Terry Wilson
3b4d9075f6
Merged revisions 327682 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r327682 | twilson | 2011-07-11 12:41:59 -0700 (Mon, 11 Jul 2011) | 9 lines
Update chan_gtalk to work with changed GMail-based calls
The messages sent by the GMail client have changed, but include the
old-style messages as well. This patch checks for this case and
uses the old-style offer.
(closes issue ASTERISK-18084)
Review: https://reviewboard.asterisk.org/r/1312/
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2011-07-11 19:49:35 +00:00
David Vossel
268ae2e8d5
Merged revisions 290479 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r290479 | dvossel | 2010-10-05 17:00:43 -0500 (Tue, 05 Oct 2010) | 6 lines
Fixes chan_gtalk to work with gmail client
This patch was written by Philippe Sultan (phsultan). Thanks
for keeping this up to date!
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@290480 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-10-05 22:01:52 +00:00
Philippe Sultan
bf13b4df4e
Changed to temporary namespaces to match with latest XEPs. As soon as
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Jingle is completely standardized, we can set those namespaces to their
final values.
Added two attributes to the jingle_pvt struct to store the content
name attributes. Reported by Robert McQueen on Telepathy's framework
mailing list :
http://lists.freedesktop.org/archives/telepathy/2008-May/001971.html
Keeping working on our Jingle stack!
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2008-05-28 14:10:48 +00:00
Philippe Sultan
3a5f263bf0
Comply with latest XEP-0166, XEP-0167, XEP-0176.
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No real Jingle implementation being available, testing was made using
two Asterisk servers relaying SIP calls over their Jingle channels:
SIP Phone 1 --- [chan_sip]Asterisk 1[chan_jingle] --- [chan_jingle]Asterisk 2[chan_sip] --- SIP Phone 2
Thus, it was possible to test the code in both ways, and make the
Jingle channel comply with the latest specifications. No sound available yet.
Main modifications include :
- modified the 'jingle_candidate' structure and the
'jingle_create_candidates' function according to XEP-0176 ;
- modified the 'jingle_action' function in order to properly terminate
a Jingle session, in conformance with XEP-0166 ;
- modified username format used in STUN requests ;
- actually make the bindaddr configuration field useable.
Todo :
- set audio paths up (no native bridging) ;
- make the CLI gtalk functions available to jingle ;
- clean up the storage space used in strings.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@83743 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-09-25 09:07:30 +00:00
Philippe Sultan
dc9dc75379
Transmit proper invitation, thus conforming to XEP-0166 (Jingle general
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specifications), XEP-0167 (Jingle Audio via RTP) and XEP-0176 (Jingle ICE
Transport).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@83055 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-09-19 12:23:56 +00:00
Philippe Sultan
43d63b1027
Assign namespace properly
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2007-09-13 15:10:08 +00:00
Philippe Sultan
92cc7aeff1
Changed Jingle and Jingle DTMF namespaces.
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As both specifications are in the Experimental status, the namespaces
specified therein shall be of the form
"http://www.xmpp.org/extensions/xep-XXXX.html#ns ".
See the Namespace issuance section in XEP-0053 :
http://www.xmpp.org/extensions/xep-0053.html#namespaces
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2007-09-13 15:05:16 +00:00
Olle Johansson
75d387acbc
Doxygen additions, corrections
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2007-02-24 20:29:41 +00:00
Matt O'Gorman
7aa1a77e75
asterisk-xmpp merge in
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2006-05-22 21:12:30 +00:00
