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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