[asterisk-bugs] [Asterisk 0005413]: [patch] [branch] Secure RTP (SRTP)

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=5413 
====================================================================== 
Reported By:                mikma
Assigned To:                twilson
====================================================================== 
Project:                    Asterisk
Issue ID:                   5413
Category:                   Channels/chan_sip/NewFeature
Reproducibility:            N/A
Severity:                   feature
Priority:                   normal
Status:                     assigned
Target Version:             1.8
Asterisk Version:           SVN 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!): 48491 
Request Review:              
====================================================================== 
Date Submitted:             2005-10-09 10:36 CDT
Last Modified:              2010-04-29 09:13 CDT
====================================================================== 
Summary:                    [patch] [branch] Secure RTP (SRTP)
Description: 
This patch adds initial support for secure RTP using libsrt[1]. It can
be used in for example an implementation of the sdecriptions draft[2].

[1] http://srtp.sourceforge.net/srtp.html
[2]
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdescriptions-12.txt


Update (16/03/2010): Branch against trunk is located here
http://svn.asterisk.org/svn/asterisk/team/group/srtp_reboot

*** IF TESTING, PLEASE USE THE ABOVE BRANCH AND NOT THE PATCHED ATTACHED
TO THIS ISSUE AS THEY ARE OUT OF DATE ***
======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
related to          0010129 Module SRTP can't loaded
====================================================================== 

---------------------------------------------------------------------- 
 (0121155) twilson (administrator) - 2010-04-29 09:13
 https://issues.asterisk.org/view.php?id=5413#c121155 
---------------------------------------------------------------------- 
save1985: use srtp_reboot. srtp_mikey is there for historical purposes. if
your phone calls in with an SRTP offer, Asterisk will set up the call with
encryption if possible. If you want to check via the dialplan whether a
call has encrypted signaling or media, you can check with
CHANNEL(secure_signaling) and CHANNEL(secure_media). If you want to force
any outgoing channels to have secure signaling and media, you can use
Set(CHANNEL(secure_bridge_signaling)=1) and
Set(CHANNEL(secure_bridge_media)=1). If you want all calls to and from a
particular SIP peer to be encrypted, you can put "encryption=yes" in
sip.conf. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-04-29 09:13 twilson        Note Added: 0121155                          
======================================================================