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

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=5413 
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Reported By:                mikma
Assigned To:                jpeeler
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Project:                    Asterisk
Issue ID:                   5413
Category:                   Core/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:              
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Date Submitted:             10-09-2005 10:36 CDT
Last Modified:              03-24-2008 12:20 CDT
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Summary:                    [patch] 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

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Relationships       ID      Summary
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related to          0010129 Module SRTP can't loaded
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---------------------------------------------------------------------- 
 jpeeler - 03-24-08 12:20  
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No particular versions need to be checked out. I was using trunk from both
projects.

It would be helpful for you to compile with DONT OPTIMIZE turned on and
then post a backtrace from gdb, perhaps even a "thread apply all bt". Which
crypto options are you using in the dialplan (_SIPSRTP_MIKEY or
_SIPSRTP_CRYPTO) ? What phones are you using for testing? 

Issue History 
Date Modified   Username       Field                    Change               
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03-24-08 12:20  jpeeler        Note Added: 0084449                          
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