[asterisk-commits] AST-2017-008: Improve RTP and RTCP packet processing. (asterisk[master])
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Thu Sep 21 14:48:17 CDT 2017
Joshua Colp has submitted this change and it was merged. ( https://gerrit.asterisk.org/6414 )
Change subject: AST-2017-008: Improve RTP and RTCP packet processing.
......................................................................
AST-2017-008: Improve RTP and RTCP packet processing.
Validate RTCP packets before processing them.
* Validate that the received packet is of a minimum length and apply the
RFC3550 RTCP packet validation checks.
* Fixed potentially reading garbage beyond the received RTCP record data.
* Fixed rtp->themssrc only being set once when the remote could change
the SSRC. We would effectively stop handling the RTCP statistic records.
* Fixed rtp->themssrc to not treat a zero value as special by adding
rtp->themssrc_valid to indicate if rtp->themssrc is available.
ASTERISK-27274
Make strict RTP learning more flexible.
Direct media can cause strict RTP to attempt to learn a remote address
again before it has had a chance to learn the remote address the first
time. Because of the rapid relearn requests, strict RTP could latch onto
the first remote address and fail to latch onto the direct media remote
address. As a result, you have one way audio until the call is placed on
and off hold.
The new algorithm learns remote addresses for a set time (1.5 seconds)
before locking the remote address. In addition, we must see a configured
number of remote packets from the same address in a row before switching.
* Fixed strict RTP learning from always accepting the first new address
packet as the new stream.
* Fixed strict RTP to initialize the expected sequence number with the
last received sequence number instead of the last transmitted sequence
number.
* Fixed the predicted next sequence number calculation in
rtp_learning_rtp_seq_update() to handle overflow.
ASTERISK-27252
Change-Id: Ia2d3aa6e0f22906c25971e74f10027d96525f31c
---
M res/res_rtp_asterisk.c
1 file changed, 432 insertions(+), 104 deletions(-)
Approvals:
George Joseph: Looks good to me, but someone else must approve
Joshua Colp: Looks good to me, approved; Approved for Submit
diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index f6a0ec0..1440eb0 100644
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -125,7 +125,9 @@
STRICT_RTP_CLOSED, /*! Drop all RTP packets not coming from source that was learned */
};
-#define DEFAULT_STRICT_RTP STRICT_RTP_CLOSED
+#define STRICT_RTP_LEARN_TIMEOUT 1500 /*!< milliseconds */
+
+#define DEFAULT_STRICT_RTP -1 /*!< Enabled */
#define DEFAULT_ICESUPPORT 1
extern struct ast_srtp_res *res_srtp;
@@ -226,9 +228,11 @@
/*! \brief RTP learning mode tracking information */
struct rtp_learning_info {
- int max_seq; /*!< The highest sequence number received */
- int packets; /*!< The number of remaining packets before the source is accepted */
+ struct ast_sockaddr proposed_address; /*!< Proposed remote address for strict RTP */
+ struct timeval start; /*!< The time learning mode was started */
struct timeval received; /*!< The time of the last received packet */
+ int max_seq; /*!< The highest sequence number received */
+ int packets; /*!< The number of remaining packets before the source is accepted */
};
#ifdef HAVE_OPENSSL_SRTP
@@ -266,7 +270,7 @@
unsigned int ssrc; /*!< Synchronization source, RFC 3550, page 10. */
char cname[AST_UUID_STR_LEN]; /*!< Our local CNAME */
unsigned int themssrc; /*!< Their SSRC */
- unsigned int rxssrc;
+ unsigned int themssrc_valid; /*!< True if their SSRC is available. */
unsigned int lastts;
unsigned int lastrxts;
unsigned int lastividtimestamp;
@@ -2036,7 +2040,7 @@
#endif
#ifdef HAVE_PJPROJECT
-static void rtp_learning_seq_init(struct rtp_learning_info *info, uint16_t seq);
+static void rtp_learning_start(struct ast_rtp *rtp);
/* PJPROJECT ICE callback */
static void ast_rtp_on_ice_complete(pj_ice_sess *ice, pj_status_t status)
@@ -2078,8 +2082,8 @@
return;
}
- rtp->strict_rtp_state = STRICT_RTP_LEARN;
- rtp_learning_seq_init(&rtp->rtp_source_learn, (uint16_t)rtp->seqno);
+ ast_verb(4, "%p -- Strict RTP learning after ICE completion\n", rtp);
+ rtp_learning_start(rtp);
ao2_unlock(instance);
}
@@ -2828,7 +2832,7 @@
*/
static void rtp_learning_seq_init(struct rtp_learning_info *info, uint16_t seq)
{
- info->max_seq = seq - 1;
+ info->max_seq = seq;
info->packets = learning_min_sequential;
memset(&info->received, 0, sizeof(info->received));
}
@@ -2845,14 +2849,17 @@
*/
static int rtp_learning_rtp_seq_update(struct rtp_learning_info *info, uint16_t seq)
{
+ /*
+ * During the learning mode the minimum amount of media we'll accept is
+ * 10ms so give a reasonable 5ms buffer just in case we get it sporadically.
