From 9c9f8feba374fc9b8407aadfbce6ab065ff0a016 Mon Sep 17 00:00:00 2001
From: Alexander Traud <pabstraud@compuserve.com>
Date: Tue, 26 Jan 2021 10:09:53 -0700
Subject: [PATCH] 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 f002449..7ab81b3 100644
--- a/configs/samples/rtp.conf.sample
+++ b/configs/samples/rtp.conf.sample
@@ -45,6 +45,18 @@
 ; 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
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
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 167a766..3e8fa1f 100644
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -166,6 +166,7 @@
 #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
 
@@ -190,6 +191,7 @@
 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
@@ -2715,7 +2717,9 @@
 #endif
 
 	if ((*in & 0xC0) && res_srtp && srtp && res_srtp->unprotect(
-		    srtp, buf, &len, rtcp || rtcp_mux(rtp, buf)) < 0) {
+		    srtp, buf, &len,
+		    (rtcp || rtcp_mux(rtp, buf)) | (srtp_replay_protection << 1)
+		) < 0) {
 	   return -1;
 	}
 
@@ -6678,6 +6682,8 @@
 	if (strictrtp) {
 		ast_cli(a->fd, "  Probation:       %d frames\n", learning_min_sequential);
 	}
+
+	ast_cli(a->fd, "  Replay Protect:  %s\n", AST_CLI_YESNO(srtp_replay_protection));
 #ifdef HAVE_PJPROJECT
 	ast_cli(a->fd, "  ICE support:     %s\n", AST_CLI_YESNO(icesupport));
 #endif
@@ -6780,6 +6786,7 @@
 	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
@@ -6859,6 +6866,9 @@
 		}
 		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 a2a8452..cab4585 100644
--- a/res/res_srtp.c
+++ b/res/res_srtp.c
@@ -336,11 +336,12 @@
 }
 
 /* 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:
