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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
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57
include/asterisk/res_srtp.h Normal file
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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2010 FIXME
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/*! \file
* \brief SRTP resource
*/
#ifndef _ASTERISK_RES_SRTP_H
#define _ASTERISK_RES_SRTP_H
struct ast_srtp;
struct ast_srtp_policy;
struct ast_rtp_instance;
struct ast_srtp_cb {
int (*no_ctx)(struct ast_rtp_instance *rtp, unsigned long ssrc, void *data);
};
struct ast_srtp_res {
int (*create)(struct ast_srtp **srtp, struct ast_rtp_instance *rtp, struct ast_srtp_policy *policy);
void (*destroy)(struct ast_srtp *srtp);
int (*add_stream)(struct ast_srtp *srtp, struct ast_srtp_policy *policy);
void (*set_cb)(struct ast_srtp *srtp, const struct ast_srtp_cb *cb, void *data);
int (*unprotect)(struct ast_srtp *srtp, void *buf, int *size, int rtcp);
int (*protect)(struct ast_srtp *srtp, void **buf, int *size, int rtcp);
int (*get_random)(unsigned char *key, size_t len);
};
/* Crypto suites */
enum ast_srtp_suite {
AST_AES_CM_128_HMAC_SHA1_80 = 1,
AST_AES_CM_128_HMAC_SHA1_32 = 2,
AST_F8_128_HMAC_SHA1_80 = 3
};
struct ast_srtp_policy_res {
struct ast_srtp_policy *(*alloc)(void);
void (*destroy)(struct ast_srtp_policy *policy);
int (*set_suite)(struct ast_srtp_policy *policy, enum ast_srtp_suite suite);
int (*set_master_key)(struct ast_srtp_policy *policy, const unsigned char *key, size_t key_len, const unsigned char *salt, size_t salt_len);
void (*set_ssrc)(struct ast_srtp_policy *policy, unsigned long ssrc, int inbound);
};
#endif /* _ASTERISK_RES_SRTP_H */