rtp: Enable srtp replay protection

Add option "srtpreplayprotection" rtp.conf to enable srtp
replay protection.

ASTERISK-29260
Reported by: Alexander Traud

Change-Id: I5cd346e3c6b6812039d1901aa4b7be688173b458

diff --git a/configs/samples/rtp.conf.sample b/configs/samples/rtp.conf.sample
index a94707e59698db5dc3704ea75a81e807d49ffd01..c60167745a02fedaf51d411b8f0e4a47d57b4f02 100644 (file)
--- a/configs/samples/rtp.conf.sample
+++ b/configs/samples/rtp.conf.sample
@@ -45,6 +45,18 @@ rtpend=20000
 ; connected. This option is set to 4 by default.
 ; probation=8
 ;
+; Enable sRTP replay protection. Buggy SIP user agents (UAs) reset the
+; sequence number (RTP-SEQ) on a re-INVITE, for example, with Session Timers
+; or on Call Hold/Resume, but keep the synchronization source (RTP-SSRC). If
+; the new RTP-SEQ is higher than the previous one, the call continues if the
+; roll-over counter (sRTP-ROC) is zero (the call lasted less than 22 minutes).
+; In all other cases, the call faces one-way audio or even no audio at all.
+; "replay check failed (index too old)" gets printed continuously. This is a
+; software bug. You have to report this to the creator of that UA. Until it is
+; fixed, you could disable sRTP replay protection (see RFC 3711 section 3.3.2).
+; This option is enabled by default.
+; srtpreplayprotection=yes
+;
 ; Whether to enable or disable ICE support. This option is enabled by default.
 ; icesupport=false
 ;

diff --git a/doc/CHANGES-staging/srtp_replay_protection.txt b/doc/CHANGES-staging/srtp_replay_protection.txt
new file mode 100644 (file)
index 0000000..945ddb5
--- /dev/null
+++ b/doc/CHANGES-staging/srtp_replay_protection.txt
@@ -0,0 +1,9 @@
+Subject: res_srtp
+
+SRTP replay protection has been added to res_srtp and
+a new configuration option "srtpreplayprotection" has
+been added to the rtp.conf config file.  For security
+reasons, the default setting is "yes".  Buggy clients
+may not handle this correctly which could result in
+no, or one way, audio and Asterisk error messages like
+"replay check failed".

diff --git a/doc/UPGRADE-staging/srtp_replay_protection.txt b/doc/UPGRADE-staging/srtp_replay_protection.txt
new file mode 100644 (file)
index 0000000..945ddb5
--- /dev/null
+++ b/doc/UPGRADE-staging/srtp_replay_protection.txt
@@ -0,0 +1,9 @@
+Subject: res_srtp
+
+SRTP replay protection has been added to res_srtp and
+a new configuration option "srtpreplayprotection" has
+been added to the rtp.conf config file.  For security
+reasons, the default setting is "yes".  Buggy clients
+may not handle this correctly which could result in
+no, or one way, audio and Asterisk error messages like
+"replay check failed".

diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index aed7a891b1284bd7f4668b53bc60434d55950924..0f428b193652bf330e49cc9b7c3ff587e8668022 100644 (file)
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -178,6 +178,7 @@ enum strict_rtp_mode {
 #define STRICT_RTP_LEARN_TIMEOUT       5000
 
 #define DEFAULT_STRICT_RTP STRICT_RTP_YES      /*!< Enabled by default */
+#define DEFAULT_SRTP_REPLAY_PROTECTION 1
 #define DEFAULT_ICESUPPORT 1
 #define DEFAULT_DTLS_MTU 1200
 
@@ -202,6 +203,7 @@ static int nochecksums;
 static int strictrtp = DEFAULT_STRICT_RTP; /*!< Only accept RTP frames from a defined source. If we receive an indication of a changing source, enter learning mode. */
 static int learning_min_sequential = DEFAULT_LEARNING_MIN_SEQUENTIAL; /*!< Number of sequential RTP frames needed from a single source during learning mode to accept new source. */
 static int learning_min_duration = DEFAULT_LEARNING_MIN_DURATION; /*!< Lowest acceptable timeout between the first and the last sequential RTP frame. */
+static int srtp_replay_protection = DEFAULT_SRTP_REPLAY_PROTECTION;
 #if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
 static int dtls_mtu = DEFAULT_DTLS_MTU;
 #endif
@@ -5927,7 +5929,7 @@ static struct ast_frame *ast_rtcp_interpret(struct ast_rtp_instance *instance, s
 
        /* If this is encrypted then decrypt the payload */
        if ((*rtcpheader & 0xC0) && res_srtp && srtp && res_srtp->unprotect(
-                   srtp, rtcpheader, &len, 1) < 0) {
+                   srtp, rtcpheader, &len, 1 | (srtp_replay_protection << 1)) < 0) {
           return &ast_null_frame;
        }
 
@@ -7009,7 +7011,7 @@ static struct ast_frame *ast_rtp_interpret(struct ast_rtp_instance *instance, st
 
        /* If this payload is encrypted then decrypt it using the given SRTP instance */
        if ((*read_area & 0xC0) && res_srtp && srtp && res_srtp->unprotect(
-                   srtp, read_area, &res, 0) < 0) {
+                   srtp, read_area, &res, 0 | (srtp_replay_protection << 1)) < 0) {
                return &ast_null_frame;
        }
 
@@ -8919,6 +8921,7 @@ static int rtp_reload(int reload)
        strictrtp = DEFAULT_STRICT_RTP;
        learning_min_sequential = DEFAULT_LEARNING_MIN_SEQUENTIAL;
        learning_min_duration = DEFAULT_LEARNING_MIN_DURATION;
+       srtp_replay_protection = DEFAULT_SRTP_REPLAY_PROTECTION;
 
        /** This resource is not "reloaded" so much as unloaded and loaded again.
         * In the case of the TURN related variables, the memory referenced by a
@@ -9000,6 +9003,9 @@ static int rtp_reload(int reload)
                }
                learning_min_duration = CALC_LEARNING_MIN_DURATION(learning_min_sequential);
        }
+       if ((s = ast_variable_retrieve(cfg, "general", "srtpreplayprotection"))) {
+               srtp_replay_protection = ast_true(s);
+       }
 #ifdef HAVE_PJPROJECT
        if ((s = ast_variable_retrieve(cfg, "general", "icesupport"))) {
                icesupport = ast_true(s);

diff --git a/res/res_srtp.c b/res/res_srtp.c
index 3e4b51fafb42b396b7cb216652edb832446e9d05..7098621c783c3c57c66fec945daaf32910d79757 100644 (file)
--- a/res/res_srtp.c
+++ b/res/res_srtp.c
@@ -364,11 +364,12 @@ static void ast_srtp_set_cb(struct ast_srtp *srtp, const struct ast_srtp_cb *cb,
 }
 
 /* Vtable functions */
-static int ast_srtp_unprotect(struct ast_srtp *srtp, void *buf, int *len, int rtcp)
+static int ast_srtp_unprotect(struct ast_srtp *srtp, void *buf, int *len, int flags)
 {
        int res = 0;
        int i;
-       int retry = 0;
+       int rtcp  = (flags & 0x01) >> 0;
+       int retry = (flags & 0x02) >> 1;
        struct ast_rtp_instance_stats stats = {0,};
 
 tryagain: