[asterisk-commits] twilson: trunk r284479 - in /trunk: ./ channels/ include/asterisk/ main/ res/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Wed Sep 1 13:52:35 CDT 2010


Author: twilson
Date: Wed Sep  1 13:52:27 2010
New Revision: 284479

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=284479
Log:
Merged revisions 284477 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r284477 | twilson | 2010-09-01 13:44:36 -0500 (Wed, 01 Sep 2010) | 17 lines
  
  Fix SRTP for changing SSRC and multiple a=crypto SDP lines
  
  Adding code to Asterisk that changed the SSRC during bridges and masquerades
  broke SRTP functionality. Also broken was handling the situation where an
  incoming INVITE had more than one crypto offer. This patch caches the SRTP
  policies the we use so that we can change the ssrc and inform libsrtp of the
  new streams. It also uses the first acceptable a=crypto line from the incoming
  INVITE.
  
  (closes issue #17563)
  Reported by: Alexcr
  Patches: 
        srtp.diff uploaded by twilson (license 396)
  Tested by: twilson
  
  Review: https://reviewboard.asterisk.org/r/878/
........

Modified:
    trunk/   (props changed)
    trunk/channels/chan_sip.c
    trunk/include/asterisk/res_srtp.h
    trunk/main/rtp_engine.c
    trunk/res/res_rtp_asterisk.c
    trunk/res/res_srtp.c

Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.8-merged' - no diff available.

Modified: trunk/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_sip.c?view=diff&rev=284479&r1=284478&r2=284479
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Wed Sep  1 13:52:27 2010
@@ -27817,6 +27817,12 @@
 		}
 	}
 
+	/* For now, when we receive an INVITE just take the first successful crypto line */
+	if ((*srtp)->crypto && !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
+		ast_debug(3, "We've already processed a crypto attribute, skipping '%s'\n", a);
+		return FALSE;
+	}
+
 	if (!(*srtp)->crypto && !((*srtp)->crypto = sdp_crypto_setup())) {
 		return FALSE;
 	}

Modified: trunk/include/asterisk/res_srtp.h
URL: http://svnview.digium.com/svn/asterisk/trunk/include/asterisk/res_srtp.h?view=diff&rev=284479&r1=284478&r2=284479
==============================================================================
--- trunk/include/asterisk/res_srtp.h (original)
+++ trunk/include/asterisk/res_srtp.h Wed Sep  1 13:52:27 2010
@@ -33,6 +33,7 @@
 	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);
+	int (*change_source)(struct ast_srtp *srtp, unsigned int from_ssrc, unsigned int to_ssrc);
 	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);

Modified: trunk/main/rtp_engine.c
URL: http://svnview.digium.com/svn/asterisk/trunk/main/rtp_engine.c?view=diff&rev=284479&r1=284478&r2=284479
==============================================================================
--- trunk/main/rtp_engine.c (original)
+++ trunk/main/rtp_engine.c Wed Sep  1 13:52:27 2010
@@ -280,6 +280,10 @@
 	if (instance->data && instance->engine->destroy(instance)) {
 		ast_debug(1, "Engine '%s' failed to destroy RTP instance '%p'\n", instance->engine->name, instance);
 		return;
+	}
+
+	if (instance->srtp) {
+		res_srtp->destroy(instance->srtp);
 	}
 
 	/* Drop our engine reference */

Modified: trunk/res/res_rtp_asterisk.c
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_rtp_asterisk.c?view=diff&rev=284479&r1=284478&r2=284479
==============================================================================
--- trunk/res/res_rtp_asterisk.c (original)
+++ trunk/res/res_rtp_asterisk.c Wed Sep  1 13:52:27 2010
@@ -717,12 +717,24 @@
 static void ast_rtp_change_source(struct ast_rtp_instance *instance)
 {
 	struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+	struct ast_srtp *srtp = ast_rtp_instance_get_srtp(instance);
 	unsigned int ssrc = ast_random();
+
+	if (!rtp->lastts) {
+		ast_debug(3, "Not changing SSRC since we haven't sent any RTP yet\n");
+		return;
+	}
 
