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Gregory Nietsky 8493c46308 Merged revisions 336936 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/10 ........ 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/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336937 65c4cc65-6c06-0410-ace0-fbb531ad65f3		14 years ago
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config_parser.h	Make all of the various rtpqos parameters in this branch available from the CHANNEL function.	16 years ago
dialog.h	Revert early destruction of RTP sessions	15 years ago
dialplan_functions.h	Make all of the various rtpqos parameters in this branch available from the CHANNEL function.	16 years ago
globals.h	Asterisk media architecture conversion - no more format bitfields	15 years ago
reqresp_parser.h	Merged revisions 304245 via svnmerge from	15 years ago
sdp_crypto.h	Merged revisions 336936 via svnmerge from	14 years ago
sip.h	Merged revisions 336936 via svnmerge from	14 years ago
sip_utils.h	Make all of the various rtpqos parameters in this branch available from the CHANNEL function.	16 years ago
srtp.h	Merged revisions 336936 via svnmerge from	14 years ago