Add SRTP support for Asterisk

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: http://svn.digium.com/svn/asterisk/trunk@268894 f38db490-d61c-443f-a65b-d21fe96a405b

3 files changed, 145 insertions, 3 deletions

diff --git a/channels/sip/include/sdp_crypto.h b/channels/sip/include/sdp_crypto.h
new file mode 100644
index 000000000..b1c153438
--- /dev/null
+++ b/channels/sip/include/sdp_crypto.h
@@ -0,0 +1,82 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2006 - 2007, Mikael Magnusson
+ *
+ * Mikael Magnusson <mikma@users.sourceforge.net>
+ *
+ * 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 sdp_crypto.h
+ *
+ * \brief SDP Security descriptions
+ *
+ * Specified in RFC 4568
+ *
+ * \author Mikael Magnusson <mikma@users.sourceforge.net>
+ */
+
+#ifndef _SDP_CRYPTO_H
+#define _SDP_CRYPTO_H
+
+#include <asterisk/rtp_engine.h>
+
+struct sdp_crypto;
+
+/*! \brief Initialize an return an sdp_crypto struct
+ *
+ * \details
+ * This function allocates a new sdp_crypto struct and initializes its values
+ *
+ * \retval NULL on failure
+ * \retval a pointer to a  new sdp_crypto structure
+ */
+struct sdp_crypto *sdp_crypto_setup(void);
+
+/*! \brief Destroy a previously allocated sdp_crypto struct */
+void sdp_crypto_destroy(struct sdp_crypto *crypto);
+
+/*! \brief Parse the a=crypto line from SDP and set appropriate values on the
+ * sdp_crypto struct.
+ *
+ * \param p A valid sdp_crypto struct
+ * \param attr the a:crypto line from SDP
+ * \param rtp The rtp instance associated with the SDP being parsed
+ *
+ * \retval 0 success
+ * \retval nonzero failure
+ */
+int sdp_crypto_process(struct sdp_crypto *p, const char *attr, struct ast_rtp_instance *rtp);
+
+
+/*! \brief Generate an SRTP a=crypto offer
+ *
+ * \details
+ * The offer is stored on the sdp_crypto struct in a_crypto
+ *
+ * \param A valid sdp_crypto struct
+ *
+ * \retval 0 success
+ * \retval nonzero failure
+ */
+int sdp_crypto_offer(struct sdp_crypto *p);
+
+
+/*! \brief Return the a_crypto value of the sdp_crypto struct
+ *
+ * \param p An sdp_crypto struct that has had sdp_crypto_offer called
+ *
+ * \retval The value of the a_crypto for p
+ */
+const char *sdp_crypto_attrib(struct sdp_crypto *p);
+
+#endif	/* _SDP_CRYPTO_H */
diff --git a/channels/sip/include/sip.h b/channels/sip/include/sip.h
index 9017d7e6b..8d6d0abcb 100644
--- a/channels/sip/include/sip.h
+++ b/channels/sip/include/sip.h
@@ -307,10 +307,8 @@
 #define SIP_PAGE2_RTAUTOCLEAR			(1 <<  1)    /*!< GP: Should we clean memory from peers after expiry? */
 #define SIP_PAGE2_RPID_UPDATE			(1 <<  2)
 #define SIP_PAGE2_Q850_REASON			(1 <<  3)    /*!< DP: Get/send cause code via Reason header */
-
 #define SIP_PAGE2_SYMMETRICRTP			(1 <<  4)    /*!< GDP: Whether symmetric RTP is enabled or not */
 #define SIP_PAGE2_STATECHANGEQUEUE		(1 <<  5)    /*!< D: Unsent state pending change exists */
-
 #define SIP_PAGE2_CONNECTLINEUPDATE_PEND	(1 <<  6)
 #define SIP_PAGE2_RPID_IMMEDIATE		(1 <<  7)
 #define SIP_PAGE2_RPORT_PRESENT			(1 <<  8)   /*!< Was rport received in the Via header? */
@@ -345,6 +343,7 @@
 #define SIP_PAGE2_UDPTL_DESTINATION		(1 << 25)  /*!< DP: Use source IP of RTP as destination if NAT is enabled */
 #define SIP_PAGE2_VIDEOSUPPORT_ALWAYS		(1 << 26)  /*!< DP: Always set up video, even if endpoints don't support it */
 #define SIP_PAGE2_HAVEPEERCONTEXT	(1 << 27)	/*< Are we associated with a configured peer context? */
+#define SIP_PAGE2_USE_SRTP              (1 << 28)    /*!< DP: Whether we should offer (only)  SRTP */
 
