[Asterisk-code-review] chan sip: Allow Preferred sRTP. (asterisk[master])

[Joshua Colp]

Hello Anonymous Coward #1000019,

I'd like you to reexamine a change.  Please visit

    https://gerrit.asterisk.org/3234

to look at the new patch set (#3).

Change subject: chan_sip: Allow Preferred sRTP.
......................................................................

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
---
M CHANGES
M channels/chan_sip.c
2 files changed, 15 insertions(+), 17 deletions(-)


  git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/34/3234/3
-- 
To view, visit https://gerrit.asterisk.org/3234
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Gerrit-MessageType: newpatchset
Gerrit-Change-Id: I42cb779df3a9c7b3dd03a629fb3a296aa4ceb0fd
Gerrit-PatchSet: 3
Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Owner: Alexander Traud <pabstraud at compuserve.com>
Gerrit-Reviewer: Alexander Traud <pabstraud at compuserve.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Walter Doekes <walter+asterisk at wjd.nu>