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

8 files changed, 781 insertions, 18 deletions

diff --git a/channels/chan_iax2.c b/channels/chan_iax2.c
index 04ce7ee7b..591e2d68a 100644
--- a/channels/chan_iax2.c
+++ b/channels/chan_iax2.c
@@ -1171,6 +1171,7 @@ static int iax2_sendhtml(struct ast_channel *c, int subclass, const char *data,
 static int iax2_sendimage(struct ast_channel *c, struct ast_frame *img);
 static int iax2_sendtext(struct ast_channel *c, const char *text);
 static int iax2_setoption(struct ast_channel *c, int option, void *data, int datalen);
+static int iax2_queryoption(struct ast_channel *c, int option, void *data, int *datalen);
 static int iax2_transfer(struct ast_channel *c, const char *dest);
 static int iax2_write(struct ast_channel *c, struct ast_frame *f);
 static int send_trunk(struct iax2_trunk_peer *tpeer, struct timeval *now);
@@ -1218,6 +1219,7 @@ static const struct ast_channel_tech iax2_tech = {
 	.write_video = iax2_write,
 	.indicate = iax2_indicate,
 	.setoption = iax2_setoption,
+	.queryoption = iax2_queryoption,
 	.bridge = iax2_bridge,
 	.transfer = iax2_transfer,
 	.fixup = iax2_fixup,
@@ -4903,6 +4905,11 @@ static int iax2_call(struct ast_channel *c, char *dest, int timeout)
 		ast_log(LOG_WARNING, "No address associated with '%s'\n", pds.peer);
 		return -1;
 	}
+	if (ast_test_flag64(iaxs[callno], IAX_FORCE_ENCRYPT) && !cai.encmethods) {
+		ast_log(LOG_WARNING, "Encryption forced for call, but not enabled\n");
+		c->hangupcause = AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
+		return -1;
+	}
 	if (ast_strlen_zero(cai.secret) && ast_test_flag64(iaxs[callno], IAX_FORCE_ENCRYPT)) {
 		ast_log(LOG_WARNING, "Call terminated. No secret given and force encrypt enabled\n");
 		return -1;
@@ -5144,6 +5151,19 @@ static int iax2_setoption(struct ast_channel *c, int option, void *data, int dat
 	case AST_OPTION_OPRMODE:
 		errno = EINVAL;
 		return -1;
+	case AST_OPTION_SECURE_SIGNALING:
+	case AST_OPTION_SECURE_MEDIA:
+	{
+		unsigned short callno = PTR_TO_CALLNO(c->tech_pvt);
+		ast_mutex_lock(&iaxsl[callno]);
+		if ((*(int *) data)) {
+			ast_set_flag64(iaxs[callno], IAX_FORCE_ENCRYPT);
+		} else {
+			ast_clear_flag64(iaxs[callno], IAX_FORCE_ENCRYPT);
+		}
+		ast_mutex_unlock(&iaxsl[callno]);
+		return 0;
+	}
 	default:
 	{
 		unsigned short callno = PTR_TO_CALLNO(c->tech_pvt);
@@ -5175,6 +5195,23 @@ static int iax2_setoption(struct ast_channel *c, int option, void *data, int dat
 	}
 }
 
+static int iax2_queryoption(struct ast_channel *c, int option, void *data, int *datalen)
+{
+	switch (option) {
+	case AST_OPTION_SECURE_SIGNALING:
+	case AST_OPTION_SECURE_MEDIA:
+	{
+		unsigned short callno = PTR_TO_CALLNO(c->tech_pvt);
+		ast_mutex_lock(&iaxsl[callno]);
+		*((int *) data) = ast_test_flag64(iaxs[callno], IAX_FORCE_ENCRYPT) ? 1 : 0;
+		ast_mutex_unlock(&iaxsl[callno]);
+		return 0;
+	}
+	default:
+		return -1;
+	}
+}
+
 static struct ast_frame *iax2_read(struct ast_channel *c) 
 {
 	ast_log(LOG_NOTICE, "I should never be called!\n");
@@ -13609,6 +13646,8 @@ static int acf_channel_read(struct ast_channel *chan, const char *funcname, char
 		ast_copy_string(buf, pvt->addr.sin_addr.s_addr ? ast_inet_ntoa(pvt->addr.sin_addr) : "", buflen);
 	} else if (!strcasecmp(args, "peername")) {
 		ast_copy_string(buf, pvt->username, buflen);
+	} else if (!strcasecmp(args, "secure_signaling") || !strcasecmp(args, "secure_media")) {
+		snprintf(buf, buflen, "%s", IAX_CALLENCRYPTED(pvt) ? "1" : "");
 	} else {
 		res = -1;
 	}
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index a1bfcfe7e..b79f35a1a 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -267,11 +267,14 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "sip/include/config_parser.h"
 #include "sip/include/reqresp_parser.h"
 #include "sip/include/sip_utils.h"
+#include "sip/include/srtp.h"
+#include "sip/include/sdp_crypto.h"
 #include "asterisk/ccss.h"
 #include "asterisk/xml.h"
 #include "sip/include/dialog.h"
 #include "sip/include/dialplan_functions.h"
 
