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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

49
main/rtp_engine.c
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@@ -39,6 +39,9 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/pbx.h"
#include "asterisk/translate.h"
struct ast_srtp_res *res_srtp = NULL;
struct ast_srtp_policy_res *res_srtp_policy = NULL;
/*! Structure that represents an RTP session (instance) */
struct ast_rtp_instance {
/*! Engine that is handling this RTP instance */
@@ -67,6 +70,8 @@ struct ast_rtp_instance {
struct ast_rtp_glue *glue;
/*! Channel associated with the instance */
struct ast_channel *chan;
/*! SRTP info associated with the instance */
struct ast_srtp *srtp;
};
/*! List of RTP engines that are currently registered */
@@ -1670,3 +1675,47 @@ struct ast_channel *ast_rtp_instance_get_chan(struct ast_rtp_instance *instance)
{
return instance->chan;
}
int ast_rtp_engine_register_srtp(struct ast_srtp_res *srtp_res, struct ast_srtp_policy_res *policy_res)
{
if (res_srtp || res_srtp_policy) {
return -1;
}
if (!srtp_res || !policy_res) {
return -1;
}
res_srtp = srtp_res;
res_srtp_policy = policy_res;
return 0;
}
void ast_rtp_engine_unregister_srtp(void)
{
res_srtp = NULL;
res_srtp_policy = NULL;
}
int ast_rtp_engine_srtp_is_registered(void)
{
return res_srtp && res_srtp_policy;
}
int ast_rtp_instance_add_srtp_policy(struct ast_rtp_instance *instance, struct ast_srtp_policy *policy)
{
if (!res_srtp) {
return -1;
}
if (!instance->srtp) {
return res_srtp->create(&instance->srtp, instance, policy);
} else {
return res_srtp->add_stream(instance->srtp, policy);
}
}
struct ast_srtp *ast_rtp_instance_get_srtp(struct ast_rtp_instance *instance)
{
return instance->srtp;
}