[asterisk-commits] res rtp asterisk: Use separate SRTP session for RTCP with DTLS (asterisk[master])
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Apr 5 05:37:45 CDT 2016
Joshua Colp has submitted this change and it was merged.
Change subject: res_rtp_asterisk: Use separate SRTP session for RTCP with DTLS
......................................................................
res_rtp_asterisk: Use separate SRTP session for RTCP with DTLS
Asterisk uses separate UDP ports for RTP and RTCP traffic and RFC 5764
explicitly states:
There MUST be a separate DTLS-SRTP session for each distinct pair of
source and destination ports used by a media session
This means RTP keying material cannot be used for DTLS RTCP, which was
the reason why RTCP encryption would fail.
ASTERISK-25642
Change-Id: I7e8779d8b63e371088081bb113131361b2847e3a
---
M include/asterisk/rtp_engine.h
M main/rtp_engine.c
M main/sdp_srtp.c
M res/res_rtp_asterisk.c
4 files changed, 42 insertions(+), 22 deletions(-)
Approvals:
Matt Jordan: Looks good to me, approved
Joshua Colp: Looks good to me, but someone else must approve; Verified
Jacek Konieczny: Looks good to me, but someone else must approve
diff --git a/include/asterisk/rtp_engine.h b/include/asterisk/rtp_engine.h
index 6734033..34fb17d 100644
--- a/include/asterisk/rtp_engine.h
+++ b/include/asterisk/rtp_engine.h
@@ -2228,20 +2228,22 @@
* \param instance the RTP instance
* \param remote_policy the remote endpoint's policy
* \param local_policy our policy for this RTP instance's remote endpoint
+ * \param rtcp 1 for dedicated RTCP policies
*
* \retval 0 Success
* \retval non-zero Failure
*/
-int ast_rtp_instance_add_srtp_policy(struct ast_rtp_instance *instance, struct ast_srtp_policy* remote_policy, struct ast_srtp_policy *local_policy);
+int ast_rtp_instance_add_srtp_policy(struct ast_rtp_instance *instance, struct ast_srtp_policy* remote_policy, struct ast_srtp_policy *local_policy, int rtcp);
/*!
* \brief Obtain the SRTP instance associated with an RTP instance
*
* \param instance the RTP instance
+ * \param rtcp 1 to request instance for RTCP
* \retval the SRTP instance on success
* \retval NULL if no SRTP instance exists
*/
-struct ast_srtp *ast_rtp_instance_get_srtp(struct ast_rtp_instance *instance);
+struct ast_srtp *ast_rtp_instance_get_srtp(struct ast_rtp_instance *instance, int rtcp);
/*! \brief Custom formats declared in codecs.conf at startup must be communicated to the rtp_engine
* so their mime type can payload number can be initialized. */
diff --git a/main/rtp_engine.c b/main/rtp_engine.c
index fd472ba..11e94c6 100644
--- a/main/rtp_engine.c
+++ b/main/rtp_engine.c
@@ -188,6 +188,8 @@
struct ast_rtp_glue *glue;
/*! SRTP info associated with the instance */
struct ast_srtp *srtp;
+ /*! SRTP info dedicated for RTCP associated with the instance */
+ struct ast_srtp *rtcp_srtp;
/*! Channel unique ID */
char channel_uniqueid[AST_MAX_UNIQUEID];
/*! Time of last packet sent */
@@ -362,6 +364,10 @@
if (instance->srtp) {
res_srtp->destroy(instance->srtp);
+ }
+
+ if (instance->rtcp_srtp) {
+ res_srtp->destroy(instance->rtcp_srtp);
}
ast_rtp_codecs_payloads_destroy(&instance->codecs);
@@ -2052,29 +2058,38 @@
return res_srtp && res_srtp_policy;
}
-int ast_rtp_instance_add_srtp_policy(struct ast_rtp_instance *instance, struct ast_srtp_policy *remote_policy, struct ast_srtp_policy *local_policy)
+int ast_rtp_instance_add_srtp_policy(struct ast_rtp_instance *instance, struct ast_srtp_policy *remote_policy, struct ast_srtp_policy *local_policy, int rtcp)
{
int res = 0;
+ struct ast_srtp **srtp;
if (!res_srtp) {
return -1;
}
- if (!instance->srtp) {
- res = res_srtp->create(&instance->srtp, instance, remote_policy);
+
+ srtp = rtcp ? &instance->rtcp_srtp : &instance->srtp;
+
+ if (!*srtp) {
+ res = res_srtp->create(srtp, instance, remote_policy);
} else {
- res = res_srtp->replace(&instance->srtp, instance, remote_policy);
+ res = res_srtp->replace(srtp, instance, remote_policy);
}
if (!res) {
- res = res_srtp->add_stream(instance->srtp, local_policy);
+ res = res_srtp->add_stream(*srtp, local_policy);
}
return res;
}
-struct ast_srtp *ast_rtp_instance_get_srtp(struct ast_rtp_instance *instance)
+struct ast_srtp *ast_rtp_instance_get_srtp(struct ast_rtp_instance *instance, int rtcp)
{
- return instance->srtp;
+ if (rtcp && instance->rtcp_srtp) {
+ return instance->rtcp_srtp;
+ }
+ else {
+ return instance->srtp;
+ }
}
int ast_rtp_instance_sendcng(struct ast_rtp_instance *instance, int level)
diff --git a/main/sdp_srtp.c b/main/sdp_srtp.c
index e576258..c92387b 100644
--- a/main/sdp_srtp.c
+++ b/main/sdp_srtp.c
@@ -183,7 +183,7 @@
}
