summaryrefslogtreecommitdiff
path: root/CHANGES
diff options
context:
space:
mode:
authorTerry Wilson <twilson@digium.com>2010-06-08 05:29:08 +0000
committerTerry Wilson <twilson@digium.com>2010-06-08 05:29:08 +0000
commit857814f4354fb26255d4d5db6e06e90749e9bad0 (patch)
treeecc27fc0db142ea1cd335a74cd1265f993fecd11 /CHANGES
parentebbf166c2d15fd233ee307e760b2a88c46d19f6b (diff)
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
Diffstat (limited to 'CHANGES')
-rw-r--r--CHANGES13
1 files changed, 13 insertions, 0 deletions
diff --git a/CHANGES b/CHANGES
index cffaec13e..3aac8c698 100644
--- a/CHANGES
+++ b/CHANGES
@@ -59,6 +59,10 @@ SIP Changes
* When dialing SIP peers, a new component may be added to the end of the dialstring
to indicate that a specific remote IP address or host should be used when dialing
the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
+ * SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
+ ability to selectively force bridged channels to also be encrypted is also
+ implemented. Branching in the dialplan can be done based on whether or not
+ a channel has secure media and/or signaling.
* Added directmediapermit/directmediadeny to limit which peers can send direct media
to each other
* Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
@@ -68,6 +72,10 @@ IAX2 Changes
-----------
* Added rtsavesysname option into iax.conf to allow the systname to be saved
on realtime updates.
+ * Added the ability for chan_iax2 to inform the dialplan whether or not
+ encryption is being used. This interoperates with the SIP SRTP implementation
+ so that a secure SIP call can be bridged to a secure IAX call when the
+ dialplan requires bridged channels to be "secure".
MGCP Changes
------------
@@ -205,6 +213,11 @@ Dialplan Functions
prefixing the name of the hash at assignment with the appropriate number of
underscores, just like variables.
* GROUP_MATCH_COUNT has been improved to allow regex matching on category
+ * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
+ whether or not channels that are bridged to the current channel will be
+ required to have secure signaling and/or media.
+ * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
+ the current channel has secure signaling and/or media.
* For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
"no_media_path" option.
Returns "0" if there is a B channel associated with the call.