+
 /*** DOCUMENTATION
 	<application name="SIPDtmfMode" language="en_US">
 		<synopsis>
@@ -1566,6 +1569,10 @@ static void handle_response_subscribe(struct sip_pvt *p, int resp, const char *r
 static int handle_response_register(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, int seqno);
 static void handle_response(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, int seqno);
 
+/*------ SRTP Support -------- */
+static int setup_srtp(struct sip_srtp **srtp);
+static int process_crypto(struct sip_pvt *p, struct ast_rtp_instance *rtp, struct sip_srtp **srtp, const char *a);
+
 /*------ T38 Support --------- */
 static int transmit_response_with_t38_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, int retrans);
 static struct ast_udptl *sip_get_udptl_peer(struct ast_channel *chan);
@@ -3910,7 +3917,16 @@ static int sip_setoption(struct ast_channel *chan, int option, void *data, int d
 			res = 0;
 		}
 		break;
+	case AST_OPTION_SECURE_SIGNALING:
+		p->req_secure_signaling = *(unsigned int *) data;
+		res = 0;
+		break;
+	case AST_OPTION_SECURE_MEDIA:
+		ast_set2_flag(&p->flags[1], *(unsigned int *) data, SIP_PAGE2_USE_SRTP);
+		res = 0;
+		break;
 	default:
+		ast_log(LOG_NOTICE, "Unknown option: %d\n", option);
 		break;
 	}
 
@@ -3961,6 +3977,14 @@ static int sip_queryoption(struct ast_channel *chan, int option, void *data, int
 		*cp = p->dsp ? 1 : 0;
 		ast_debug(1, "Reporting digit detection %sabled on %s\n", *cp ? "en" : "dis", chan->name);
 		break;
+	case AST_OPTION_SECURE_SIGNALING:
+		*((unsigned int *) data) = p->req_secure_signaling;
+		res = 0;
+		break;
+	case AST_OPTION_SECURE_MEDIA:
+		*((unsigned int *) data) = ast_test_flag(&p->flags[1], SIP_PAGE2_USE_SRTP) ? 1 : 0;
+		res = 0;
+		break;
 	case AST_OPTION_DEVICE_NAME:
 		if (p && p->outgoing_call) {
 			cp = (char *) data;
@@ -5004,6 +5028,35 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout)
 		}
 	}
 
+	/* Check to see if we should try to force encryption */
+	if (p->req_secure_signaling && p->socket.type != SIP_TRANSPORT_TLS) {
+	   ast_log(LOG_WARNING, "Encrypted signaling is required\n");
+	   ast->hangupcause = AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
+	   return -1;
+	}
+
+	if (ast_test_flag(&p->flags[1], SIP_PAGE2_USE_SRTP)) {
+		if (ast_test_flag(&p->flags[0], SIP_REINVITE)) {
+			ast_debug(1, "Direct media not possible when using SRTP, ignoring canreinvite setting\n");
+			ast_clear_flag(&p->flags[0], SIP_REINVITE);
+		}
+
+		if (p->rtp && !p->srtp && setup_srtp(&p->srtp) < 0) {
+			ast_log(LOG_WARNING, "SRTP audio setup failed\n");
+			return -1;
+		}
+
+		if (p->vrtp && !p->vsrtp && setup_srtp(&p->vsrtp) < 0) {
+			ast_log(LOG_WARNING, "SRTP video setup failed\n");
+			return -1;
+		}
+
+		if (p->trtp && !p->vsrtp && setup_srtp(&p->tsrtp) < 0) {
+			ast_log(LOG_WARNING, "SRTP text setup failed\n");
+			return -1;
+		}
+	}
+
 	res = 0;
 	ast_set_flag(&p->flags[0], SIP_OUTGOING);
 
@@ -5211,6 +5264,21 @@ void __sip_destroy(struct sip_pvt *p, int lockowner, int lockdialoglist)
 		p->chanvars = NULL;
 	}
 
+	if (p->srtp) {
+		sip_srtp_destroy(p->srtp);
+		p->srtp = NULL;
+	}
+
+	if (p->vsrtp) {
+		sip_srtp_destroy(p->vsrtp);
+		p->vsrtp = NULL;
+	}
+
+	if (p->tsrtp) {
+		sip_srtp_destroy(p->tsrtp);
+		p->tsrtp = NULL;
+	}
+
 	if (p->directmediaha) {
 		ast_free_ha(p->directmediaha);
 		p->directmediaha = NULL;
@@ -7671,6 +7739,10 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 	const char *codecs;
 	int codec;
 
