[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 20:33 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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 (0097409) notthematrix (reporter) - 2009-01-09 20:33
 http://bugs.digium.com/view.php?id=5413#c97409 
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This is what my grandstream bixes send
the expect a savp header when talking crypto


<--- SIP read from TLS:98.99.11.22:2049 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS
11.222.222.111:5060;branch=z9hG4bK156c3ff3;rport=5060;received=119.116.0.116
From: "0031201231234"
<sip:0031201231234 at 11.222.222.111:5060>;tag=as1eb4d033
To: <sip:31201234321 at 192.168.1.108:38553;transport=tls>;tag=1091078781
Call-ID: 126c5dd4529a32973a20a05d400e65ae at 11.222.222.111
CSeq: 102 INVITE
Contact: <sip:31201234321 at 192.168.1.108:38553;transport=tls>
Supported: replaces, path, timer
User-Agent: Grandstream HT-503  V1.1B 1.0.0.15*
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER,
UPDATE
Content-Type: application/sdp
Content-Length:   416

v=0
o=31201234321 8000 8000 IN IP4 192.168.1.108
s=SIP Call
c=IN IP4 192.168.1.108
t=0 0
m=audio 37014 RTP/SAVP 0 8 18 4 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32-36,54
a=silenceSupp:off - - - -
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:iXjPIrDcmHEJmHI4YSgquMfU/B+Ce8ObDm+jlxjC
a=fmtp:101 0-16,32-36,54
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:KK/kI8C3HrPeai+KP+TJtdGO6LjtagU6S2kUCe+x
a=crypto:2 AES_CM_128_HMAC_SHA1_32
inline:/8dJ4TaZmFcNdsE8AAEgybbxV56qzFivf6MYk6wH

I changed the pulic adresees and phone numbers used in the header
but the difreces in the IP adresses are real, they differ also here but
are changed to fictive valeus,

It looks like Grandstream is using and expecting SAVP 
Hope this helps 

Issue History 
Date Modified    Username       Field                    Change               
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2009-01-09 20:33 notthematrix   Note Added: 0097409                          
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