asterisk

Commit Graph

Author	SHA1	Message	Date
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
Russell Bryant	f0f5e237bf	Merged revisions 317474 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317474 \| russell \| 2011-05-05 17:36:33 -0500 (Thu, 05 May 2011) \| 2 lines Fix more "set but unused" warnings. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317475 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	857814f435	Add SRTP support for Asterisk 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	15 years ago

Author

SHA1

Message

Date

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

Russell Bryant

f0f5e237bf

Merged revisions 317474 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r317474 | russell | 2011-05-05 17:36:33 -0500 (Thu, 05 May 2011) | 2 lines
  
  Fix more "set but unused" warnings.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317475 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Terry Wilson

857814f435

Add SRTP support for Asterisk

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

3 Commits (ebaf59a656ef94e19332c08e0d0562b8cbe59f65)