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authorAlexander Traud <pabstraud@compuserve.com>2016-07-19 16:41:44 +0200
committerJoshua Colp <jcolp@digium.com>2016-09-07 11:45:23 +0000
commit7a12355dbdf7ee49713746d993c653f91b0fcbd0 (patch)
tree2841526be7bb9695f0c267f18e4b49cd3ad72933 /CHANGES
parente6cad17d6dd4e335632ec27478ccee40bb574cc3 (diff)
chan_sip: Allow Preferred sRTP.
Following the Encrypt-all-the-things paradigm: The user enters his SIP-URI and password. Thanks to DNS-NAPTR, the phone determines SIP-over-TLS as preferred transport. In SIP/SDP, the phone starts the call with a crypto attribute, but not as RTP/sAVP but the RTP/AVP profile (sRTP is preferred aka optional; not mandatory). If the VoIP server does not support sRTP and TLS, the phone shows an open padlock icon. This paradigm is supported by several VoIP/SIP clients on default. Some implementations even cannot be changed to RTP/sAVP. Therefore here, this change allows Preferred sRTP for ingress. For egress, please, create a dial plan which starts with RTP/SAVP, and when rejected tries again with RTP/AVP. ASTERISK-20234 #close Reported by: tootai Tested by: tootai, Alexander Traud patches: srtp_patches.diff submitted by Matt Jordan Change-Id: I42cb779df3a9c7b3dd03a629fb3a296aa4ceb0fd
Diffstat (limited to 'CHANGES')
-rw-r--r--CHANGES7
1 files changed, 7 insertions, 0 deletions
diff --git a/CHANGES b/CHANGES
index 5857165c8..8a6c673bc 100644
--- a/CHANGES
+++ b/CHANGES
@@ -12,6 +12,13 @@
--- Functionality changes from Asterisk 14 to Asterisk 15 --------------------
------------------------------------------------------------------------------
+chan_sip
+------------------
+ * If an offer is received with optional SRTP (a media stream with RTP/AVP but
+ which contains a crypto line) chan_sip will now accept it and enable SRTP.
+ If you would like to do optional SRTP on outbound you will need to create
+ a dialplan that dials with it enabled initially and if it fails fall back to
+ without.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 14.0.0 to Asterisk 14.1.0 ----------