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


On Wed, 2005-07-27 at 23:44 +0200, Damien Sandras wrote:
> 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 It will be named
> and will host several new services for our users :
>    * A SIP Registrar, allowing each user to have a universal
> 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,

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