+ */
if (!ast_tvzero(info->received) && ast_tvdiff_ms(ast_tvnow(), info->received) < 5) {
- /* During the probation period the minimum amount of media we'll accept is
- * 10ms so give a reasonable 5ms buffer just in case we get it sporadically.
+ /*
+ * Reject a flood of packets as acceptable for learning.
+ * Reset the needed packets.
*/
- return 1;
- }
-
- if (seq == info->max_seq + 1) {
+ info->packets = learning_min_sequential - 1;
+ } else if (seq == (uint16_t) (info->max_seq + 1)) {
/* packet is in sequence */
info->packets--;
} else {
@@ -2862,7 +2869,23 @@
info->max_seq = seq;
info->received = ast_tvnow();
- return (info->packets == 0);
+ return info->packets;
+}
+
+/*!
+ * \brief Start the strictrtp learning mode.
+ *
+ * \param rtp RTP session description
+ *
+ * \return Nothing
+ */
+static void rtp_learning_start(struct ast_rtp *rtp)
+{
+ rtp->strict_rtp_state = STRICT_RTP_LEARN;
+ memset(&rtp->rtp_source_learn.proposed_address, 0,
+ sizeof(rtp->rtp_source_learn.proposed_address));
+ rtp->rtp_source_learn.start = ast_tvnow();
+ rtp_learning_seq_init(&rtp->rtp_source_learn, (uint16_t) rtp->lastrxseqno);
}
#ifdef HAVE_PJPROJECT
@@ -3123,10 +3146,7 @@
{
int x, startplace;
- rtp->strict_rtp_state = (strictrtp ? STRICT_RTP_LEARN : STRICT_RTP_OPEN);
- if (strictrtp) {
- rtp_learning_seq_init(&rtp->rtp_source_learn, (uint16_t)rtp->seqno);
- }
+ rtp->strict_rtp_state = (strictrtp ? STRICT_RTP_CLOSED : STRICT_RTP_OPEN);
/* Create a new socket for us to listen on and use */
if ((rtp->s =
@@ -3762,7 +3782,7 @@
struct ast_sockaddr remote_address = { { 0, } };
struct ast_rtp_rtcp_report_block *report_block = NULL;
RAII_VAR(struct ast_rtp_rtcp_report *, rtcp_report,
- ast_rtp_rtcp_report_alloc(rtp->themssrc ? 1 : 0),
+ ast_rtp_rtcp_report_alloc(rtp->themssrc_valid ? 1 : 0),
ao2_cleanup);
if (!rtp || !rtp->rtcp) {
@@ -3782,7 +3802,7 @@
calculate_lost_packet_statistics(rtp, &lost_packets, &fraction_lost);
gettimeofday(&now, NULL);
- rtcp_report->reception_report_count = rtp->themssrc ? 1 : 0;
+ rtcp_report->reception_report_count = rtp->themssrc_valid ? 1 : 0;
rtcp_report->ssrc = rtp->ssrc;
rtcp_report->type = sr ? RTCP_PT_SR : RTCP_PT_RR;
if (sr) {
@@ -3792,7 +3812,7 @@
rtcp_report->sender_information.octet_count = rtp->txoctetcount;
}
- if (rtp->themssrc) {
+ if (rtp->themssrc_valid) {
report_block = ast_calloc(1, sizeof(*report_block));
if (!report_block) {
return 1;
@@ -4179,6 +4199,10 @@
/*
* RTCP was stopped.
