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Add SRTP support for Asterisk

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After 5 years in mantis and over a year on reviewboard, SRTP support is finally
being comitted. This includes generic CHANNEL dialplan functions that work for
getting the status of whether a call has secure media or signaling as defined
by the underlying channel technology and for setting whether or not a new
channel being bridged to a calling channel should have secure signaling or
media. See doc/tex/secure-calls.tex for examples.

Original patch by mikma, updated for trunk and revised by me.

(closes issue #5413)
Reported by: mikma
Tested by: twilson, notthematrix, hemanshurpatel

Review: https://reviewboard.asterisk.org/r/191/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268894 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Terry Wilson
2010-06-08 05:29:08 +00:00
parent ebbf166c2d
commit 857814f435
28 changed files with 9227 additions and 30793 deletions
Download Patch File Download Diff File Expand all files Collapse all files

5
include/asterisk/global_datastores.h
View File

@@ -27,10 +27,15 @@
#include "asterisk/channel.h"
extern const struct ast_datastore_info dialed_interface_info;
extern const struct ast_datastore_info secure_call_info;
struct ast_dialed_interface {
AST_LIST_ENTRY(ast_dialed_interface) list;
char interface[1];
};
struct ast_secure_call_store {
unsigned int signaling:1;
unsigned int media:1;
};
#endif