[GnomeMeeting-list] When Ekiga rings... follow-up
- From: Fabien Chevalier <fabchevalier free fr>
- To: GnomeMeeting mailing list <gnomemeeting-list gnome org>
- Subject: [GnomeMeeting-list] When Ekiga rings... follow-up
- Date: Thu, 22 Jun 2006 20:33:09 +0200
Hello All,
Many thanks for your answers, suggestions and remarks regarding my
previous post :-)
After having read all your answers, i feel the need to add some more
fuel to the debate :-)
In a form a a small Q&A.
Q: Why the way Ekiga currently rings makes it unusuable (or at least
really painful to use) ?
Given the fact that the number of ring tones variates from one call to
another, i have no way to count the tones to avoid the voicemail.
Being able to avoid the voice mail for me is an important for the
following to cases:
- i want to call the guys i work with that are in a remote location.
However if they are not there, i get redirected to a team assistant. The
fact that the team assistant assists ~ 30 people makes her unlikely to
be of any interest for me. Which means when i call the guy, i give up
after a few tones, and would try later rather than having to explain my
case to this tem assistant. Ok, this case is rather tricky, and is just
too "myself-oriented". But let see the next one.
- when i call my friends on their mobile phone and their mobile phone
is not up, i sometimes don't wanna let a message in there voicemail
either. Using ekiga, i won't be able to avoid their voicemail, and will
be charged like hell by their mobile network operator. (well, like hell
is a bit too much, i know :-) )
Q:Why is it important to solve this issue ?
The SIP provider i use is a broadband Internet provider that has ~ 1.5
Million ADSL subsribers. It is known to be Linux friendly (i.e. it
publicly advertises it uses Linux internally, has contributes back to
VideoLAN project when it launched it's TV service, makes no trouble when
you connect your Linux box to their network). As such it is the
Internet provider most people using Linux use nowadays in France.
The SIP service is free, and provides free calls to landlines in France
and in many Internationnal country. It is likely to be the beginning of
the "SIP revolution" where you don't have to pay anything to call
anybody anywhere.
As this service is still experimental, it hasn't been advertised, and
many people don't use it yet. However when it will be launched in
September, and i wouldn't be surprised to have tens of thousands of
Linux users willing to use Ekiga with it. :-) (For Damien : no it does
not send call progress media... too bad !!)
How to fix the issue ? Here are 3 possible solutions:
1 - Do nothing until we receive the "ringing" SIP message. Then start
ringing. I would tend to believe that the user does not need any
feedback to know that the call is in progress. In fact the user asked
the phone to dial somewhere. It wouldn't make sense for the phone to
*silently* ignore the user action. The phone should throw up un error
message saying that something bad prevented it to start dialing instead.
But i bet that it is already the way Ekiga works, isn't it ? (and by the
way, it is the way most UNIX commands work too :-) - They do not print
anything on success)
2 - Create a new king of tone : a "dialing tone". Play the "dialing
tone" until we receive the "ringing" SIP message. Then play the
"ringing" tone.
3 - implement a Throbber to notify the user that the call is in
progress. Launch the ring tone only when we receive the "ringing" SIP
message.
I would personnally go for 1 or 2. 3 would work too but requires more work.
For your information, SJPhone implements solution 2.
Polls are opens!! What do you guys think of these solutions ??
Cheers,
Fabien
[
Date Prev][
Date Next] [
Thread Prev][
Thread Next]
[
Thread Index]
[
Date Index]
[
Author Index]