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

Asterisk Bug Tracker noreply at bugs.digium.com
Thu Dec 25 13:31:21 CST 2008


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:                otherwiseguy
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Project:                    Asterisk
Issue ID:                   5413
Category:                   Channels/chan_sip/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:             2005-10-09 10:36 CDT
Last Modified:              2008-12-25 13:31 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
======================================================================
Relationships       ID      Summary
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related to          0010129 Module SRTP can't loaded
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---------------------------------------------------------------------- 
 (0096953) notthematrix (reporter) - 2008-12-25 13:31
 http://bugs.digium.com/view.php?id=5413#c96953 
---------------------------------------------------------------------- 
oke i TRYED srtpenable=yes

but it did not enable srtp for some reason.
my ata is set to accept srtp only calls but when tho call was picked up it
refused it because for some reason it did not see srtpenable=yes
maby it can only do peers now not friends?
the solution with srtpenable=yes is good becouse it can be easaly patched
in to a setup system as freepbx or others.

this is my sip config....

sip_additional.conf

[31251788103]
type=friend
secret=xxxxxxx
qualify=yes
port=5060
pickupgroup=
nat=yes
mailbox=31251788103 at device
host=dynamic
dtmfmode=rfc2833
dial=SIP/31251788103
context=klant-31-123-123456
canreinvite=no
callgroup=
callerid=device <31251788103>
accountcode=
call-limit=50

sip_custom_post.conf

[31251788103](+)
transport=tls
srtpenable=yes 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-12-25 13:31 notthematrix   Note Added: 0096953                          
======================================================================




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