Re: [GnomeMeeting-list] When Ekiga rings... follow-up



On Thu, 22 Jun 2006 20:33:09 +0200
Fabien Chevalier <fabchevalier free fr> wrote:

> 
> 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:

..deleted

There are intrinsic differences between the way VoIP and normal phone calls
work. Not all of those differences can be overcome.

The "post dial delay" (time between end of dialing and start of ringback)
is always short for PSTN calls, because the PSTN uses a dedicated
circuit switched network. For VoIP, this delay is usually much longer
because of the delays intrinsic to IP networks. You may have to do DNS
lookups, or wait for slow servers, or timeout or retry calls. All of
which takes time. During that time, there is no audio coming from the
remote end.

The PSTN network will always provide ringing/busy/error tones to the
handset. On VoIP networks, this is not guaranteed. This means that a
VoIP handset may have to generate it's own local ringback or error tone,
and that tone may not have any similarity to the tone that you would
hear if you called the same number on the the PSTN. Some gateways may
provide hints if the protocol supports it.

..deleted

> 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 !!)

This problem has existed with VoIP for years, and different vendors have
different solutions.
 
> 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)

This approach will work provided the post-dial-delay is not too large.

>    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.

Many VoIP phones also do this.

>    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.

Or this...

> 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 ??

I would propose the following (and have implemented this myself before)

After the number has been dialed, wait no more than 3 seconds to receive
the 180 Ringing/ALERTING. If it is not received, play a message saying
"Your call is being connected". This message can be repeated
occasionally until one of three things happens:

1) The local user gives up :)

2) The call fails. In this case, play a busy or disconnected tone

3) "180 Ringing" or ALERTING is received and no media is started. In
this case, play a local ringback tone

4) If media of any form is received from the other end at any time then
it must be played to the user.

    Craig

-----------------------------------------------------------------------
 Craig Southeren          Post Increment ? VoIP Consulting and Software
 craigs postincrement com au                   www.postincrement.com.au

 Phone:  +61 243654666      ICQ: #86852844
 Fax:    +61 243656905      MSN: craig_southeren hotmail com
 Mobile: +61 417231046      

 "It takes a man to suffer ignorance and smile.
  Be yourself, no matter what they say."   Sting




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