+	/* SRTP */
+	int secure_audio = FALSE;
+	int secure_video = FALSE;
+
 	/* Others */
 	int sendonly = -1;
 	int vsendonly = -1;
@@ -7770,6 +7842,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 		int video = FALSE;
 		int image = FALSE;
 		int text = FALSE;
+		char protocol[5] = {0,};
 		int x;
 
 		numberofports = 1;
@@ -7780,8 +7853,14 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 		nextm = get_sdp_iterate(&next, req, "m");
 
 		/* Search for audio media definition */
-		if ((sscanf(m, "audio %30u/%30u RTP/AVP %n", &x, &numberofports, &len) == 2 && len > 0) ||
-		    (sscanf(m, "audio %30u RTP/AVP %n", &x, &len) == 1 && len > 0)) {
+		if ((sscanf(m, "audio %30u/%30u RTP/%4s %n", &x, &numberofports, protocol, &len) == 3 && len > 0) ||
+		    (sscanf(m, "audio %30u RTP/%4s %n", &x, protocol, &len) == 2 && len > 0)) {
+			if (!strcmp(protocol, "SAVP")) {
+				secure_audio = 1;
+			} else if (strcmp(protocol, "AVP")) {
+				ast_log(LOG_WARNING, "unknown SDP media protocol in offer: %s\n", protocol);
+				continue;
+			}
 			audio = TRUE;
 			p->offered_media[SDP_AUDIO].offered = TRUE;
 			numberofmediastreams++;
@@ -7801,8 +7880,14 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 				ast_rtp_codecs_payloads_set_m_type(&newaudiortp, NULL, codec);
 			}
 		/* Search for video media definition */
-		} else if ((sscanf(m, "video %30u/%30u RTP/AVP %n", &x, &numberofports, &len) == 2 && len > 0) ||
-			   (sscanf(m, "video %30u RTP/AVP %n", &x, &len) == 1 && len >= 0)) {
+		} else if ((sscanf(m, "video %30u/%30u RTP/%4s %n", &x, &numberofports, protocol, &len) == 3 && len > 0) ||
+			   (sscanf(m, "video %30u RTP/%4s %n", &x, protocol, &len) == 2 && len >= 0)) {
+			if (!strcmp(protocol, "SAVP")) {
+				secure_video = 1;
+			} else if (strcmp(protocol, "AVP")) {
+				ast_log(LOG_WARNING, "unknown SDP media protocol in offer: %s\n", protocol);
+				continue;
+			}
 			video = TRUE;
 			p->novideo = FALSE;
 			p->offered_media[SDP_VIDEO].offered = TRUE;
@@ -7864,8 +7949,6 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 		if (numberofports > 1)
 			ast_log(LOG_WARNING, "SDP offered %d ports for media, not supported by Asterisk. Will try anyway...\n", numberofports);
 		
-
-
 		/* Media stream specific parameters */
 		while ((type = get_sdp_line(&iterator, next - 1, req, &value)) != '\0') {
 			int processed = FALSE;
@@ -7899,6 +7982,8 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 				if (audio) {
 					if (process_sdp_a_sendonly(value, &sendonly))
 						processed = TRUE;
+					else if (process_crypto(p, p->rtp, &p->srtp, value))
+						processed = TRUE;
 					else if (process_sdp_a_audio(value, p, &newaudiortp, &last_rtpmap_codec))
 						processed = TRUE;
 				}
@@ -7906,6 +7991,8 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 				else if (video) {
 					if (process_sdp_a_sendonly(value, &vsendonly))
 						processed = TRUE;
+					else if (process_crypto(p, p->vrtp, &p->vsrtp, value))
+						processed = TRUE;
 					else if (process_sdp_a_video(value, p, &newvideortp, &last_rtpmap_codec))
 						processed = TRUE;
 				}
@@ -7913,6 +8000,8 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 				else if (text) {
 					if (process_sdp_a_text(value, p, &newtextrtp, red_fmtp, &red_num_gen, red_data_pt, &last_rtpmap_codec))
 						processed = TRUE;
+					else if (process_crypto(p, p->trtp, &p->tsrtp, value))
+						processed = TRUE;
 				}
 				/* Image (T.38 FAX) specific scanning */
 				else if (image) {
@@ -7937,20 +8026,48 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 		return -1;
 	}
 
-	if (portno == -1 && vportno == -1 && udptlportno == -1  && tportno == -1)
+	if (portno == -1 && vportno == -1 && udptlportno == -1  && tportno == -1) {
 		/* No acceptable offer found in SDP  - we have no ports */
 		/* Do not change RTP or VRTP if this is a re-invite */
+		ast_log(LOG_WARNING, "Failing due to no acceptable offer found\n");
 		return -2;
+	}
 
