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

Asterisk Bug Tracker noreply at bugs.digium.com
Wed Mar 3 10:17:36 CST 2010


A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=5413 
====================================================================== 
Reported By:                mikma
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   5413
Category:                   Channels/chan_sip/NewFeature
Reproducibility:            N/A
Severity:                   feature
Priority:                   normal
Status:                     confirmed
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:              2010-03-03 10:17 CST
====================================================================== 
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 (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
====================================================================== 

---------------------------------------------------------------------- 
 (0118853) twilson (administrator) - 2010-03-03 10:17
 https://issues.asterisk.org/view.php?id=5413#c118853 
---------------------------------------------------------------------- 
oh, I wonder if you either have directmedia=yes or canreinvite=yes... if
you take out the || p->srtp that I mentioned above and instead change
things to this:

if (p->srtp) {
    res = AST_RTP_GLUE_RESULT_FORBID;
} else if (ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA |
SIP_DIRECT_MEDIA_NAT)) {
    res = AST_RTP_GLUE_RESULT_REMOTE;
} else if (ast_test_flag(&global_jbconf, AST_JB_FORCED)) {
    res = AST_RTP_GLUE_RESULT_FORBID;
}

and let me know if it fixes it for you. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-03-03 10:17 twilson        Note Added: 0118853                          
======================================================================




More information about the asterisk-bugs mailing list