[asterisk-bugs] [Asterisk 0005413]: [patch] [branch] Secure RTP (SRTP)
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed Mar 3 10:17:36 CST 2010
A NOTE has been added to this issue.
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https://issues.asterisk.org/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
Target Version: 1.8
Asterisk Version: SVN
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!): 48491
Request Review:
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Date Submitted: 2005-10-09 10:36 CDT
Last Modified: 2010-03-03 10:17 CST
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Summary: [patch] [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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(0118853) twilson (administrator) - 2010-03-03 10:17
https://issues.asterisk.org/view.php?id=5413#c118853
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oh, I wonder if you either have directmedia=yes or canreinvite=yes... if
you take out the || p->srtp that I mentioned above and instead change
things to this:
if (p->srtp) {
res = AST_RTP_GLUE_RESULT_FORBID;
} else if (ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA |
SIP_DIRECT_MEDIA_NAT)) {
res = AST_RTP_GLUE_RESULT_REMOTE;
} else if (ast_test_flag(&global_jbconf, AST_JB_FORCED)) {
res = AST_RTP_GLUE_RESULT_FORBID;
}
and let me know if it fixes it for you.
Issue History
Date Modified Username Field Change
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2010-03-03 10:17 twilson Note Added: 0118853
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