[asterisk-dev] Jingle/chan_xmpp progress?

[John Todd]

I've seen Matt O'Gorman's SVN branch for chan_xmpp, which has seen 
quite a bit of activity (but I'll admit that I haven't checked out or 
compiled it yet, due to lack of time.)

A few questions for Matt, and the community at large who may have 
comments as well:

   - will the stack have the ability to connect between SIP and Jingle 
audio media, without RTP transport?  (since both SIP and Jingle use 
RTP, it should be possible to do "native" bridging)

   - in what way (if any?) will Asterisk be able to obtain XMPP 
statuses of other elements?

   - in what way (if any?) will Asterisk be able to modify or publish 
presentity status for itself?

I'm really interested in getting this running, so Asterisk can start 
to link into the XMPP network(s) without the kludges that have 
previously been required.  Is this ready to test?  I'd be happy to 
try to test and comment on it if it's ready for basic 
compilation/configuration/testing.

JT