*/
+ return 0;
+ }
+ if (!rtp->themssrc_valid) {
+ /* We don't know their SSRC value so we don't know who to update. */
return 0;
}
@@ -4778,93 +4802,293 @@
return found;
}
+static const char *rtcp_payload_type2str(unsigned int pt)
+{
+ const char *str;
+
+ switch (pt) {
+ case RTCP_PT_SR:
+ str = "Sender Report";
+ break;
+ case RTCP_PT_RR:
+ str = "Receiver Report";
+ break;
+ case RTCP_PT_FUR:
+ /* Full INTRA-frame Request / Fast Update Request */
+ str = "H.261 FUR";
+ break;
+ case RTCP_PT_PSFB:
+ /* Payload Specific Feed Back */
+ str = "PSFB";
+ break;
+ case RTCP_PT_SDES:
+ str = "Source Description";
+ break;
+ case RTCP_PT_BYE:
+ str = "BYE";
+ break;
+ default:
+ str = "Unknown";
+ break;
+ }
+ return str;
+}
+
+/*
+ * Unshifted RTCP header bit field masks
+ */
+#define RTCP_LENGTH_MASK 0xFFFF
+#define RTCP_PAYLOAD_TYPE_MASK 0xFF
+#define RTCP_REPORT_COUNT_MASK 0x1F
+#define RTCP_PADDING_MASK 0x01
+#define RTCP_VERSION_MASK 0x03
+
+/*
+ * RTCP header bit field shift offsets
+ */
+#define RTCP_LENGTH_SHIFT 0
+#define RTCP_PAYLOAD_TYPE_SHIFT 16
+#define RTCP_REPORT_COUNT_SHIFT 24
+#define RTCP_PADDING_SHIFT 29
+#define RTCP_VERSION_SHIFT 30
+
+#define RTCP_VERSION 2U
+#define RTCP_VERSION_SHIFTED (RTCP_VERSION << RTCP_VERSION_SHIFT)
+#define RTCP_VERSION_MASK_SHIFTED (RTCP_VERSION_MASK << RTCP_VERSION_SHIFT)
+
+/*
+ * RTCP first packet record validity header mask and value.
+ *
+ * RFC3550 intentionally defines the encoding of RTCP_PT_SR and RTCP_PT_RR
+ * such that they differ in the least significant bit. Either of these two
+ * payload types MUST be the first RTCP packet record in a compound packet.
+ *
+ * RFC3550 checks the padding bit in the algorithm they use to check the
+ * RTCP packet for validity. However, we aren't masking the padding bit
+ * to check since we don't know if it is a compound RTCP packet or not.
+ */
+#define RTCP_VALID_MASK (RTCP_VERSION_MASK_SHIFTED | (((RTCP_PAYLOAD_TYPE_MASK & ~0x1)) << RTCP_PAYLOAD_TYPE_SHIFT))
+#define RTCP_VALID_VALUE (RTCP_VERSION_SHIFTED | (RTCP_PT_SR << RTCP_PAYLOAD_TYPE_SHIFT))
+
+#define RTCP_SR_BLOCK_WORD_LENGTH 5
+#define RTCP_RR_BLOCK_WORD_LENGTH 6
+#define RTCP_HEADER_SSRC_LENGTH 2
+
static struct ast_frame *ast_rtcp_interpret(struct ast_rtp_instance *instance, const unsigned char *rtcpdata, size_t size, struct ast_sockaddr *addr)
{
struct ast_rtp_instance *transport = instance;
struct ast_rtp *transport_rtp = ast_rtp_instance_get_data(instance);
unsigned int *rtcpheader = (unsigned int *)(rtcpdata);
- int packetwords, position = 0;
+ unsigned int packetwords;
+ unsigned int position;
+ unsigned int first_word;
+ /*! True if we have seen an acceptable SSRC to learn the remote RTCP address */
+ unsigned int ssrc_seen;
int report_counter = 0;
struct ast_rtp_rtcp_report_block *report_block;
struct ast_frame *f = &ast_null_frame;
packetwords = size / 4;
- ast_debug(1, "Got RTCP report of %zu bytes\n", size);
+ ast_debug(1, "Got RTCP report of %zu bytes from %s\n",
+ size, ast_sockaddr_stringify(addr));
+ /*
+ * Validate the RTCP packet according to an adapted and slightly
+ * modified RFC3550 validation algorithm.
+ */
+ if (packetwords < RTCP_HEADER_SSRC_LENGTH) {
+ ast_debug(1, "%p -- RTCP from %s: Frame size (%u words) is too short\n",
+ transport_rtp, ast_sockaddr_stringify(addr), packetwords);
+ return &ast_null_frame;
+ }
+ position = 0;
+ first_word = ntohl(rtcpheader[position]);
+ if ((first_word & RTCP_VALID_MASK) != RTCP_VALID_VALUE) {
+ ast_debug(1, "%p -- RTCP from %s: Failed first packet validity check\n",
+ transport_rtp, ast_sockaddr_stringify(addr));
+ return &ast_null_frame;
+ }
+ do {
+ position += ((first_word >> RTCP_LENGTH_SHIFT) & RTCP_LENGTH_MASK) + 1;
+ if (packetwords <= position) {
+ break;
+ }
+ first_word = ntohl(rtcpheader[position]);
+ } while ((first_word & RTCP_VERSION_MASK_SHIFTED) == RTCP_VERSION_SHIFTED);
+ if (position != packetwords) {
+ ast_debug(1, "%p -- RTCP from %s: Failed packet version or length check\n",
+ transport_rtp, ast_sockaddr_stringify(addr));
+ return &ast_null_frame;
+ }
+
+ /*
+ * Note: RFC3605 points out that true NAT (vs NAPT) can cause RTCP
+ * to have a different IP address and port than RTP. Otherwise, when
+ * strictrtp is enabled we could reject RTCP packets not coming from
+ * the learned RTP IP address if it is available.