 	/* We simply set this bit so that the next packet sent will have the marker bit turned on */
 	ast_set_flag(rtp, FLAG_NEED_MARKER_BIT);
 
 	ast_debug(3, "Changing ssrc from %u to %u due to a source change\n", rtp->ssrc, ssrc);
+
+	if (srtp) {
+		ast_debug(3, "Changing ssrc for SRTP from %u to %u\n", rtp->ssrc, ssrc);
+		res_srtp->change_source(srtp, rtp->ssrc, ssrc);
+	}
+
 	rtp->ssrc = ssrc;
 
 	return;

Modified: trunk/res/res_srtp.c
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_srtp.c?view=diff&rev=284479&r1=284478&r2=284479
==============================================================================
--- trunk/res/res_srtp.c (original)
+++ trunk/res/res_srtp.c Wed Sep  1 13:52:27 2010
@@ -53,14 +53,15 @@
 #include "asterisk/module.h"
 #include "asterisk/options.h"
 #include "asterisk/rtp_engine.h"
+#include "asterisk/astobj2.h"
 
 struct ast_srtp {
 	struct ast_rtp_instance *rtp;
+	struct ao2_container *policies;
 	srtp_t session;
 	const struct ast_srtp_cb *cb;
 	void *data;
 	unsigned char buf[8192 + AST_FRIENDLY_OFFSET];
-	unsigned int has_stream:1;
 };
 
 struct ast_srtp_policy {
@@ -73,6 +74,7 @@
 static int ast_srtp_create(struct ast_srtp **srtp, struct ast_rtp_instance *rtp, struct ast_srtp_policy *policy);
 static void ast_srtp_destroy(struct ast_srtp *srtp);
 static int ast_srtp_add_stream(struct ast_srtp *srtp, struct ast_srtp_policy *policy);
+static int ast_srtp_change_source(struct ast_srtp *srtp, unsigned int from_ssrc, unsigned int to_ssrc);
 
 static int ast_srtp_unprotect(struct ast_srtp *srtp, void *buf, int *len, int rtcp);
 static int ast_srtp_protect(struct ast_srtp *srtp, void **buf, int *len, int rtcp);
@@ -90,6 +92,7 @@
 	.create = ast_srtp_create,
 	.destroy = ast_srtp_destroy,
 	.add_stream = ast_srtp_add_stream,
+	.change_source = ast_srtp_change_source,
 	.set_cb = ast_srtp_set_cb,
 	.unprotect = ast_srtp_unprotect,
 	.protect = ast_srtp_protect,
@@ -144,12 +147,43 @@
 	}
 }
 
+static int policy_hash_fn(const void *obj, const int flags)
+{
+	const struct ast_srtp_policy *policy = obj;
+
+	return policy->sp.ssrc.type == ssrc_specific ? policy->sp.ssrc.value : policy->sp.ssrc.type;
+}
+
+static int policy_cmp_fn(void *obj, void *arg, int flags)
+{
+	const struct ast_srtp_policy *one = obj, *two = arg;
+
+	return one->sp.ssrc.type == two->sp.ssrc.type && one->sp.ssrc.value == two->sp.ssrc.value;
+}
+
+static struct ast_srtp_policy *find_policy(struct ast_srtp *srtp, const srtp_policy_t *policy, int flags)
+{
+	struct ast_srtp_policy tmp = {
+		.sp = {
+			.ssrc.type = policy->ssrc.type,
+			.ssrc.value = policy->ssrc.value,
+		},
+	};
+
+	return ao2_t_find(srtp->policies, &tmp, flags, "Looking for policy");
+}
+
 static struct ast_srtp *res_srtp_new(void)
 {
 	struct ast_srtp *srtp;
 
 	if (!(srtp = ast_calloc(1, sizeof(*srtp)))) {
 		ast_log(LOG_ERROR, "Unable to allocate memory for srtp\n");
+		return NULL;
+	}
+
+	if (!(srtp->policies = ao2_t_container_alloc(5, policy_hash_fn, policy_cmp_fn, "SRTP policy container"))) {
+		ast_free(srtp);
 		return NULL;
 	}
 
