Files
asterisk/channels
Philippe Sultan 3a5f263bf0 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: https://origsvn.digium.com/svn/asterisk/trunk@83743 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-09-25 09:07:30 +00:00
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