/* Add the SRTP policies */
- if (ast_rtp_instance_add_srtp_policy(rtp, remote_policy, local_policy)) {
+ if (ast_rtp_instance_add_srtp_policy(rtp, remote_policy, local_policy, 0)) {
ast_log(LOG_WARNING, "Could not set SRTP policies\n");
goto err;
}
diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index 9c7432e..789535e 100644
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -1982,19 +1982,20 @@
return 0;
}
-static int dtls_srtp_setup(struct ast_rtp *rtp, struct ast_srtp *srtp, struct ast_rtp_instance *instance)
+static int dtls_srtp_setup(struct ast_rtp *rtp, struct ast_srtp *srtp, struct ast_rtp_instance *instance, int rtcp)
{
unsigned char material[SRTP_MASTER_LEN * 2];
unsigned char *local_key, *local_salt, *remote_key, *remote_salt;
struct ast_srtp_policy *local_policy, *remote_policy = NULL;
struct ast_rtp_instance_stats stats = { 0, };
int res = -1;
+ struct dtls_details *dtls = !rtcp ? &rtp->dtls : &rtp->rtcp->dtls;
/* If a fingerprint is present in the SDP make sure that the peer certificate matches it */
if (rtp->dtls_verify & AST_RTP_DTLS_VERIFY_FINGERPRINT) {
X509 *certificate;
- if (!(certificate = SSL_get_peer_certificate(rtp->dtls.ssl))) {
+ if (!(certificate = SSL_get_peer_certificate(dtls->ssl))) {
ast_log(LOG_WARNING, "No certificate was provided by the peer on RTP instance '%p'\n", instance);
return -1;
}
@@ -2028,14 +2029,14 @@
}
/* Ensure that certificate verification was successful */
- if ((rtp->dtls_verify & AST_RTP_DTLS_VERIFY_CERTIFICATE) && SSL_get_verify_result(rtp->dtls.ssl) != X509_V_OK) {
+ if ((rtp->dtls_verify & AST_RTP_DTLS_VERIFY_CERTIFICATE) && SSL_get_verify_result(dtls->ssl) != X509_V_OK) {
ast_log(LOG_WARNING, "Peer certificate on RTP instance '%p' failed verification test\n",
instance);
return -1;
}
/* Produce key information and set up SRTP */
- if (!SSL_export_keying_material(rtp->dtls.ssl, material, SRTP_MASTER_LEN * 2, "EXTRACTOR-dtls_srtp", 19, NULL, 0, 0)) {
+ if (!SSL_export_keying_material(dtls->ssl, material, SRTP_MASTER_LEN * 2, "EXTRACTOR-dtls_srtp", 19, NULL, 0, 0)) {
ast_log(LOG_WARNING, "Unable to extract SRTP keying material from DTLS-SRTP negotiation on RTP instance '%p'\n",
instance);
return -1;
@@ -2090,7 +2091,7 @@
res_srtp_policy->set_ssrc(remote_policy, 0, 1);
- if (ast_rtp_instance_add_srtp_policy(instance, remote_policy, local_policy)) {
+ if (ast_rtp_instance_add_srtp_policy(instance, remote_policy, local_policy, rtcp)) {
ast_log(LOG_WARNING, "Could not set policies when setting up DTLS-SRTP on '%p'\n", rtp);
goto error;
}
@@ -2121,7 +2122,7 @@
{
int len;
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
- struct ast_srtp *srtp = ast_rtp_instance_get_srtp(instance);
+ struct ast_srtp *srtp = ast_rtp_instance_get_srtp(instance, rtcp);
char *in = buf;
#ifdef HAVE_PJPROJECT
struct ast_sockaddr *loop = rtcp ? &rtp->rtcp_loop : &rtp->rtp_loop;
@@ -2178,10 +2179,8 @@
if (SSL_is_init_finished(dtls->ssl)) {
/* Any further connections will be existing since this is now established */
dtls->connection = AST_RTP_DTLS_CONNECTION_EXISTING;
- if (!rtcp) {
- /* Use the keying material to set up key/salt information */
- res = dtls_srtp_setup(rtp, srtp, instance);
- }
+ /* Use the keying material to set up key/salt information */
+ res = dtls_srtp_setup(rtp, srtp, instance, rtcp);
} else {
/* Since we've sent additional traffic start the timeout timer for retransmission */
dtls_srtp_start_timeout_timer(instance, rtp, rtcp);
@@ -2250,7 +2249,7 @@
int len = size;
void *temp = buf;
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
- struct ast_srtp *srtp = ast_rtp_instance_get_srtp(instance);
+ struct ast_srtp *srtp = ast_rtp_instance_get_srtp(instance, rtcp);
int res;
*ice = 0;
@@ -2950,7 +2949,8 @@
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);
+ struct ast_srtp *srtp = ast_rtp_instance_get_srtp(instance, 0);
+ struct ast_srtp *rtcp_srtp = ast_rtp_instance_get_srtp(instance, 1);
unsigned int ssrc = ast_random();
if (!rtp->lastts) {
@@ -2966,6 +2966,9 @@
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);
+ if (rtcp_srtp != srtp) {
+ res_srtp->change_source(srtp, rtp->ssrc, ssrc);
+ }
}
rtp->ssrc = ssrc;
--
To view, visit https://gerrit.asterisk.org/2492
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-MessageType: merged
Gerrit-Change-Id: I7e8779d8b63e371088081bb113131361b2847e3a
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Owner: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: Jacek Konieczny <jkonieczny at eggsoft.pl>
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Matt Jordan <mjordan at digium.com>
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