RE: [GnomeMeeting-list] News from 2.00

[Cheng LI <cheng li st com>]

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?
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?

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

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