Re: [GnomeMeeting-devel-list] RE: [GnomeMeeting-list] News from 2.00

[Damien Sandras <dsandras seconix com>]

Hello,

Le jeudi 04 août 2005 à 18:16 +0800, Cheng LI a écrit :
> Hello,
> 
> I am very interested at:
> Is there any consideration about Lip Synchronization in the design of GM2.00
> or the current GM1.2.1?

Not currently.

> If yes, where should I find the corresponding code, in GM or in Openh323
> package? If not, do you have
> any consideration on this issue for future GM?
> 

It is a very interesting research topic, but I do not plan to work on
that in the future. However, if you have some Open Source code, it will
surely be integrated.

Thank you,

> Best Regards,
> 
> 
> 
> -----Original Message-----
> From: gnomemeeting-list-bounces gnome org
> [mailto:gnomemeeting-list-bounces gnome org] On Behalf Of Damien Sandras
> Sent: Thursday, July 28, 2005 5:45 AM
> To: gnomemeeting-devel-list gnome org
> Cc: gnomemeeting-list gnome org
> Subject: [GnomeMeeting-list] News from 2.00
> 
> 
> Hello to all,
> 
> ---
> I have a few good news concerning the 2.00 release development.
> 
> You probably know that except for video, most of important features are
> already implemented. There were 2 *big* exceptions :
>   - you could not be transferred to a remote endpoint (except when using
> Asterisk which intercepts the call). It is now implemented for SIP.
>   - some proxies like Asterisk issue Re-INVITES during sessions. That allows
> to change the remote IP address/port where to send RTP data, but also the
> codec, during a call. It is now implemented. You can for example be in a
> call with an IP Phone using G.711, the traffic going directly between
> GnomeMeeting and the IP Phone, then the IP Phone user decides to put the
> call on hold. Asterisk will then take the relay and send an MP3 directly to
> GnomeMeeting using another codec than G.711, e.g. GSM (the remote party is
> not the IP Phone anymore, but Asterisk, so a Re-INVITE is issued). That
> feature is unique in the Linux softphone world, and some CISCO IP Phones
> even crash if you are using it, but GnomeMeeting supports it.
> 
> I would say that except for Video (on which Robert is working), the SIP
> features list is almost complete. 
> 
> Basically, here is what remains to do :
> * SIP: bugfixing and stability testing
> * H323: Call Hold and Call Transfer must be reimplemented from OpenH323
> * General: audio codecs and video 
> (Robert has worked on video, and it seems that raw video can already be
> transmitted between 2 SIP/H.323 endpoints without using any codec)
> * GnomeMeeting: Various UI enhancements (Druid, Instant Messenging, ...)
> 
> ---
> 
> Another good news is that a french provider will most probably (I have not
> signed yet) provide a P4 server with 1GB of RAM and 20Mbits/s of bandwidth
> to host the new generation seconix.com. It will be named gnomemeeting.net
> and will host several new services for our users :
>    * A SIP Registrar, allowing each user to have a universal
> @gnomemeeting.net SIP address, callable from anywhere in the world with any
> SIP softphone
>    * A public conference room for audio-only and for a limited number of
> users
>    * Probably VoiceMail, but it is not sure yet
>    * Various other services
> 
> More news to come later,
> 
-- 
 _      Damien Sandras
(o-     GnomeMeeting: http://www.gnomemeeting.org/
//\     FOSDEM 2005 : http://www.fosdem.org
v_/_    H.323 phone : callto:ils.seconix.com/dsandras seconix com