+ */
+
+ /*
+ * strictrtp safety needs SSRC to match before we use the
+ * sender's address for symmetrical RTP to send our RTCP
+ * reports.
+ *
+ * If strictrtp is not enabled then claim to have already seen
+ * a matching SSRC so we'll accept this packet's address for
+ * symmetrical RTP.
+ */
+ ssrc_seen = transport_rtp->strict_rtp_state == STRICT_RTP_OPEN;
+
+ position = 0;
while (position < packetwords) {
- int i, pt, rc;
+ unsigned int i;
+ unsigned int pt;
+ unsigned int rc;
+ unsigned int ssrc;
+ /*! True if the ssrc value we have is valid and not garbage because it doesn't exist. */
+ unsigned int ssrc_valid;
unsigned int length;
+ unsigned int min_length;
+
struct ast_json *message_blob;
RAII_VAR(struct ast_rtp_rtcp_report *, rtcp_report, NULL, ao2_cleanup);
struct ast_rtp_instance *child;
struct ast_rtp *rtp;
i = position;
- length = ntohl(rtcpheader[i]);
- pt = (length & 0xff0000) >> 16;
- rc = (length & 0x1f000000) >> 24;
- length &= 0xffff;
+ first_word = ntohl(rtcpheader[i]);
+ pt = (first_word >> RTCP_PAYLOAD_TYPE_SHIFT) & RTCP_PAYLOAD_TYPE_MASK;
+ rc = (first_word >> RTCP_REPORT_COUNT_SHIFT) & RTCP_REPORT_COUNT_MASK;
+ /* RFC3550 says 'length' is the number of words in the packet - 1 */
+ length = ((first_word >> RTCP_LENGTH_SHIFT) & RTCP_LENGTH_MASK) + 1;
- rtcp_report = ast_rtp_rtcp_report_alloc(rc);
- if (!rtcp_report) {
+ /* Check expected RTCP packet record length */
+ min_length = RTCP_HEADER_SSRC_LENGTH;
+ switch (pt) {
+ case RTCP_PT_SR:
+ min_length += RTCP_SR_BLOCK_WORD_LENGTH;
+ /* fall through */
+ case RTCP_PT_RR:
+ min_length += (rc * RTCP_RR_BLOCK_WORD_LENGTH);
+ break;
+ case RTCP_PT_FUR:
+ case RTCP_PT_PSFB:
+ break;
+ case RTCP_PT_SDES:
+ case RTCP_PT_BYE:
+ /*
+ * There may not be a SSRC/CSRC present. The packet is
+ * useless but still valid if it isn't present.
+ *
+ * We don't know what min_length should be so disable the check
+ */
+ min_length = length;
+ break;
+ default:
+ ast_debug(1, "%p -- RTCP from %s: %u(%s) skipping record\n",
+ transport_rtp, ast_sockaddr_stringify(addr), pt, rtcp_payload_type2str(pt));
+ if (rtcp_debug_test_addr(addr)) {
+ ast_verbose("\n");
+ ast_verbose("RTCP from %s: %u(%s) skipping record\n",
+ ast_sockaddr_stringify(addr), pt, rtcp_payload_type2str(pt));
+ }
+ position += length;
+ continue;
+ }
+ if (length < min_length) {
+ ast_debug(1, "%p -- RTCP from %s: %u(%s) length field less than expected minimum. Min:%u Got:%u\n",
+ transport_rtp, ast_sockaddr_stringify(addr), pt, rtcp_payload_type2str(pt),
+ min_length - 1, length - 1);
return &ast_null_frame;
}
- rtcp_report->reception_report_count = rc;
- rtcp_report->ssrc = ntohl(rtcpheader[i + 1]);
- if ((i + length) > packetwords) {
- if (rtpdebug) {
- ast_debug(1, "RTCP Read too short\n");
+ /* Get the RTCP record SSRC if defined for the record */
