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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

4
channels/sip/dialplan_functions.c
View File

@@ -214,6 +214,10 @@ int sip_acf_channel_read(struct ast_channel *chan, const char *funcname, char *p
ast_log(LOG_WARNING, "Unrecognized argument '%s' to %s\n", preparse, funcname);
return -1;
}
} else if (!strcasecmp(args.param, "secure_signaling")) {
snprintf(buf, buflen, "%s", p->socket.type == SIP_TRANSPORT_TLS ? "1" : "");
} else if (!strcasecmp(args.param, "secure_media")) {
snprintf(buf, buflen, "%s", p->srtp ? "1" : "");
} else {
res = -1;
}