Gregory Nietsky 8493c46308 Merged revisions 336936 via svnmerge from ...

https://origsvn.digium.com/svn/asterisk/branches/10

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  r336936 | irroot | 2011-09-20 18:51:59 +0200 (Tue, 20 Sep 2011) | 14 lines
  
  
  Allow Setting Auth Tag Bit length Based on invite or config option
  
  Update the SIP SRTP API to allow use of 32 or 80 bit taglen.
  Curently only 80 bit is supported.
  
  The outgoing invite will use the taglen of the incoming invite preventing
  one-way audio.
  
  (Closes issue ASTERISK-17895)
  
  Review: https://reviewboard.asterisk.org/r/1173/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336937 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2011-09-20 16:56:11 +00:00

..

config_parser.h

Make all of the various rtpqos parameters in this branch available from the CHANNEL function.

2010-02-17 06:25:15 +00:00

dialog.h

Revert early destruction of RTP sessions

2010-07-13 21:42:42 +00:00

dialplan_functions.h

Make all of the various rtpqos parameters in this branch available from the CHANNEL function.

2010-02-17 06:25:15 +00:00

globals.h

Asterisk media architecture conversion - no more format bitfields

2011-02-03 16:22:10 +00:00

reqresp_parser.h

Merged revisions 304245 via svnmerge from

2011-01-26 20:44:47 +00:00

sdp_crypto.h

Merged revisions 336936 via svnmerge from

2011-09-20 16:56:11 +00:00

sip_utils.h

Make all of the various rtpqos parameters in this branch available from the CHANNEL function.

2010-02-17 06:25:15 +00:00

sip.h

Merged revisions 336936 via svnmerge from

2011-09-20 16:56:11 +00:00

srtp.h

Merged revisions 336936 via svnmerge from

2011-09-20 16:56:11 +00:00