+ ssrc_valid = 1;
+ switch (pt) {
+ case RTCP_PT_SR:
+ case RTCP_PT_RR:
+ rtcp_report = ast_rtp_rtcp_report_alloc(rc);
+ if (!rtcp_report) {
+ return &ast_null_frame;
}
- return &ast_null_frame;
+ rtcp_report->reception_report_count = rc;
+
+ ssrc = ntohl(rtcpheader[i + 1]);
+ rtcp_report->ssrc = ssrc;
+ break;
+ case RTCP_PT_FUR:
+ case RTCP_PT_PSFB:
+ ssrc = ntohl(rtcpheader[i + 1]);
+ break;
+ case RTCP_PT_SDES:
+ case RTCP_PT_BYE:
+ default:
+ ssrc = 0;
+ ssrc_valid = 0;
+ break;
}
if (rtcp_debug_test_addr(addr)) {
- ast_verbose("\n\nGot RTCP from %s\n",
- ast_sockaddr_stringify(addr));
- ast_verbose("PT: %d(%s)\n", pt, (pt == RTCP_PT_SR) ? "Sender Report" :
- (pt == RTCP_PT_RR) ? "Receiver Report" :
- (pt == RTCP_PT_FUR) ? "H.261 FUR" : "Unknown");
- ast_verbose("Reception reports: %d\n", rc);
- ast_verbose("SSRC of sender: %u\n", rtcp_report->ssrc);
+ ast_verbose("\n");
+ ast_verbose("RTCP from %s\n", ast_sockaddr_stringify(addr));
+ ast_verbose("PT: %u(%s)\n", pt, rtcp_payload_type2str(pt));
+ ast_verbose("Reception reports: %u\n", rc);
+ ast_verbose("SSRC of sender: %u\n", ssrc);
}
/* Determine the appropriate instance for this */
- child = rtp_find_instance_by_ssrc(transport, transport_rtp, rtcp_report->ssrc);
- if (child != transport) {
- /* It is safe to hold the child lock while holding the parent lock, we guarantee that the locking order
- * is always parent->child or that the child lock is not held when acquiring the parent lock.
- */
- ao2_lock(child);
- instance = child;
- rtp = ast_rtp_instance_get_data(instance);
+ if (ssrc_valid) {
+ child = rtp_find_instance_by_ssrc(transport, transport_rtp, ssrc);
+ if (child != transport) {
+ /*
+ * It is safe to hold the child lock while holding the parent lock.
+ * We guarantee that the locking order is always parent->child or
+ * that the child lock is not held when acquiring the parent lock.
+ */
+ ao2_lock(child);
+ instance = child;
+ rtp = ast_rtp_instance_get_data(instance);
+ } else {
+ /* The child is the parent! We don't need to unlock it. */
+ child = NULL;
+ rtp = transport_rtp;
+ }
} else {
- /* The child is the parent! We don't need to unlock it. */
child = NULL;
rtp = transport_rtp;
}
- if ((rtp->strict_rtp_state != STRICT_RTP_OPEN) && (rtcp_report->ssrc != rtp->themssrc)) {
- /* Skip over this RTCP record as it does not contain the correct SSRC */
- position += (length + 1);
- ast_debug(1, "%p -- Received RTCP report from %s, dropping due to strict RTP protection. Received SSRC '%u' but expected '%u'\n",
- rtp, ast_sockaddr_stringify(addr), rtcp_report->ssrc, rtp->themssrc);
- continue;
+ if (ssrc_valid && rtp->themssrc_valid) {
+ if (ssrc != rtp->themssrc) {
+ /*
+ * Skip over this RTCP record as it does not contain the
+ * correct SSRC. We should not act upon RTCP records
+ * for a different stream.