-	if (numberofmediastreams > 3)
+	if (numberofmediastreams > 3) {
 		/* We have too many fax, audio and/or video and/or text media streams, fail this offer */
+		ast_log(LOG_WARNING, "Faling due to too many media streams\n");
 		return -3;
+	}
+
+	if (secure_audio && !(p->srtp && (ast_test_flag(p->srtp, SRTP_CRYPTO_OFFER_OK)))) {
+		ast_log(LOG_WARNING, "Can't provide secure audio requested in SDP offer\n");
+		return -4;
+	}
+
+	if (!secure_audio && p->srtp) {
+		ast_log(LOG_WARNING, "We are requesting SRTP, but they responded without it!\n");
+		return -4;
+	}
+
+	if (secure_video && !(p->vsrtp && (ast_test_flag(p->vsrtp, SRTP_CRYPTO_OFFER_OK)))) {
+		ast_log(LOG_WARNING, "Can't provide secure video requested in SDP offer\n");
+		return -4;
+	}
+
+	if (!p->novideo && !secure_video && p->vsrtp) {
+		ast_log(LOG_WARNING, "We are requesting SRTP, but they responded without it!\n");
+		return -4;
+	}
+
+	if (!(secure_audio || secure_video) && ast_test_flag(&p->flags[1], SIP_PAGE2_USE_SRTP)) {
+		ast_log(LOG_WARNING, "Matched device setup to use SRTP, but request was not!\n");
+		return -4;
+	}
 
 	if (udptlportno == -1) {
 		change_t38_state(p, T38_DISABLED);
 	}
 
-
 	/* Now gather all of the codecs that we are asked for: */
 	ast_rtp_codecs_payload_formats(&newaudiortp, &peercapability, &peernoncodeccapability);
 	ast_rtp_codecs_payload_formats(&newvideortp, &vpeercapability, &vpeernoncodeccapability);
@@ -9843,6 +9960,23 @@ static void get_our_media_address(struct sip_pvt *p, int needvideo, int needtext
 	}
 }
 
+static void get_crypto_attrib(struct sip_srtp *srtp, const char **a_crypto)
+{
+	/* Set encryption properties */
+	if (srtp) {
+		if (!srtp->crypto) {
+			srtp->crypto = sdp_crypto_setup();
+		}
+		if (srtp->crypto && (sdp_crypto_offer(srtp->crypto) >= 0)) {
+			*a_crypto = sdp_crypto_attrib(srtp->crypto);
+		}
+
+		if (!*a_crypto) {
+			ast_log(LOG_WARNING, "No SRTP key management enabled\n");
+		}
+	}
+}
+
 /*! \brief Add Session Description Protocol message
 
     If oldsdp is TRUE, then the SDP version number is not incremented. This mechanism
@@ -9879,6 +10013,9 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
 	struct ast_str *a_video = ast_str_alloca(1024); /* Attributes for video */
 	struct ast_str *a_text = ast_str_alloca(1024);  /* Attributes for text */
 	struct ast_str *a_modem = ast_str_alloca(1024); /* Attributes for modem */
+	const char *a_crypto = NULL;
+	const char *v_a_crypto = NULL;
+	const char *t_a_crypto = NULL;
 
 	format_t x;
 	format_t capability = 0;
@@ -9962,7 +10099,9 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
 		/* Ok, we need video. Let's add what we need for video and set codecs.
 		   Video is handled differently than audio since we can not transcode. */
 		if (needvideo) {
-			ast_str_append(&m_video, 0, "m=video %d RTP/AVP", ntohs(vdest.sin_port));
+			get_crypto_attrib(p->vsrtp, &v_a_crypto);
+			ast_str_append(&m_video, 0, "m=video %d RTP/%s", ntohs(vdest.sin_port),
+				v_a_crypto ? "SAVP" : "AVP");
 
 			/* Build max bitrate string */
 			if (p->maxcallbitrate)
@@ -9976,7 +10115,9 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
 		if (needtext) {
 			if (sipdebug_text)
 				ast_verbose("Lets set up the text sdp\n");
-			ast_str_append(&m_text, 0, "m=text %d RTP/AVP", ntohs(tdest.sin_port));
+			get_crypto_attrib(p->tsrtp, &t_a_crypto);
+			ast_str_append(&m_text, 0, "m=text %d RTP/%s", ntohs(tdest.sin_port),
+				t_a_crypto ? "SAVP" : "AVP");
 			if (debug) /* XXX should I use tdest below ? */
 				ast_verbose("Text is at %s port %d\n", ast_inet_ntoa(p->ourip.sin_addr), ntohs(tsin.sin_port));	
 
@@ -9987,7 +10128,9 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
 		/* We break with the "recommendation" and send our IP, in order that our
 		   peer doesn't have to ast_gethostbyname() us */
 
