[asterisk-bugs] [Asterisk 0005413]: [branch] Secure RTP (SRTP)
Asterisk Bug Tracker
noreply at bugs.digium.com
Fri Jan 9 11:13:57 CST 2009
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 11:13 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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(0097321) otherwiseguy (administrator) - 2009-01-09 11:13
http://bugs.digium.com/view.php?id=5413#c97321
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Actually, come to think of it, this part needs work anyway because the
a=crypto lines are actually specific to the media description, not the
session. And of course asterisk doesn't actually parse things this way at
all:
/* Next, scan through each "a=xxxx:" line, noting each
* specified RTP payload type (with corresponding MIME subtype):
*/
/* XXX This needs to be done per media stream, since it's media stream
specific */
iterator = req->sdp_start;
while ((a = get_sdp_iterate(&iterator, req, "a"))[0] != '\0') {
/me sighs
Issue History
Date Modified Username Field Change
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2009-01-09 11:13 otherwiseguy Note Added: 0097321
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