[asterisk-bugs] [Asterisk 0005413]: [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:                     feedback
Target Version:             1.6.3.0
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:              2009-10-14 17:20 CDT
====================================================================== 
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
====================================================================== 

---------------------------------------------------------------------- 
 (0112308) st (reporter) - 2009-10-14 17:20
 https://issues.asterisk.org/view.php?id=5413#c112308 
---------------------------------------------------------------------- 
Currently I habe two problems with my Snom 360 and this srtp branch

Snom 360 (7.3.26) tries to make a call to Asterisk
(SVN-group-srtp-r183146-/trunk)

Dialplan application answer fails:

  DEBUG[9036] rtp.c: Got RTCP report of 68 bytes

  DEBUG[9036] res_srtp.c: SRTP unprotect: authentication failure

  WARNING[9036] rtp.c: RTP Read error: Success.  Hanging up.

  DEBUG[9036] channel.c: Hangup of channel SIP/snom2-08caa980 detected in
answer routine


Second problem is, that I don't get much debugging information. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-10-14 17:20 st             Note Added: 0112308                          
======================================================================