+ */
+ position += length;
+ ast_debug(1, "%p -- RTCP from %s: Skipping record, received SSRC '%u' != expected '%u'\n",
+ rtp, ast_sockaddr_stringify(addr), ssrc, rtp->themssrc);
+ if (child) {
+ ao2_unlock(child);
+ }
+ continue;
+ }
+ ssrc_seen = 1;
}
- if (ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_NAT)) {
+ if (ssrc_seen && ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_NAT)) {
/* Send to whoever sent to us */
if (ast_sockaddr_cmp(&rtp->rtcp->them, addr)) {
ast_sockaddr_copy(&rtp->rtcp->them, addr);
if (rtpdebug) {
ast_debug(0, "RTCP NAT: Got RTCP from other end. Now sending to address %s\n",
- ast_sockaddr_stringify(&rtp->rtcp->them));
+ ast_sockaddr_stringify(addr));
}
}
}
- i += 2; /* Advance past header and ssrc */
+ i += RTCP_HEADER_SSRC_LENGTH; /* Advance past header and ssrc */
switch (pt) {
case RTCP_PT_SR:
gettimeofday(&rtp->rtcp->rxlsr, NULL);
@@ -4888,7 +5112,7 @@
rtcp_report->sender_information.packet_count,
rtcp_report->sender_information.octet_count);
}
- i += 5;
+ i += RTCP_SR_BLOCK_WORD_LENGTH;
/* Intentional fall through */
case RTCP_PT_RR:
if (rtcp_report->type != RTCP_PT_SR) {
@@ -4948,9 +5172,9 @@
*/
message_blob = ast_json_pack("{s: s, s: s, s: f}",
- "from", ast_sockaddr_stringify(&transport_rtp->rtcp->them),
- "to", transport_rtp->rtcp->local_addr_str,
- "rtt", rtp->rtcp->rtt);
+ "from", ast_sockaddr_stringify(addr),
+ "to", transport_rtp->rtcp->local_addr_str,
+ "rtt", rtp->rtcp->rtt);
ast_rtp_publish_rtcp_message(instance, ast_rtp_rtcp_received_type(),
rtcp_report,
message_blob);
@@ -4996,21 +5220,19 @@
case RTCP_PT_SDES:
if (rtcp_debug_test_addr(addr)) {
ast_verbose("Received an SDES from %s\n",
- ast_sockaddr_stringify(&transport_rtp->rtcp->them));
+ ast_sockaddr_stringify(addr));
}
break;
case RTCP_PT_BYE:
if (rtcp_debug_test_addr(addr)) {
ast_verbose("Received a BYE from %s\n",
- ast_sockaddr_stringify(&transport_rtp->rtcp->them));
+ ast_sockaddr_stringify(addr));
}
break;
default:
- ast_debug(1, "Unknown RTCP packet (pt=%d) received from %s\n",
- pt, ast_sockaddr_stringify(&transport_rtp->rtcp->them));
break;
}
- position += (length + 1);
+ position += length;
rtp->rtcp->rtcp_info = 1;
if (child) {
@@ -5019,7 +5241,6 @@
}
return f;
-
}
/*! \pre instance is locked */
@@ -5343,31 +5564,133 @@
}
/* If strict RTP protection is enabled see if we need to learn the remote address or if we need to drop the packet */
- if (rtp->strict_rtp_state == STRICT_RTP_LEARN) {
- if (!ast_sockaddr_cmp(&rtp->strict_rtp_address, &addr)) {
- /* We are learning a new address but have received traffic from the existing address,
- * accept it but reset the current learning for the new source so it only takes over
- * once sufficient traffic has been received. */
- rtp_learning_seq_init(&rtp->rtp_source_learn, seqno);
- } else {
- /* Start trying to learn from the new address. If we pass a probationary period with
- * it, that means we've stopped getting RTP from the original source and we should
- * switch to it.
- */
- if (rtp_learning_rtp_seq_update(&rtp->rtp_source_learn, seqno)) {
- ast_debug(1, "%p -- Received RTP packet from %s, dropping due to strict RTP protection. Will switch to it in %d packets\n",
- rtp, ast_sockaddr_stringify(&addr), rtp->rtp_source_learn.packets);
- return &ast_null_frame;
- }
- ast_sockaddr_copy(&rtp->strict_rtp_address, &addr);
-
- ast_verb(4, "%p -- Probation passed - setting RTP source address to %s\n", rtp, ast_sockaddr_stringify(&addr));
+ switch (rtp->strict_rtp_state) {
+ case STRICT_RTP_LEARN:
+ /*
+ * Scenario setup:
+ * PartyA -- Ast1 -- Ast2 -- PartyB
+ *
+ * The learning timeout is necessary for Ast1 to handle the above
+ * setup where PartyA calls PartyB and Ast2 initiates direct media
+ * between Ast1 and PartyB. Ast1 may lock onto the Ast2 stream and
+ * never learn the PartyB stream when it starts. The timeout makes
+ * Ast1 stay in the learning state long enough to see and learn the
+ * RTP stream from PartyB.
+ *
+ * To mitigate against attack, the learning state cannot switch
+ * streams while there are competing streams. The competing streams
+ * interfere with each other's qualification. Once we accept a
+ * stream and reach the timeout, an attacker cannot interfere
+ * anymore.
+ *
+ * Here are a few scenarios and each one assumes that the streams
+ * are continuous:
+ *
+ * 1) We already have a known stream source address and the known
+ * stream wants to change to a new source address. An attacking
+ * stream will block learning the new stream source. After the
+ * timeout we re-lock onto the original stream source address which
+ * likely went away. The result is one way audio.
