OK, I have to apologize for re-starting this thread but I'm still catching up on mail from November and December! So...
https://jabberstudio.org/pipermail/jdev/2003-November/017108.html There is a long, long thread starting there. Lots of talk about Speex, H.323, p2p vs. client-server, and so on. As far as I can see, no consensus ever emerged. It seems that people want some kind of voice integration (maybe video too, but I think that's farther out). They want to do 1-to-1 voice chat and maybe even multi-user voice-conferencing. They want to be able to negotiate that over Jabber and then go out of band to do the voice stuff. They want this to work from behind NATs and firewalls. They don't want to open crazy ports in the firewall (or turn off the firewall entirely!) in order to get this done. The only message I posted in that thread pointed out that research indicates people don't actually upgrade from IM to voice or video all that often (by "upgrade" I mean something as simple as picking up the phone or meeting f2f, not necessarily switching from IM to VoIP or whatever). So I still have my doubts about how necessary or important this really is, but I do hear the question more and more: "When is Jabber going to support voice?" It seems to me that first of all we need to get clear on the use cases and requirements. Do we want the ability to negotiate telephone-quality voice chat between two IM users? That seems to be the base case (after all we treat chat and groupchat differently in Jabber, why not treat voicechat and voice-conference differently?). [Of course maybe it is stupid to treat chat and groupchat differently, but we burned that bridge a long, long time ago! :-)] So how do we negotiate one-to-one voicechat via Jabber? Is it just a stream initiation profile (see JEP-0095)? Can we treat this in a similar fashion to file transfer and send data through a SOCKS5 Bytestreams (JEP-0065) proxy as a fallback if p2p won't work? Can SOCKS5 Bytestreams handle something like Speex? I notice in draft-herlein-speex-rtp-profile-02.txt that the author mentions sending Speex data over TCP: This transport type signifies that the content is to be interpreted according to this document if the contents are transmitted over RTP. Should this transport type appear over a lossless streaming protocol such as TCP, the content encapsulation should be interpreted as an Ogg Stream in accordance with RFC 3534, with the exception that the content of the Ogg Stream may be assumed to be Speex audio and Speex audio only. So could we potentially do Speex over TCP using a JEP-0065 proxy (or p2p as defined in that JEP) for voicechat? I realize that it would not work for voice-conference and might not be perfect, but is it possible? Just curious. Again, I'm sorry if we've hashed all this out already -- that was a long thread to catch up on and I am not deeply knowledgeable about this voice/video stuff. /psa _______________________________________________ jdev mailing list [EMAIL PROTECTED] https://jabberstudio.org/mailman/listinfo/jdev
