Hi!
I stumbled upon http://jitsi.org/index.php/GSOC2011/KamailioJingle What is the current status? Has the project advanced?
Jingle support would make my life way easier. If time / alternative summer job permits, all I care for is the functionality. Doing that while google pays would be more like sweet bonus.
Thanks for answer.
2011/12/13 kasip@elektromaniak.wz.cz:
I stumbled upon http://jitsi.org/index.php/GSOC2011/KamailioJingle What is the current status? Has the project advanced?
This is the kind of project that will never succeed. Kamailio is a SIP proxy, not a XMPP server/proxy/whatever. It's unfeasible to expect that a magic new Kamailio module will behave as a perfect B2BUA between SIP and XMPP in both, the signaling an the media plane, 100% unfeasible.
Dne 13.12.2011 v 14:26:50, IĂąaki Baz Castillo ibc@aliax.net napsal:
I never said it would be perfect.. and I never said I would create a new XMPP module.. Adding jingle support to existing one seems way better. Sure, if both clients support only incompatible sets of codecs, then it won't work. Which is normal and happens to me with both Jingle-to-Jingle and SIP-to-SIP from time to time.
MMlosh
2011/12/13 kasip@elektromaniak.wz.cz:
It's not just that. SIP uses a plain/raw SDP format while XMPP uses a XML version of SDP. Just an example of the hard task a SIP-XMPP-proxy should fulfill to enable full interoperability between both words.
Regards.
Dne 13.12.2011 v 16:16:54, IĂąaki Baz Castillo ibc@aliax.net napsal:
Sure.. it would be a bit easier (measured in lines of code) if one could simply forward SDP around. Doing some kind of conversions & keeping it light/fast/reliable is harder than sending what I received.
I thought about putting the feature as a transport on a jabber server, but then it would have to act as a sip router as well, creating way more complications and issues.
Maybe I should try poking around first.. If you give me some more directions / possible issue information, I can find out how hard it is and stop asking.
Thanks
Dne 14.12.2011 v 17:18:08, "Olle E. Johansson" oej@edvina.net napsal:
I don't see the point of trying to bridge a SIP call with a jingle call in a proxy. Maybe you can explain that?
I use openser as tiny local sip server (not the indended proxy usecase). No uplink, no special hardware, no PSTN, not even writable storage. Openser already has jabber module, adding Jingle did not feel that hard.
Using asterisk's functionality would work, if I ran Asterisk. Running asterisk solely for ONE (maybe up to 5 in the future) jabber transport seems like an overkill. That's probably the reason. Not the one that could persuade anyone I guess.