[asterisk-bugs] [Asterisk 0005413]: [branch] Secure RTP (SRTP)
Asterisk Bug Tracker
noreply at bugs.digium.com
Fri Dec 19 01:38:41 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:
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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: 2008-12-19 01:38 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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(0096696) umberto71 (reporter) - 2008-12-19 01:38
http://bugs.digium.com/view.php?id=5413#c96696
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Many thanks.I just fixed some issues in rtp.c (of JPEELER old branch)
1) void ast_rtp_new_source(struct ast_rtp *rtp)
{
rtp->set_marker_bit = 1;
//rtp->ssrc = ast_random(); <----- (As in 1.6.0.1 and new trunk)
return;
}
2)Change sendto into rtp_sendto...
3)Fixed Add_line in chan_sip.c (sometimes invite packet is cut .. but in
1.6.0.1 there isn't this problem)
I've got a look at http://svn.digium.com/svn/asterisk/team/group/srtp. I
think this issues are fixed. I'will try it next week.
Issue History
Date Modified Username Field Change
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2008-12-19 01:38 umberto71 Note Added: 0096696
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