[asterisk-bugs] [Asterisk 0005413]: [patch] 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:                
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Project:                    Asterisk
Issue ID:                   5413
Category:                   Channels/chan_sip/NewFeature
Reproducibility:            N/A
Severity:                   feature
Priority:                   normal
Status:                     confirmed
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:             2005-10-09 10:36 CDT
Last Modified:              2008-11-24 10:57 CST
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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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 (0095380) umberto71 (reporter) - 2008-11-24 10:57
 http://bugs.digium.com/view.php?id=5413#c95380 
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Tested using http://svn.digium.com/svn/asterisk/team/jpeeler/srtp/ . I
builded it on fedora core 7. I build also minisip latest revision before. I
enable SRTP:
sip.conf
[400]
type=peer
host=dynamic
secret=
context=tutorial
mailbox=400 at default
canreinvite=no
nat=no

[401]
type=peer
host=dynamic
secret=
context=tutorial
mailbox=401 at default
canreinvite=no
nat=no
extension.conf
[tutorial]
exten => 400, 1, Set(_SIP_SRTP_SDES=1)
exten => 400, n, Set(_SIPSRTP=require)
exten => 400, n, Set(_SIPSRTP_CRYPTO=enable)
exten => 400, n, Dial(SIP/400)

[tutorial]
exten => 401, 1, Set(_SIP_SRTP_SDES=1)
exten => 401, n, Set(_SIPSRTP=require)
exten => 401, n, Set(_SIPSRTP_CRYPTO=enable)
exten => 401, n, Dial(SIP/401)
Result no Audio: [Nov 24 17:26:20] NOTICE[8990]: rtp.c:4017
ast_rtp_bridge: Cannot native bridge in SRTP.

If i disable SRTP in one phone Audio is Half Duplex. (SNOM 300, AASTRA57,
...)
Does anyone know how I can resolve this? I'm doing something wrong ? 

Issue History 
Date Modified    Username       Field                    Change               
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2008-11-24 10:57 umberto71      Note Added: 0095380                          
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