[asterisk-bugs] [Asterisk 0005413]: [branch] Secure RTP (SRTP)
Asterisk Bug Tracker
noreply at bugs.digium.com
Fri Dec 19 16:59:28 CST 2008
A NOTE has been added to this issue.
======================================================================
http://bugs.digium.com/view.php?id=5413
======================================================================
Reported By: mikma
Assigned To:
======================================================================
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:
======================================================================
Date Submitted: 2005-10-09 10:36 CDT
Last Modified: 2008-12-19 16:59 CST
======================================================================
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
======================================================================
Relationships ID Summary
----------------------------------------------------------------------
related to 0010129 Module SRTP can't loaded
======================================================================
----------------------------------------------------------------------
(0096737) otherwiseguy (administrator) - 2008-12-19 16:59
http://bugs.digium.com/view.php?id=5413#c96737
----------------------------------------------------------------------
Ok, I've gone through and cleaned up most of the code to make it conform to
coding guidelines, make it use the ast_ memory allocation functions, etc.
What I haven't really touched is res/mikey.cc. So here are my questions:
1) Has anyone tested the MIKEY code lately? 2) Do any phones actually
support MIKEY 3) Since asterisk has TCP/TLS support now, do we even want to
support MIKEY?
I ask because the code in res/mikey.cc is full of debugging code that
writes to stdout/stderr, the MIKEY stuff in general depends on a bunch of
libraries that aren't actually packaged by distros from what I can tell,
and I haven't seen any actual phones that support it and therefore haven't
tested it myself. My initial inclination is to just remove the support if
nothing/no one uses it.
Anyone have any thoughts/feelings on the subject?
Issue History
Date Modified Username Field Change
======================================================================
2008-12-19 16:59 otherwiseguy Note Added: 0096737
======================================================================
More information about the asterisk-bugs
mailing list