-		ast_str_append(&m_audio, 0, "m=audio %d RTP/AVP", ntohs(dest.sin_port));
+		get_crypto_attrib(p->srtp, &a_crypto);
+		ast_str_append(&m_audio, 0, "m=audio %d RTP/%s", ntohs(dest.sin_port),
+			a_crypto ? "SAVP" : "AVP");
 
 		if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD) == SIP_PAGE2_CALL_ONHOLD_ONEDIR)
 			hold = "a=recvonly\r\n";
@@ -10149,7 +10292,15 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
  		len += m_text->used + a_text->used + strlen(hold);
 	if (add_t38)
 		len += m_modem->used + a_modem->used;
-
+	if (a_crypto) {
+		len += strlen(a_crypto);
+	}
+	if (v_a_crypto) {
+		len += strlen(v_a_crypto);
+	}
+	if (t_a_crypto) {
+		len += strlen(t_a_crypto);
+	}
 	add_header(resp, "Content-Type", "application/sdp");
 	add_header_contentLength(resp, len);
 	add_line(resp, version);
@@ -10163,6 +10314,9 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
 		add_line(resp, m_audio->str);
 		add_line(resp, a_audio->str);
 		add_line(resp, hold);
+		if (a_crypto) {
+			add_line(resp, a_crypto);
+		}
 	} else if (p->offered_media[SDP_AUDIO].offered) {
 		snprintf(dummy_answer, sizeof(dummy_answer), "m=audio 0 RTP/AVP %s\r\n", p->offered_media[SDP_AUDIO].codecs);
 		add_line(resp, dummy_answer);
@@ -10171,6 +10325,9 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
 		add_line(resp, m_video->str);
 		add_line(resp, a_video->str);
 		add_line(resp, hold);	/* Repeat hold for the video stream */
+		if (v_a_crypto) {
+			add_line(resp, v_a_crypto);
+		}
 	} else if (p->offered_media[SDP_VIDEO].offered) {
 		snprintf(dummy_answer, sizeof(dummy_answer), "m=video 0 RTP/AVP %s\r\n", p->offered_media[SDP_VIDEO].codecs);
 		add_line(resp, dummy_answer);
@@ -10179,6 +10336,9 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
 		add_line(resp, m_text->str);
 		add_line(resp, a_text->str);
 		add_line(resp, hold);	/* Repeat hold for the text stream */
+		if (t_a_crypto) {
+			add_line(resp, t_a_crypto);
+		}
 	} else if (p->offered_media[SDP_TEXT].offered) {
 		snprintf(dummy_answer, sizeof(dummy_answer), "m=text 0 RTP/AVP %s\r\n", p->offered_media[SDP_TEXT].codecs);
 		add_line(resp, dummy_answer);
@@ -17482,6 +17642,8 @@ static int function_sippeer(struct ast_channel *chan, const char *cmd, char *dat
 		ast_copy_string(buf, peer->cid_num, len);
 	} else  if (!strcasecmp(colname, "codecs")) {
 		ast_getformatname_multiple(buf, len -1, peer->capability);
+	} else if (!strcasecmp(colname, "encryption")) {
+		snprintf(buf, len, "%d", ast_test_flag(&peer->flags[1], SIP_PAGE2_USE_SRTP));
 	} else  if (!strncasecmp(colname, "chanvar[", 8)) {
 		char *chanvar=colname + 8;
 		struct ast_variable *v;
@@ -21015,8 +21177,13 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
 				transmit_response_with_t38_sdp(p, "200 OK", req, (reinvite ? XMIT_RELIABLE : (req->ignore ?  XMIT_UNRELIABLE : XMIT_CRITICAL)));
 			} else if (p->t38.state == T38_DISABLED) {
 				/* If this is not a re-invite or something to ignore - it's critical */
-				ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
-				transmit_response_with_sdp(p, "200 OK", req, (reinvite ? XMIT_RELIABLE : (req->ignore ?  XMIT_UNRELIABLE : XMIT_CRITICAL)), p->session_modify == TRUE ? FALSE : TRUE, FALSE);
+				if (p->srtp && !ast_test_flag(p->srtp, SRTP_CRYPTO_OFFER_OK)) {
+					ast_log(LOG_WARNING, "Target does not support required crypto\n");
+					transmit_response_reliable(p, "488 Not Acceptable Here (crypto)", req);
+				} else {
+					ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
+					transmit_response_with_sdp(p, "200 OK", req, (reinvite ? XMIT_RELIABLE : (req->ignore ?  XMIT_UNRELIABLE : XMIT_CRITICAL)), p->session_modify == TRUE ? FALSE : TRUE, FALSE);
+				}
 			}
 
