[Asterisk-Users] Re: Newbie needs advice

Paul digium-list at 9ux.com
Wed Sep 14 15:24:20 MST 2005


jose luis campos wrote:

> Hi All
>  
> Finally, after some reading  ;^), I can do some basic sip channel 
> configuration (two softphones communicating each other), the next step 
> i did was to register with a sip provider (voipjet), and with my free 
> 25cents I call my mother, my girlfriend and some friends (Yeahhh).
>  
> What I notice is that my voice sounds strange, like with interference 
> (how i know this, I also spoke with my phone answer machine lol, hey 
> man Im excited!). The question is, how can I improve the quality of 
> the service, I now that if I rent a T1 I could made it, but all I got 
> is my own 512kbps connection provided by Mexican Monopoly (named 
> Telmex) provider, could someone please explain a roadmap to achieve 
> this goal, make asterisk work better.
>  
> Thank you very much, any hel will be highly appreciated.
>  
> Salu2

Hola Jose,

First thing I will mention is that you posted  to the developer list. 
This is a topic for the user list. So I am replying on the user list and 
cc'ing you in case you haven't subscribed to that list.

Second thing I am wondering is if you are running asterisk or just 
playing with softphones. If you are not running asterisk, you should be 
seeking help in the lists and forums for the softphones you are using.

You need about 90kbps in and out to have a single high quality call in 
progress. You need to learn about codecs if you want to use less 
bandwidth(changes call quality). Be sure that you are not doing anything 
else that consumes all your bandwidth while evaluating call quality. Set 
you softphone to use 711u codec which is high bandwidth and high 
quality. If you still have problems it might be the telmex network 
causing it.

Did you do a traceroute and ping to the voipjet server you are using for 
your calls? That would tell us something about your telmex connection 
quality.




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