+ *
+ * 2) We already have a known stream source address and the known
+ * stream doesn't want to change source addresses. An attacking
+ * stream will not be able to replace the known stream. After the
+ * timeout we re-lock onto the known stream. The call is not
+ * affected.
+ *
+ * 3) We don't have a known stream source address. This presumably
+ * is the start of a call. Competing streams will result in staying
+ * in learning mode until a stream becomes the victor and we reach
+ * the timeout. We cannot exit learning if we have no known stream
+ * to lock onto. The result is one way audio until there is a victor.
+ *
+ * If we learn a stream source address before the timeout we will be
+ * in scenario 1) or 2) when a competing stream starts.
+ */
+ if (!ast_sockaddr_isnull(&rtp->strict_rtp_address)
+ && STRICT_RTP_LEARN_TIMEOUT < ast_tvdiff_ms(ast_tvnow(), rtp->rtp_source_learn.start)) {
+ ast_verb(4, "%p -- Strict RTP learning complete - Locking on source address %s\n",
+ rtp, ast_sockaddr_stringify(&rtp->strict_rtp_address));
rtp->strict_rtp_state = STRICT_RTP_CLOSED;
+ } else {
+ struct ast_sockaddr target_address;
+
+ if (!ast_sockaddr_cmp(&rtp->strict_rtp_address, &addr)) {
+ /*
+ * We are open to learning a new address but have received
+ * traffic from the current address, accept it and reset
+ * the learning counts for a new source. When no more
+ * current source packets arrive a new source can take over
+ * once sufficient traffic is received.
+ */
+ rtp_learning_seq_init(&rtp->rtp_source_learn, seqno);
+ break;
+ }
+
+ /*
+ * We give preferential treatment to the requested target address
+ * (negotiated SDP address) where we are to send our RTP. However,
+ * the other end has no obligation to send from that address even
+ * though it is practically a requirement when NAT is involved.
+ */
+ ast_rtp_instance_get_requested_target_address(instance, &target_address);
+ if (!ast_sockaddr_cmp(&target_address, &addr)) {
+ /* Accept the negotiated target RTP stream as the source */
+ ast_verb(4, "%p -- Strict RTP switching to RTP target address %s as source\n",
+ rtp, ast_sockaddr_stringify(&addr));
+ ast_sockaddr_copy(&rtp->strict_rtp_address, &addr);
+ rtp_learning_seq_init(&rtp->rtp_source_learn, seqno);
+ break;
+ }
+
+ /*
+ * Trying to learn a new address. If we pass a probationary period
+ * with it, that means we've stopped getting RTP from the original
+ * source and we should switch to it.
+ */
+ if (!ast_sockaddr_cmp(&rtp->rtp_source_learn.proposed_address, &addr)) {
+ if (!rtp_learning_rtp_seq_update(&rtp->rtp_source_learn, seqno)) {
+ /* Accept the new RTP stream */
+ ast_verb(4, "%p -- Strict RTP switching source address to %s\n",
+ rtp, ast_sockaddr_stringify(&addr));
+ ast_sockaddr_copy(&rtp->strict_rtp_address, &addr);
+ rtp_learning_seq_init(&rtp->rtp_source_learn, seqno);
+ break;
+ }
+ /* Not ready to accept the RTP stream candidate */
+ ast_debug(1, "%p -- Received RTP packet from %s, dropping due to strict RTP protection. Will switch to it in %d packets.\n",
+ rtp, ast_sockaddr_stringify(&addr), rtp->rtp_source_learn.packets);
+ } else {
+ /*
+ * This is either an attacking stream or
+ * the start of the expected new stream.
+ */
+ ast_sockaddr_copy(&rtp->rtp_source_learn.proposed_address, &addr);
+ rtp_learning_seq_init(&rtp->rtp_source_learn, seqno);
+ ast_debug(1, "%p -- Received RTP packet from %s, dropping due to strict RTP protection. Qualifying new stream.\n",
+ rtp, ast_sockaddr_stringify(&addr));
+ }
+ return &ast_null_frame;
}
- } else if (rtp->strict_rtp_state == STRICT_RTP_CLOSED && ast_sockaddr_cmp(&rtp->strict_rtp_address, &addr)) {
+ /* Fall through */
+ case STRICT_RTP_CLOSED:
+ /*
+ * We should not allow a stream address change if the SSRC matches
+ * once strictrtp learning is closed. Any kind of address change
+ * like this should have happened while we were in the learning
+ * state. We do not want to allow the possibility of an attacker
+ * interfering with the RTP stream after the learning period.
+ * An attacker could manage to get an RTCP packet redirected to
+ * them which can contain the SSRC value.