 			p->invitestate = INV_TERMINATED;
@@ -25374,6 +25541,8 @@ static struct sip_peer *build_peer(const char *name, struct ast_variable *v, str
 				ast_string_field_set(peer, unsolicited_mailbox, v->value);
 			} else if (!strcasecmp(v->name, "use_q850_reason")) {
 				ast_set2_flag(&peer->flags[1], ast_true(v->value), SIP_PAGE2_Q850_REASON);
+			} else if (!strcasecmp(v->name, "encryption")) {
+				ast_set2_flag(&peer->flags[1], ast_true(v->value), SIP_PAGE2_USE_SRTP);
 			} else if (!strcasecmp(v->name, "snom_aoc_enabled")) {
 				ast_set2_flag(&peer->flags[2], ast_true(v->value), SIP_PAGE3_SNOM_AOC);
 			}
@@ -26666,6 +26835,10 @@ static enum ast_rtp_glue_result sip_get_rtp_peer(struct ast_channel *chan, struc
 		res = AST_RTP_GLUE_RESULT_FORBID;
 	}
 
+	if (p->srtp) {
+		res = AST_RTP_GLUE_RESULT_FORBID;
+	}
+
 	sip_pvt_unlock(p);
 
 	return res;
@@ -27084,6 +27257,50 @@ static void sip_send_all_mwi_subscriptions(void)
 	} while (0));
 }
 
