summaryrefslogtreecommitdiff
path: root/res/res_srtp.c
AgeCommit message (Collapse)Author
2010-09-01Merged revisions 284477 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284477 | twilson | 2010-09-01 13:44:36 -0500 (Wed, 01 Sep 2010) | 17 lines Fix SRTP for changing SSRC and multiple a=crypto SDP lines Adding code to Asterisk that changed the SSRC during bridges and masquerades broke SRTP functionality. Also broken was handling the situation where an incoming INVITE had more than one crypto offer. This patch caches the SRTP policies the we use so that we can change the ssrc and inform libsrtp of the new streams. It also uses the first acceptable a=crypto line from the incoming INVITE. (closes issue #17563) Reported by: Alexcr Patches: srtp.diff uploaded by twilson (license 396) Tested by: twilson Review: https://reviewboard.asterisk.org/r/878/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284479 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-20Add load priority order, such that preload becomes unnecessary in most casesTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278132 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-08Add SRTP support for AsteriskTerry Wilson
After 5 years in mantis and over a year on reviewboard, SRTP support is finally being comitted. This includes generic CHANNEL dialplan functions that work for getting the status of whether a call has secure media or signaling as defined by the underlying channel technology and for setting whether or not a new channel being bridged to a calling channel should have secure signaling or media. See doc/tex/secure-calls.tex for examples. Original patch by mikma, updated for trunk and revised by me. (closes issue #5413) Reported by: mikma Tested by: twilson, notthematrix, hemanshurpatel Review: https://reviewboard.asterisk.org/r/191/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268894 65c4cc65-6c06-0410-ace0-fbb531ad65f3