 #define SIP_PAGE2_FLAGS_TO_COPY \
 	(SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_IGNORESDPVERSION | \
@@ -352,7 +351,7 @@
 	SIP_PAGE2_BUGGY_MWI | SIP_PAGE2_TEXTSUPPORT | SIP_PAGE2_FAX_DETECT | \
 	SIP_PAGE2_UDPTL_DESTINATION | SIP_PAGE2_VIDEOSUPPORT_ALWAYS | SIP_PAGE2_PREFERRED_CODEC | \
 	SIP_PAGE2_RPID_IMMEDIATE | SIP_PAGE2_RPID_UPDATE | SIP_PAGE2_SYMMETRICRTP |\
-	SIP_PAGE2_Q850_REASON | SIP_PAGE2_HAVEPEERCONTEXT)
+	SIP_PAGE2_Q850_REASON | SIP_PAGE2_HAVEPEERCONTEXT | SIP_PAGE2_USE_SRTP)
 
 
 #define SIP_PAGE3_SNOM_AOC               (1 << 0)  /*!< DPG: Allow snom aoc messages */
@@ -965,6 +964,7 @@ struct sip_pvt {
 	                                       *   or respect the other endpoint's request for frame sizes (on)
 	                                       *   for incoming calls
 	                                       */
+	unsigned short req_secure_signaling:1;/*!< Whether we are required to have secure signaling or not */
 	char tag[11];                     /*!< Our tag for this session */
 	int timer_t1;                     /*!< SIP timer T1, ms rtt */
 	int timer_b;                      /*!< SIP timer B, ms */
@@ -1048,6 +1048,9 @@ struct sip_pvt {
 	AST_LIST_HEAD_NOLOCK(request_queue, sip_request) request_queue; /*!< Requests that arrived but could not be processed immediately */
 	struct sip_invite_param *options;   /*!< Options for INVITE */
 	struct sip_st_dlg *stimer;          /*!< SIP Session-Timers */
+	struct sip_srtp *srtp;              /*!< Structure to hold Secure RTP session data for audio */
+	struct sip_srtp *vsrtp;             /*!< Structure to hold Secure RTP session data for video */
+	struct sip_srtp *tsrtp;             /*!< Structure to hold Secure RTP session data for text */
 
 	int red;                            /*!< T.140 RTP Redundancy */
 	int hangupcause;                    /*!< Storage of hangupcause copied from our owner before we disconnect from the AST channel (only used at hangup) */
diff --git a/channels/sip/include/srtp.h b/channels/sip/include/srtp.h
new file mode 100644
index 000000000..b7a3fc30b
--- /dev/null
+++ b/channels/sip/include/srtp.h
@@ -0,0 +1,57 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2006 - 2007, Mikael Magnusson
+ *
+ * Mikael Magnusson <mikma@users.sourceforge.net>
+ *
+ * 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 sip_srtp.h
+ *
+ * \brief SIP Secure RTP (SRTP)
+ *
+ * Specified in RFC 3711
+ *
+ * \author Mikael Magnusson <mikma@users.sourceforge.net>
+ */
+
+#ifndef _SIP_SRTP_H
+#define _SIP_SRTP_H
+
+#include "sdp_crypto.h"
+
+/* SRTP flags */
+#define SRTP_ENCR_OPTIONAL	(1 << 1)	/* SRTP encryption optional */
+#define SRTP_CRYPTO_ENABLE	(1 << 2)
+#define SRTP_CRYPTO_OFFER_OK	(1 << 3)
+
+/*! \brief structure for secure RTP audio */
+struct sip_srtp {
+	unsigned int flags;
+	struct sdp_crypto *crypto;
+};
+
+/*!
+ * \brief allocate a sip_srtp structure
+ * \retval a new malloc'd sip_srtp structure on success
+ * \retval NULL on failure
+*/
+struct sip_srtp *sip_srtp_alloc(void);
+
+/*!
+ * \brief free a sip_srtp structure
+ * \param srtp a sip_srtp structure
+*/
+void sip_srtp_destroy(struct sip_srtp *srtp);
+
+#endif	/* _SIP_SRTP_H */