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

[Asterisk Bug Tracker]

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:                twilson
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Project:                    Asterisk
Issue ID:                   5413
Category:                   Channels/chan_sip/NewFeature
Reproducibility:            N/A
Severity:                   feature
Priority:                   normal
Status:                     assigned
Target Version:             1.6.3
Asterisk Version:           SVN 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!): 48491 
Request Review:              
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Date Submitted:             2005-10-09 10:36 CDT
Last Modified:              2009-03-04 09:15 CST
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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
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Relationships       ID      Summary
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related to          0010129 Module SRTP can't loaded
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---------------------------------------------------------------------- 
 (0101189) munnymun (reporter) - 2009-03-04 09:15
 http://bugs.digium.com/view.php?id=5413#c101189 
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So...
#svn checkout svn.digium.com/svn/asterisk/team/group/srtp is the only
thing you need to do to make Asterisk works with SRTP? Very cool!

If so, that means there is something wrong with my sip.conf or
extensions.conf then.

my sip.conf

[2000]
type=friend
username=2000
secret=123456
host=dynamic
context=test
transport=tls
[2001]
type=friend
username=2001
secret=123456
host=dynamic
context=test
transport=tls

my extensions.conf

exten => 2000,1,Set(_SIP_SRTP_SDES=1)
exten => 2000,2,Set(_SIPSRTP=enable)
exten => 2000,3,Set(_SIPSRTP_CRYPTO=enable)
exten => 2000,4,Dial(SIP/2000)
exten => 2001,1,Set(_SIP_SRTP_SDES=1)
exten => 2001,2,Set(_SIPSRTP=enable)
exten => 2001,3,Set(_SIPSRTP_CRYPTO=enable)
exten => 2001,4,Dial(SIP/2001)

The two endpoints are Eyebeam 1.5. They are both registered over TLS.
Could you tell what could be wrong with these two conf files that prevents
Asterisk to accept SDP with SDES?

I can provide SIP trace if needed.

Thanks! 

Issue History 
Date Modified    Username       Field                    Change               
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2009-03-04 09:15 munnymun       Note Added: 0101189                          
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