a traditional BT line should I use G.711? And what
happens if I use another codec?

>The equipment at both ends of a link needs to support the same codec for the
>call to complete.


Presumably a mobile (GSM) phone uses GSM codec. If so, how is it that
a mobile phone can successfully talk to a traditional fixed line phone
if that uses G.711?

Is there somewhere that gives a simple explanation of all the
different codecs and what they are used for?

 
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Phil Thompson
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      05-14-2005
On Sat, 14 May 2005 21:36:42 +0100, Al <> wrote:

> If so, how is it that
>a mobile phone can successfully talk to a traditional fixed line phone
>if that uses G.711?


I guess it becomes a G.711 signal when it hits the phone system ie the
GSM network or VoIP provider has to take the internally coded stuff
and make it PSTN compatible at the internconnect point (ISDN-30 or the
like)


Phil
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Linus Surguy
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      05-14-2005
Al <> wrote:

>On Fri, 13 May 2005 20:20:18 GMT, Linus Surguy wrote:
>
>>A 'Codec' is the technique used to encode your voice as a data stream. G.711
>>codec is basically the same one that is used to encode your voice as digital
>>data on the PSTN phone network.

>
>So if I'm calling someone who is using a standard wired phone
>connected to a traditional BT line should I use G.711? And what
>happens if I use another codec?


It doesnt matter what you use, as long as the VoIP provider at the other end
supports it. It is their job to convert from one to another. If you use G.711
a-law and call a PSTN phone in Europe, there will be no conversion. If you use
GSM, or G.729 or other compressed codec, the VoIP provider will convert prior to
transmission to the phone network. One reason to use a compressed codec is when
you need to reduce the bandwidth used for the call.

Linus


--
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services. www.magrathea-telecom.co.uk www.uknumber.co.uk www.callthrough.co.uk
www.telesave.co.uk: UK 2.5p/1.5/1p South Africa 6p US,France,Germany,Eire 2.5p
Looking for VoIP ? We're the largest wholesale numbering supplier in the UK!
 
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Al
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      05-18-2005
On Thu, 12 May 2005 21:36:23 GMT, ßodincus wrote:

>On Thu, 12 May 2005 08:27:15 +0100, Al said...
>|My broadband uplink speed is only 64kb. I presume this is the main
>|reason for the fact that people I call using X-Lite softphone tell me
>|they have problems hearing me (even though I can hear them crystal
>|clear). Is it worth my playing with the codecs to try to improve the
>|situation and is there somewhere that describes the pros and cons of
>|the various codecs. Is there any other tuning I can do?
>|
>Actually, if your voice is 'garbled' at the other end, maybe not just a
>bandwidth issue (since the 80K pushed down the line are the MAXIMUM
>PEAK data rate used to deliver your voice, and the average is fair
>less) but a lag instead.


What I can report is that following the comments made here, I have
restricted the codec to GSM (previously I had BOTH GSM and G711a set).
I have to report that now my VoIP calls through X-Lite have improved
dramatically. No one at the other end has complained and the lag I
previously noted seems to have all but disappeared.

So either my broadband connection has dramatically improved, or
exclusive use of the GSM codec has solved it. So thanks to those who
suggested GSM.

It would still be nice if I could find a simple explanation as to how
VoIP hangs together with particular reference to codecs. I know that
in theory the cry could be "google it", but that search currently
produces about a quarter million results. It would be handy if someone
has a recommendation for background reading.
 
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Peter Corlett
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      05-18-2005
Al <> wrote:
[...]
> It would still be nice if I could find a simple explanation as to
> how VoIP hangs together with particular reference to codecs. I know
> that in theory the cry could be "google it", but that search
> currently produces about a quarter million results. It would be
> handy if someone has a recommendation for background reading.


http://www.voip-info.org is mainly Asterisk-based, but there's some
general information there too. Besides, you're going to trip over
Asterisk anyway if you do more than trivial stuff with VoIP so you
might as well read up on it.

The search engine on that stuff is about as useful as a Telewest
teletubby, so you'll find that using Google with "site:voip-info.org"
as one of the query terms is helpful. Try these URLs for a start:

http://www.google.co.uk/search?q=sit...info.org+codec
http://www.voip-info.org/wiki-Codecs
http://www.voip-info.org/wiki-Bandwidth+consumption

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