UKVoIPTalk.com

422 kb/s down

43 kb/s up

Hope that helps

Thanks

"ddd" <dddd.net> wrote in message
news:1178686066.15339.0@despina.uk.clara.net...[color=blue]
> 422 kb/s down
>
> 43 kb/s up
>
> Hope that helps
>
> Thanks
>[/color]


Hi,


Yep... you are right on the limit their with your upstream. I don't know if
you may be able to change away from RADSL (i.e. not have rate adaptive adsl
which reduces upstream to maximise downstream) or try a 256kbp/s type
service which should help maintain / support better upstream.

Anyhow, that would require a deal more consideration, discussion,
investigation, etc.

Turning to the key point. At 43 k you are on the limit - iLBC or Speex (or
possibly GSM or G.729) (I think they are) might just fit - but you have to
allow 10% on either side for protocol overhead, packet headers, etc.

The list a fellow poster suggested seems very good for showing you this:

[url]http://www.en.voipforo.com:80/codec/codecs.php[/url]

(You will see that quite a few will fit within your [call it a "working"] 35
kbps)


I can't claim to know a great deal about Tesco Internet Phone services...
but generally the more "branded" or packaged services tend not to allow so
much low level configuration or setting adjustment. However, some do! You
could look in the user manuals or any accompanying software for "low
bandwidth" or similar options (perhaps some might mention for use with low
bandwidth dial-up 56k modems). Otherwise, I understand that you can use the
physical Tesco Internet Phone with other software / softphones (such as the
very good X-Lite software - I believe / understand) and then you can choose
one of many very good providers (telecoms side providers) and they will
mostly support a range of codecs - typically including the more common g*
series of codecs and sometimes iLBC and speex, etc. You can always
experiment for free using X-Lite and various free test numbers (you do not
have to even register with any to call them and you can try turning on / off
different codecs in X-Lite and see what works well - perhaps for example,
using a free VoIP echo test number, etc.).

Hope some of this is some use - at least to give you a bit more information
or to get you started.

I am sure others here and elsewhere would be able to help you further as
well :) .


Best wishes,



News Reader

"News Reader" <no@email.invalid> wrote in message
news:f1sup1$kgh$1@inews.gazeta.pl...[color=blue]
>
> "ddd" <dddd.net> wrote in message
> news:1178686066.15339.0@despina.uk.clara.net...[color=green]
>> 422 kb/s down
>>
>> 43 kb/s up
>>
>> Hope that helps
>>
>> Thanks
>>[/color]
>
>
> Hi,
>
>
> Yep... you are right on the limit their with your upstream. I don't know
> if you may be able to change away from RADSL (i.e. not have rate adaptive
> adsl which reduces upstream to maximise downstream) or try a 256kbp/s type
> service which should help maintain /[/color]


<SNIP>

[color=blue]
>
> Best wishes,
>
>
>
> News Reader
>
>
>[/color]


P.s. It might be a good idea to double check with your ISP that this is a
genuine issue of line length and not for example capacity problems on their
side, etc.

On 8 May, 15:21, "ddd" <dddd.net> wrote:[color=blue]
> Hi
>
> We live in an area where ADSL is available (512/12), but we are a long way
> from the excahnge, which is reducing our Upload speed. We can hear people,
> but they have problems hearing us
>
> Which VoIP needs the least Upload capacity to work very well?
>
> Thanks folks[/color]

I think iLBC or SpeeX codec can fit your requirement, but it depends
on whether your SIP softphone or SIP client can support it or not.

On 8 May, 15:21, "ddd" <dddd.net> wrote:[color=blue]
> Hi
>
> We live in an area where ADSL is available (512/12), but we are a long way
> from the excahnge, which is reducing our Upload speed. We can hear people,
> but they have problems hearing us
>
> Which VoIP needs the least Upload capacity to work very well?
>
> Thanks folks[/color]

If your SIP softphone can support iLBC or SpeeX but your SIP provider
cannot support them, I think you have to establish your own VOIP
environment. In this case, Asterisk or miniSipServer is a good choice.