+ */
+ if (!ast_sockaddr_cmp(&rtp->strict_rtp_address, &addr)) {
+ break;
+ }
ast_debug(1, "%p -- Received RTP packet from %s, dropping due to strict RTP protection.\n",
rtp, ast_sockaddr_stringify(&addr));
return &ast_null_frame;
+ case STRICT_RTP_OPEN:
+ break;
}
/* If symmetric RTP is enabled see if the remote side is not what we expected and change where we are sending audio */
@@ -5404,7 +5727,7 @@
AST_LIST_HEAD_INIT_NOLOCK(&frames);
/* Force a marker bit and change SSRC if the SSRC changes */
- if (rtp->rxssrc && rtp->rxssrc != ssrc) {
+ if (rtp->themssrc_valid && rtp->themssrc != ssrc) {
struct ast_frame *f, srcupdate = {
AST_FRAME_CONTROL,
.subclass.integer = AST_CONTROL_SRCCHANGE,
@@ -5432,8 +5755,10 @@
rtp->rtcp->received_prior = 0;
}
}
-
- rtp->rxssrc = ssrc;
+ /* Bundled children cannot change/learn their SSRC implicitly. */
+ ast_assert(!child || (rtp->themssrc_valid && rtp->themssrc == ssrc));
+ rtp->themssrc = ssrc; /* Record their SSRC to put in future RR */
+ rtp->themssrc_valid = 1;
/* Remove any padding bytes that may be present */
if (padding) {
@@ -5486,10 +5811,6 @@
prev_seqno = rtp->lastrxseqno;
rtp->lastrxseqno = seqno;
-
- if (!rtp->themssrc) {
- rtp->themssrc = ntohl(rtpheader[2]); /* Record their SSRC to put in future RR */
- }
/* If we are directly bridged to another instance send the audio directly out,
@@ -5899,13 +6220,14 @@
rtp->rxseqno = 0;
- if (strictrtp && rtp->strict_rtp_state != STRICT_RTP_OPEN && !ast_sockaddr_isnull(addr) &&
- ast_sockaddr_cmp(addr, &rtp->strict_rtp_address)) {
+ if (strictrtp && rtp->strict_rtp_state != STRICT_RTP_OPEN
+ && !ast_sockaddr_isnull(addr) && ast_sockaddr_cmp(addr, &rtp->strict_rtp_address)) {
/* We only need to learn a new strict source address if we've been told the source is
* changing to something different.
*/
- rtp->strict_rtp_state = STRICT_RTP_LEARN;
- rtp_learning_seq_init(&rtp->rtp_source_learn, rtp->seqno);
+ ast_verb(4, "%p -- Strict RTP learning after remote address set to: %s\n",
+ rtp, ast_sockaddr_stringify(addr));
+ rtp_learning_start(rtp);
}
}
@@ -6189,11 +6511,12 @@
return rtp->cname;
}
+/*! \pre instance is locked */
static void ast_rtp_set_remote_ssrc(struct ast_rtp_instance *instance, unsigned int ssrc)
{
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
- if (rtp->themssrc == ssrc) {
+ if (rtp->themssrc_valid && rtp->themssrc == ssrc) {
return;
}
@@ -6201,6 +6524,8 @@
if (rtp->bundled) {
struct ast_rtp *bundled_rtp;
int index;
+
+ ast_assert(rtp->themssrc_valid);
ao2_unlock(instance);
@@ -6223,6 +6548,7 @@
}
rtp->themssrc = ssrc;
+ rtp->themssrc_valid = 1;
}
static void ast_rtp_set_stream_num(struct ast_rtp_instance *instance, int stream_num)
@@ -6232,6 +6558,7 @@
rtp->stream_num = stream_num;
}
+/*! \pre child is locked */
static int ast_rtp_bundle(struct ast_rtp_instance *child, struct ast_rtp_instance *parent)
{
struct ast_rtp *child_rtp = ast_rtp_instance_get_data(child);
@@ -6274,6 +6601,7 @@
/* Children maintain a reference to the parent to guarantee that the transport doesn't go away on them */
child_rtp->bundled = ao2_bump(parent);
+ ast_assert(child_rtp->themssrc_valid);
mapping.ssrc = child_rtp->themssrc;
mapping.instance = child;
--
To view, visit https://gerrit.asterisk.org/6414
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-MessageType: merged
Gerrit-Change-Id: Ia2d3aa6e0f22906c25971e74f10027d96525f31c
Gerrit-Change-Number: 6414
Gerrit-PatchSet: 2
Gerrit-Owner: Richard Mudgett <rmudgett at digium.com>
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Jenkins2
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Richard Mudgett <rmudgett at digium.com>
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