[asterisk-dev] Re: Jingle/chan_xmpp progress?

mogorman mogorman at digium.com
Wed Mar 8 14:17:38 MST 2006


>
>
>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)
>  
>
    that is one of the goals, is the ability to act like any other rtp 
channel driver we already have, but it probably wont do that at first.

>   - in what way (if any?) will Asterisk be able to obtain XMPP 
>statuses of other elements?
>
>  
>
    as it is now astjab is capable of doing that and has functions for 
obtaining that information, and a dial plan up xmpp_status()

>   - in what way (if any?) will Asterisk be able to modify or publish 
>presentity status for itself?
>
>  
>
    I have looked deeply into this , and there are some ways for doing 
it, but none of them are too pretty, there is no standard transport for 
doing this like we have with asterisk manager.  The way asterisk-im does 
it currently is with a jive specific component that allows the code 
there to do pretty much anything you could want.  I have thought about 
writing a similar component for ejabberd, as I think it is probably the 
best jabber server out there right now.  However it has been explained 
to me that this might be acomplishable via a component but I havent been 
able to verify it across all servers , I know that it works for jabberd2 
and not for jabberd1.4 though.

>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.
>
>  
>
    It is ready for testing but there is 0 documentation avaialable 
right now, as I am trying to finish the channel driver end, currently 
the channel driver will set up the call but not pass audio : ( .  I hope 
to have something much more playful very soon.  The other features 
however, those in res_xmpp, are very useful and stable.  However there 
are still some random crashes that I have not tracked down do to mall 
formed xml, but any normal server or client should not expose these.

>JT
>


Hope this helps.

Mog



More information about the asterisk-dev mailing list