summaryrefslogtreecommitdiff
diff options
context:
space:
mode:
authorJoshua Colp <jcolp@digium.com>2017-09-19 10:56:44 -0500
committerGerrit Code Review <gerrit2@gerrit.digium.api>2017-09-19 10:56:44 -0500
commit3274c13b126edc0d846036ab475ec53122f5a702 (patch)
tree157ed3e0eded3609e891311ed13788f4d6015c2e
parent864e76374c0220dc857998a396b22846e169e1a8 (diff)
parent4a57e20d52726c03088578d4d1431e566deffff5 (diff)
Merge "AST-2017-008: Improve RTP and RTCP packet processing." into 14
-rw-r--r--res/res_rtp_asterisk.c498
1 files changed, 404 insertions, 94 deletions
diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index 2d43495df..fcad130fa 100644
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -124,7 +124,9 @@ enum strict_rtp_state {
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;
@@ -225,9 +227,11 @@ static AST_RWLIST_HEAD_STATIC(host_candidates, ast_ice_host_candidate);
/*! \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
@@ -256,7 +260,7 @@ struct ast_rtp {
unsigned char rawdata[8192 + AST_FRIENDLY_OFFSET];
unsigned int ssrc; /*!< Synchronization source, RFC 3550, page 10. */
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;
@@ -1991,7 +1995,7 @@ static void dtls_perform_handshake(struct ast_rtp_instance *instance, struct dtl
#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)
@@ -2030,8 +2034,8 @@ static void ast_rtp_on_ice_complete(pj_ice_sess *ice, pj_status_t status)
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);
}
@@ -2760,7 +2764,7 @@ static int create_new_socket(const char *type, int af)
*/
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));
}
@@ -2777,14 +2781,17 @@ static void rtp_learning_seq_init(struct rtp_learning_info *info, uint16_t seq)
*/
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 {
@@ -2794,7 +2801,23 @@ static int rtp_learning_rtp_seq_update(struct rtp_learning_info *info, uint16_t
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
@@ -3068,9 +3091,6 @@ static int ast_rtp_new(struct ast_rtp_instance *instance,
rtp->ssrc = ast_random();
rtp->seqno = ast_random() & 0x7fff;
rtp->strict_rtp_state = (strictrtp ? STRICT_RTP_CLOSED : STRICT_RTP_OPEN);
- if (strictrtp) {
- rtp_learning_seq_init(&rtp->rtp_source_learn, (uint16_t)rtp->seqno);
- }
/* Create a new socket for us to listen on and use */
if ((rtp->s =
@@ -3644,7 +3664,7 @@ static int ast_rtcp_write_report(struct ast_rtp_instance *instance, int sr)
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) {
@@ -3664,7 +3684,7 @@ static int ast_rtcp_write_report(struct ast_rtp_instance *instance, int sr)
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) {
@@ -3674,7 +3694,7 @@ static int ast_rtcp_write_report(struct ast_rtp_instance *instance, int sr)
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;
@@ -4019,6 +4039,10 @@ static int ast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *fr
*/
return 0;
}
+ if (!rtp->themssrc_valid) {
+ /* We don't know their SSRC value so we don't know who to update. */
+ return 0;
+ }
/* Prepare RTCP FIR (PT=206, FMT=4) */
rtp->rtcp->firseq++;
@@ -4597,75 +4621,265 @@ static void update_lost_stats(struct ast_rtp *rtp, unsigned int lost_packets)
rtp->rtcp->reported_normdev_lost = reported_normdev_lost_current;
}
+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 *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",
+ 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",
+ 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",
+ 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 = 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);
i = position;
- length = ntohl(rtcpheader[i]);
- pt = (length & 0xff0000) >> 16;
- rc = (length & 0x1f000000) >> 24;
- length &= 0xffff;
-
- rtcp_report = ast_rtp_rtcp_report_alloc(rc);
- if (!rtcp_report) {
+ 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;
+
+ /* 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",
+ 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",
+ 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 ((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 (rtcp_debug_test_addr(addr)) {
+ 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);
+ }
+
+ 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);
+ 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));
}
}
}
- 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);
- }
-
- 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);
@@ -4689,7 +4903,7 @@ static struct ast_frame *ast_rtcp_interpret(struct ast_rtp_instance *instance, c
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) {
@@ -4746,9 +4960,9 @@ static struct ast_frame *ast_rtcp_interpret(struct ast_rtp_instance *instance, c
*/
message_blob = ast_json_pack("{s: s, s: s, s: f}",
- "from", ast_sockaddr_stringify(&rtp->rtcp->them),
- "to", rtp->rtcp->local_addr_str,
- "rtt", rtp->rtcp->rtt);
+ "from", ast_sockaddr_stringify(addr),
+ "to", rtp->rtcp->local_addr_str,
+ "rtt", rtp->rtcp->rtt);
ast_rtp_publish_rtcp_message(instance, ast_rtp_rtcp_received_type(),
rtcp_report,
message_blob);
@@ -4771,26 +4985,23 @@ static struct ast_frame *ast_rtcp_interpret(struct ast_rtp_instance *instance, c
case RTCP_PT_SDES:
if (rtcp_debug_test_addr(addr)) {
ast_verbose("Received an SDES from %s\n",
- ast_sockaddr_stringify(&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(&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(&rtp->rtcp->them));
break;
}
- position += (length + 1);
+ position += length;
}
rtp->rtcp->rtcp_info = 1;
return f;
-
}
/*! \pre instance is locked */
@@ -5082,31 +5293,133 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
}
/* 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);
+ 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 {
- /* 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.
+ 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.
*/
- 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_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;
}
- 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));
- rtp->strict_rtp_state = STRICT_RTP_CLOSED;
+ /*
+ * 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;
+ }
+ /* 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;
}
- } else if (rtp->strict_rtp_state == STRICT_RTP_CLOSED && ast_sockaddr_cmp(&rtp->strict_rtp_address, &addr)) {
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 */
@@ -5139,7 +5452,7 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
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,
@@ -5167,8 +5480,8 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
rtp->rtcp->received_prior = 0;
}
}
-
- rtp->rxssrc = 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) {
@@ -5222,10 +5535,6 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
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,
* but only after updating core information about the received traffic so that
@@ -5624,13 +5933,14 @@ static void ast_rtp_remote_address_set(struct ast_rtp_instance *instance, struct
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);
}
}