+/* SRTP */
+static int setup_srtp(struct sip_srtp **srtp)
+{
+	if (!ast_rtp_engine_srtp_is_registered()) {
+		ast_log(LOG_ERROR, "No SRTP module loaded, can't setup SRTP session.\n");
+		return -1;
+	}
+
+	if (!(*srtp = sip_srtp_alloc())) { /* Allocate SRTP data structure */
+		return -1;
+	}
+
+	return 0;
+}
+
+static int process_crypto(struct sip_pvt *p, struct ast_rtp_instance *rtp, struct sip_srtp **srtp, const char *a)
+{
+	if (strncasecmp(a, "crypto:", 7)) {
+		return FALSE;
+	}
+	if (!*srtp) {
+		if (ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
+			ast_log(LOG_WARNING, "Ignoring unexpected crypto attribute in SDP answer\n");
+			return FALSE;
+		}
+
+		if (setup_srtp(srtp) < 0) {
+			return FALSE;
+		}
+	}
+
+	if (!(*srtp)->crypto && !((*srtp)->crypto = sdp_crypto_setup())) {
+		return FALSE;
+	}
+
+	if (sdp_crypto_process((*srtp)->crypto, a, rtp) < 0) {
+		return FALSE;
+	}
+
+	ast_set_flag(*srtp, SRTP_CRYPTO_OFFER_OK);
+
+	return TRUE;
+}
+
 /*! \brief Reload module */
 static int sip_do_reload(enum channelreloadreason reason)
 {
diff --git a/channels/sip/dialplan_functions.c b/channels/sip/dialplan_functions.c
index d39a2779e..d09627ed8 100644
--- a/channels/sip/dialplan_functions.c
+++ b/channels/sip/dialplan_functions.c
@@ -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;
 	}
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 */
diff --git a/channels/sip/sdp_crypto.c b/channels/sip/sdp_crypto.c
new file mode 100644
index 000000000..94bf85619
--- /dev/null
+++ b/channels/sip/sdp_crypto.c
@@ -0,0 +1,310 @@
+/*
+ * 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.c
+ *
+ * \brief SDP Security descriptions
+ *
+ * Specified in RFC 4568
+ *
+ * \author Mikael Magnusson <mikma@users.sourceforge.net>
+ */
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/options.h"
+#include "asterisk/utils.h"
+#include "include/sdp_crypto.h"
+
+#define SRTP_MASTER_LEN 30
+#define SRTP_MASTERKEY_LEN 16
+#define SRTP_MASTERSALT_LEN ((SRTP_MASTER_LEN) - (SRTP_MASTERKEY_LEN))
+#define SRTP_MASTER_LEN64 (((SRTP_MASTER_LEN) * 8 + 5) / 6 + 1)
+
+extern struct ast_srtp_res *res_srtp;
+extern struct ast_srtp_policy_res *res_srtp_policy;
+
+struct sdp_crypto {
+	char *a_crypto;
+	unsigned char local_key[SRTP_MASTER_LEN];
+	char local_key64[SRTP_MASTER_LEN64];
+};
+
+static int set_crypto_policy(struct ast_srtp_policy *policy, int suite_val, const unsigned char *master_key, unsigned long ssrc, int inbound);
+
+static struct sdp_crypto *sdp_crypto_alloc(void)
+{
+	struct sdp_crypto *crypto;
+
+	return crypto = ast_calloc(1, sizeof(*crypto));
+}
+
+void sdp_crypto_destroy(struct sdp_crypto *crypto)
+{
+	ast_free(crypto->a_crypto);
+	crypto->a_crypto = NULL;
+	ast_free(crypto);
+}
+
+struct sdp_crypto *sdp_crypto_setup(void)
+{
+	struct sdp_crypto *p;
+	int key_len;
+	unsigned char remote_key[SRTP_MASTER_LEN];
+
+	if (!ast_rtp_engine_srtp_is_registered()) {
+		return NULL;
+	}
+
+	if (!(p = sdp_crypto_alloc())) {
+		return NULL;
+	}
+
+	if (res_srtp->get_random(p->local_key, sizeof(p->local_key)) < 0) {
+		sdp_crypto_destroy(p);
+		return NULL;
+	}
+
+	ast_base64encode(p->local_key64, p->local_key, SRTP_MASTER_LEN, sizeof(p->local_key64));
+
+	key_len = ast_base64decode(remote_key, p->local_key64, sizeof(remote_key));
+
+	if (key_len != SRTP_MASTER_LEN) {
+		ast_log(LOG_ERROR, "base64 encode/decode bad len %d != %d\n", key_len, SRTP_MASTER_LEN);
+		ast_free(p);
+		return NULL;
+	}
+
+	if (memcmp(remote_key, p->local_key, SRTP_MASTER_LEN)) {
+		ast_log(LOG_ERROR, "base64 encode/decode bad key\n");
+		ast_free(p);
+		return NULL;
+	}
+
+	ast_debug(1 , "local_key64 %s len %zu\n", p->local_key64, strlen(p->local_key64));
+
+	return p;
+}
+
+static int set_crypto_policy(struct ast_srtp_policy *policy, int suite_val, const unsigned char *master_key, unsigned long ssrc, int inbound)
+{
+	const unsigned char *master_salt = NULL;
+
+	if (!ast_rtp_engine_srtp_is_registered()) {
+		return -1;
+	}
+
+	master_salt = master_key + SRTP_MASTERKEY_LEN;
+	if (res_srtp_policy->set_master_key(policy, master_key, SRTP_MASTERKEY_LEN, master_salt, SRTP_MASTERSALT_LEN) < 0) {
+		return -1;
+	}
+
+	if (res_srtp_policy->set_suite(policy, suite_val)) {
+		ast_log(LOG_WARNING, "Could not set remote SRTP suite\n");
+		return -1;
+	}
+
+	res_srtp_policy->set_ssrc(policy, ssrc, inbound);
+
+	return 0;
+}
+
+static int sdp_crypto_activate(struct sdp_crypto *p, int suite_val, unsigned char *remote_key, struct ast_rtp_instance *rtp)
+{
+	struct ast_srtp_policy *local_policy = NULL;
+	struct ast_srtp_policy *remote_policy = NULL;
+	struct ast_rtp_instance_stats stats = {0,};
+	int res = -1;
+
+	if (!ast_rtp_engine_srtp_is_registered()) {
