[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:                otherwiseguy
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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 
Disclaimer on File?:        Yes 
Request Review:              
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Date Submitted:             2005-10-09 10:36 CDT
Last Modified:              2009-01-09 20:58 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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 (0097411) notthematrix (reporter) - 2009-01-09 20:58
 http://bugs.digium.com/view.php?id=5413#c97411 
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To: <sip:*69 at 111.222.111.222:5060>;tag=as17e0f7b2
Call-ID: 2021476396-38553-39 at 192.168.1.108
CSeq: 381 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:*69 at 187.111.222.48:5060;transport=TLS>
Content-Type: application/sdp
Content-Length: 455

v=0
o=root 1485261276 1485261276 IN IP4 187.111.222.48
s=Asterisk PBX
c=IN IP4 187.111.222.48
t=0 0
m=audio 19236 RTP/SAVP 8 0 18 4 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:b0KfjEls63OJY3cYFyEpQHlvDeCWc+ee9ry8BDLW

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Issue History 
Date Modified    Username       Field                    Change               
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2009-01-09 20:58 notthematrix   Note Added: 0097411                          
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