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: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Damien Sandras > Sent: Thursday, July 28, 2005 5:45 AM > To: [email protected] > Cc: [email protected] > 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/[EMAIL PROTECTED] _______________________________________________ GnomeMeeting-list mailing list [email protected] http://mail.gnome.org/mailman/listinfo/gnomemeeting-list
