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authorphsultan <phsultan@f38db490-d61c-443f-a65b-d21fe96a405b>2007-09-25 09:07:30 +0000
committerphsultan <phsultan@f38db490-d61c-443f-a65b-d21fe96a405b>2007-09-25 09:07:30 +0000
commit70c0a8fbbfd9d5e1732623890935e657bb3d6fe1 (patch)
tree281d2ca1f4d90762cff70b79d3dbf7e34767f3cd /configs
parentea4c42d1e627b40ad14121338c7d793b492f5210 (diff)
Comply with latest XEP-0166, XEP-0167, XEP-0176.
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: http://svn.digium.com/svn/asterisk/trunk@83743 f38db490-d61c-443f-a65b-d21fe96a405b
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