+		return -1;
+	}
+
+	if (!p) {
+		return -1;
+	}
+
+	if (!(local_policy = res_srtp_policy->alloc())) {
+		return -1;
+	}
+
+	if (!(remote_policy = res_srtp_policy->alloc())) {
+		goto err;
+	}
+
+	if (ast_rtp_instance_get_stats(rtp, &stats, AST_RTP_INSTANCE_STAT_LOCAL_SSRC)) {
+		goto err;
+	}
+
+	if (set_crypto_policy(local_policy, suite_val, p->local_key, stats.local_ssrc, 0) < 0) {
+		goto err;
+	}
+
+	if (set_crypto_policy(remote_policy, suite_val, remote_key, 0, 1) < 0) {
+		goto err;
+	}
+
+	/* FIXME MIKMA */
+	/* ^^^ I wish I knew what needed fixing... */
+	if (ast_rtp_instance_add_srtp_policy(rtp, local_policy)) {
+		ast_log(LOG_WARNING, "Could not set local SRTP policy\n");
+		goto err;
+	}
+
+	if (ast_rtp_instance_add_srtp_policy(rtp, remote_policy)) {
+		ast_log(LOG_WARNING, "Could not set remote SRTP policy\n");
+		goto err;
+	}
+
+	ast_debug(1 , "SRTP policy activated\n");
+	res = 0;
+
+err:
+	if (local_policy) {
+		res_srtp_policy->destroy(local_policy);
+	}
+
+	if (remote_policy) {
+		res_srtp_policy->destroy(remote_policy);
+	}
+
+	return res;
+}
+
+int sdp_crypto_process(struct sdp_crypto *p, const char *attr, struct ast_rtp_instance *rtp)
+{
+	char *str = NULL;
+	char *name = NULL;
+	char *tag = NULL;
+	char *suite = NULL;
+	char *key_params = NULL;
+	char *key_param = NULL;
+	char *session_params = NULL;
+	char *key_salt = NULL;
+	char *lifetime = NULL;
+	int found = 0;
+	int attr_len = strlen(attr);
+	int key_len = 0;
+	int suite_val = 0;
+	unsigned char remote_key[SRTP_MASTER_LEN];
+
+	if (!ast_rtp_engine_srtp_is_registered()) {
+		return -1;
+	}
+
+	str = ast_strdupa(attr);
+
+	name = strsep(&str, ":");
+	tag = strsep(&str, " ");
+	suite = strsep(&str, " ");
+	key_params = strsep(&str, " ");
+	session_params = strsep(&str, " ");
+
+	if (!tag || !suite) {
+		ast_log(LOG_WARNING, "Unrecognized a=%s", attr);
+		return -1;
+	}
+
+	if (session_params) {
+		ast_log(LOG_WARNING, "Unsupported crypto parameters: %s", session_params);
+		return -1;
+	}
+
+	if (!strcmp(suite, "AES_CM_128_HMAC_SHA1_80")) {
+		suite_val = AST_AES_CM_128_HMAC_SHA1_80;
+	} else if (!strcmp(suite, "AES_CM_128_HMAC_SHA1_32")) {
+		suite_val = AST_AES_CM_128_HMAC_SHA1_32;
+	} else {
+		ast_log(LOG_WARNING, "Unsupported crypto suite: %s\n", suite);
+		return -1;
+	}
+
+	while ((key_param = strsep(&key_params, ";"))) {
+		char *method = NULL;
+		char *info = NULL;
+
+		method = strsep(&key_param, ":");
+		info = strsep(&key_param, ";");
+
+		if (!strcmp(method, "inline")) {
+			key_salt = strsep(&info, "|");
+			lifetime = strsep(&info, "|");
+
+			if (lifetime) {
+				ast_log(LOG_NOTICE, "Crypto life time unsupported: %s\n", attr);
+				continue;
+			}
+
+			found = 1;
+			break;
+		}
+	}
+
+	if (!found) {
+		ast_log(LOG_NOTICE, "SRTP crypto offer not acceptable\n");
+		return -1;
+	}
+
+
+	if ((key_len = ast_base64decode(remote_key, key_salt, sizeof(remote_key))) != SRTP_MASTER_LEN) {
+		ast_log(LOG_WARNING, "SRTP sdescriptions key %d != %d\n", key_len, SRTP_MASTER_LEN);
+		return -1;
+	}
+
+	if (sdp_crypto_activate(p, suite_val, remote_key, rtp) < 0) {
+		return -1;
+	}
+
+	if (!p->a_crypto) {
+		if (!(p->a_crypto = ast_calloc(1, attr_len + 11))) {
+			ast_log(LOG_ERROR, "Could not allocate memory for a_crypto\n");
+			return -1;
+		}
+
+		snprintf(p->a_crypto, attr_len + 10, "a=crypto:%s %s inline:%s\r\n", tag, suite, p->local_key64);
+	}
+
+	return 0;
+}
+
+int sdp_crypto_offer(struct sdp_crypto *p)
+{
+	char crypto_buf[128];
+	const char *crypto_suite = "AES_CM_128_HMAC_SHA1_80"; /* Crypto offer */
+
+	if (p->a_crypto) {
+		ast_free(p->a_crypto);
+	}
+
+	if (snprintf(crypto_buf, sizeof(crypto_buf), "a=crypto:1 %s inline:%s\r\n",  crypto_suite, p->local_key64) < 1) {
+		return -1;
+	}
+
+	if (!(p->a_crypto = ast_strdup(crypto_buf))) {
+		return -1;
+	}
+
+	return 0;
+}
+
+const char *sdp_crypto_attrib(struct sdp_crypto *p)
+{
+	return p->a_crypto;
+}
diff --git a/channels/sip/srtp.c b/channels/sip/srtp.c
new file mode 100644
index 000000000..3b55106ab
--- /dev/null
+++ b/channels/sip/srtp.c
@@ -0,0 +1,51 @@
+/*
+ * 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.c
+ *
+ * \brief SIP Secure RTP (SRTP)
+ *
+ * Specified in RFC 3711
+ *
+ * \author Mikael Magnusson <mikma@users.sourceforge.net>
+ */
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/utils.h"
+#include "include/srtp.h"
+
+struct sip_srtp *sip_srtp_alloc(void)
+{
+	struct sip_srtp *srtp;
+
+	srtp = ast_calloc(1, sizeof(*srtp));
+
+	return srtp;
+}
+
+void sip_srtp_destroy(struct sip_srtp *srtp)
+{
+	if (srtp->crypto) {
+		sdp_crypto_destroy(srtp->crypto);
+	}
+	srtp->crypto = NULL;
+	ast_free(srtp);
+}