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

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=5413 
====================================================================== 
Reported By:                mikma
Assigned To:                otherwiseguy
====================================================================== 
Project:                    Asterisk
Issue ID:                   5413
Category:                   Channels/chan_sip/NewFeature
Reproducibility:            N/A
Severity:                   feature
Priority:                   normal
Status:                     assigned
Asterisk Version:           SVN 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!): 48491 
Disclaimer on File?:        Yes 
Request Review:              
====================================================================== 
Date Submitted:             2005-10-09 10:36 CDT
Last Modified:              2008-12-22 18:56 CST
====================================================================== 
Summary:                    [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 (17/12/2008): Branch against trunk is located here
http://svn.digium.com/svn/asterisk/team/group/srtp
======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
related to          0010129 Module SRTP can't loaded
====================================================================== 

---------------------------------------------------------------------- 
 (0096869) notthematrix (reporter) - 2008-12-22 18:56
 http://bugs.digium.com/view.php?id=5413#c96869 
---------------------------------------------------------------------- 
dit the followibng in my dialplan

added following lines in my freepbx dialplan
exten => 103,n,Set(_SIP_SRTP_SDES=1)
exten => 103,n,Set(_SIPSRTP=optional)
exten => 103,n,Set(_SIPSRTP_CRYPTO=enable)

it worked if I set the device the ht-503 on forced srtp 
when set to enebled it works too bu when set to disabled it fails 
wondering if there is any $variable in asterisk were you can check if a 
certan divice is registerd with TLS if so enable srtp.
for now I might put it in the dialplan it might also be an option to put
in srtp.c self but thats not my peace of cake.

test result SRTP and TLS asterisk SVN-group-srtp-r166392M with
patch 21060 is funtioning properly. 
wth grandstream HT-503 firmware 10.0.15
hope you guys can implement one of my sugestions to make live easyer :) 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-12-22 18:56 notthematrix   Note Added: 0096869                          
======================================================================