Merge "res_rtp_asterisk.c: Disable packet flood detection for video streams." into 15

4 files changed, 74 insertions, 14 deletions

diff --git a/configs/samples/rtp.conf.sample b/configs/samples/rtp.conf.sample
index 9bc3de3cf..de9d59007 100644
--- a/configs/samples/rtp.conf.sample
+++ b/configs/samples/rtp.conf.sample
@@ -21,9 +21,17 @@ rtpend=20000
 ; rtcpinterval = 5000 	; Milliseconds between rtcp reports
 			;(min 500, max 60000, default 5000)
 ;
-; Enable strict RTP protection. This will drop RTP packets that
-; do not come from the source of the RTP stream. This option is
-; enabled by default.
+; Enable strict RTP protection.  This will drop RTP packets that do not come
+; from the recoginized source of the RTP stream.  Strict RTP qualifies RTP
+; packet stream sources before accepting them upon initial connection and
+; when the connection is renegotiated (e.g., transfers and direct media).
+; Initial connection and renegotiation starts a learning mode to qualify
+; stream source addresses.  Once Asterisk has recognized a stream it will
+; allow other streams to qualify and replace the current stream for 5
+; seconds after starting learning mode.  Once learning mode completes the
+; current stream is locked in and cannot change until the next
+; renegotiation.
+; This option is enabled by default.
 ; strictrtp=yes
 ;
 ; Number of packets containing consecutive sequence values needed
diff --git a/include/asterisk/rtp_engine.h b/include/asterisk/rtp_engine.h
index f9d686aca..c77be4584 100644
--- a/include/asterisk/rtp_engine.h
+++ b/include/asterisk/rtp_engine.h
@@ -1383,6 +1383,16 @@ int ast_rtp_codecs_payloads_set_rtpmap_type_rate(struct ast_rtp_codecs *codecs,
 void ast_rtp_codecs_payloads_unset(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int payload);
 
 /*!
+ * \brief Determine the type of RTP stream media from the codecs mapped.
+ * \since 13.19.0
+ *
+ * \param codecs Codecs structure to look in
+ *
+ * \return Media type or AST_MEDIA_TYPE_UNKNOWN if no codecs mapped.
+ */
+enum ast_media_type ast_rtp_codecs_get_stream_type(struct ast_rtp_codecs *codecs);
+
+/*!
  * \brief Retrieve rx payload mapped information by payload type
  *
  * \param codecs Codecs structure to look in
diff --git a/main/rtp_engine.c b/main/rtp_engine.c
index 2431ffc0c..68c53e7ff 100644
--- a/main/rtp_engine.c
+++ b/main/rtp_engine.c
@@ -1176,6 +1176,25 @@ void ast_rtp_codecs_payloads_unset(struct ast_rtp_codecs *codecs, struct ast_rtp
 	ast_rwlock_unlock(&codecs->codecs_lock);
 }
 
+enum ast_media_type ast_rtp_codecs_get_stream_type(struct ast_rtp_codecs *codecs)
+{
+	enum ast_media_type stream_type = AST_MEDIA_TYPE_UNKNOWN;
+	int payload;
+	struct ast_rtp_payload_type *type;
+
+	ast_rwlock_rdlock(&codecs->codecs_lock);
+	for (payload = 0; payload < AST_VECTOR_SIZE(&codecs->payload_mapping_rx); ++payload) {
+		type = AST_VECTOR_GET(&codecs->payload_mapping_rx, payload);
+		if (type && type->asterisk_format) {
+			stream_type = ast_format_get_type(type->format);
+			break;
+		}
+	}
+	ast_rwlock_unlock(&codecs->codecs_lock);
+
+	return stream_type;
+}
+
 struct ast_rtp_payload_type *ast_rtp_codecs_get_payload(struct ast_rtp_codecs *codecs, int payload)
 {
 	struct ast_rtp_payload_type *type = NULL;
diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index bdc83301e..51e509c77 100644
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -256,6 +256,8 @@ struct rtp_learning_info {
 	struct timeval received; /*!< The time of the first received packet */
 	int max_seq;	/*!< The highest sequence number received */
 	int packets;	/*!< The number of remaining packets before the source is accepted */
+	/*! Type of media stream carried by the RTP instance */
+	enum ast_media_type stream_type;
 };
 
 #ifdef HAVE_OPENSSL_SRTP
@@ -3095,18 +3097,30 @@ static int rtp_learning_rtp_seq_update(struct rtp_learning_info *info, uint16_t
 		info->received = ast_tvnow();
 	}
 
-	/*
-	 * Protect against packet floods by checking that we
-	 * received the packet sequence in at least the minimum
-	 * allowed time.
-	 */
-	if (ast_tvzero(info->received)) {
-		info->received = ast_tvnow();
-	} else if (!info->packets && (ast_tvdiff_ms(ast_tvnow(), info->received) < learning_min_duration )) {
-		/* Packet flood; reset */
-		info->packets = learning_min_sequential - 1;
-		info->received = ast_tvnow();
+	switch (info->stream_type) {
+	case AST_MEDIA_TYPE_UNKNOWN:
+	case AST_MEDIA_TYPE_AUDIO:
+		/*
+		 * Protect against packet floods by checking that we
+		 * received the packet sequence in at least the minimum
+		 * allowed time.
+		 */
+		if (ast_tvzero(info->received)) {
+			info->received = ast_tvnow();
+		} else if (!info->packets
+			&& ast_tvdiff_ms(ast_tvnow(), info->received) < learning_min_duration) {
+			/* Packet flood; reset */
+			info->packets = learning_min_sequential - 1;
+			info->received = ast_tvnow();
+		}
+		break;
+	case AST_MEDIA_TYPE_VIDEO:
+	case AST_MEDIA_TYPE_IMAGE:
+	case AST_MEDIA_TYPE_TEXT:
+	case AST_MEDIA_TYPE_END:
+		break;
 	}
+
 	info->max_seq = seq;
 
 	return info->packets;
@@ -5951,6 +5965,15 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
 			 * source and we should switch to it.
 			 */
 			if (!ast_sockaddr_cmp(&rtp->rtp_source_learn.proposed_address, &addr)) {
+				if (rtp->rtp_source_learn.stream_type == AST_MEDIA_TYPE_UNKNOWN) {
+					struct ast_rtp_codecs *codecs;
+
+					codecs = ast_rtp_instance_get_codecs(instance);
+					rtp->rtp_source_learn.stream_type =
+						ast_rtp_codecs_get_stream_type(codecs);
+					ast_verb(4, "%p -- Strict RTP qualifying stream type: %s\n",
+						rtp, ast_codec_media_type2str(rtp->rtp_source_learn.stream_type));
+				}
 				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",