@@ -188,24 +222,30 @@
 	}
 }
 
-static struct ast_srtp_policy *ast_srtp_policy_alloc()
-{
-	struct ast_srtp_policy *tmp;
-
-	if (!(tmp = ast_calloc(1, sizeof(*tmp)))) {
-		ast_log(LOG_ERROR, "Unable to allocate memory for srtp_policy\n");
-	}
-
-	return tmp;
-}
-
-static void ast_srtp_policy_destroy(struct ast_srtp_policy *policy)
-{
+static void policy_destructor(void *obj)
+{
+	struct ast_srtp_policy *policy = obj;
+
 	if (policy->sp.key) {
 		ast_free(policy->sp.key);
 		policy->sp.key = NULL;
 	}
-	ast_free(policy);
+}
+
+static struct ast_srtp_policy *ast_srtp_policy_alloc()
+{
+	struct ast_srtp_policy *tmp;
+
+	if (!(tmp = ao2_t_alloc(sizeof(*tmp), policy_destructor, "Allocating policy"))) {
+		ast_log(LOG_ERROR, "Unable to allocate memory for srtp_policy\n");
+	}
+
+	return tmp;
+}
+
+static void ast_srtp_policy_destroy(struct ast_srtp_policy *policy)
+{
+	ao2_t_ref(policy, -1, "Destroying policy");
 }
 
 static int policy_set_suite(crypto_policy_t *p, enum ast_srtp_suite suite)
@@ -344,6 +384,8 @@
 	temp->rtp = rtp;
 	*srtp = temp;
 
+	ao2_t_link((*srtp)->policies, policy, "Created initial policy");
+
 	return 0;
 }
 
@@ -353,16 +395,52 @@
 		srtp_dealloc(srtp->session);
 	}
 
+	ao2_t_callback(srtp->policies, OBJ_UNLINK | OBJ_NODATA | OBJ_MULTIPLE, NULL, NULL, "Unallocate policy");
+	ao2_t_ref(srtp->policies, -1, "Destroying container");
+
 	ast_free(srtp);
 }
 
 static int ast_srtp_add_stream(struct ast_srtp *srtp, struct ast_srtp_policy *policy)
 {
-	if (!srtp->has_stream && srtp_add_stream(srtp->session, &policy->sp) != err_status_ok) {
-		return -1;
-	}
-
-	srtp->has_stream = 1;
+	struct ast_srtp_policy *match;
+
+	if ((match = find_policy(srtp, &policy->sp, OBJ_POINTER))) {
+		ast_debug(3, "Policy already exists, not re-adding\n");
+		ao2_t_ref(match, -1, "Unreffing already existing policy");
+		return -1;
+	}
+
+	if (srtp_add_stream(srtp->session, &policy->sp) != err_status_ok) {
+		return -1;
+	}
+
+	ao2_t_link(srtp->policies, policy, "Added additional stream");
+
+	return 0;
+}
+
+static int ast_srtp_change_source(struct ast_srtp *srtp, unsigned int from_ssrc, unsigned int to_ssrc)
+{
+	struct ast_srtp_policy *match;
+	struct srtp_policy_t sp = {
+		.ssrc.type = ssrc_specific,
+		.ssrc.value = from_ssrc,
+	};
+	err_status_t status;
+
+	/* If we find a mach, return and unlink it from the container so we
+	 * can change the SSRC (which is part of the hash) and then have
+	 * ast_srtp_add_stream link it back in if all is well */
+	if ((match = find_policy(srtp, &sp, OBJ_POINTER | OBJ_UNLINK))) {
+		match->sp.ssrc.value = to_ssrc;
+		if (ast_srtp_add_stream(srtp, match)) {
+			ast_log(LOG_WARNING, "Couldn't add stream\n");
+		} else if ((status = srtp_remove_stream(srtp->session, from_ssrc))) {
+			ast_debug(3, "Couldn't remove stream (%d)\n", status);
+		}
+		ao2_t_ref(match, -1, "Unreffing found policy in change_source");
+	}
 
 	return 0;
 }




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