mediasoup 3.16.0 → 3.16.2

package/package.json

@@ -1,6 +1,6 @@
 {
 	"name": "mediasoup",
-	"version": "3.16.0",
+	"version": "3.16.2",
 	"description": "Cutting Edge WebRTC Video Conferencing",
 	"contributors": [
 		"Iñaki Baz Castillo <ibc@aliax.net> (https://inakibaz.me)",
@@ -100,24 +100,24 @@
 		"tar": "^7.4.3"
 	},
 	"devDependencies": {
-		"@eslint/js": "^9.27.0",
+		"@eslint/js": "^9.29.0",
 		"@octokit/rest": "^22.0.0",
 		"@types/debug": "^4.1.12",
-		"@types/jest": "^29.5.14",
-		"@types/node": "^22.15.21",
-		"eslint": "^9.27.0",
+		"@types/jest": "^30.0.0",
+		"@types/node": "^24.0.4",
+		"eslint": "^9.29.0",
 		"eslint-config-prettier": "^10.1.5",
-		"eslint-plugin-jest": "^28.11.0",
-		"eslint-plugin-prettier": "^5.4.0",
+		"eslint-plugin-jest": "^29.0.1",
+		"eslint-plugin-prettier": "^5.5.1",
 		"globals": "^16.2.0",
-		"jest": "^29.7.0",
-		"marked": "^15.0.12",
+		"jest": "^30.0.3",
+		"marked": "^16.0.0",
 		"open-cli": "^8.0.0",
 		"pick-port": "^2.1.0",
-		"prettier": "^3.5.3",
+		"prettier": "^3.6.2",
 		"sctp": "^1.0.0",
-		"ts-jest": "^29.3.4",
+		"ts-jest": "^29.4.0",
 		"typescript": "^5.8.3",
-		"typescript-eslint": "^8.32.1"
+		"typescript-eslint": "^8.35.0"
 	}
 }

package/worker/include/Logger.hpp

@@ -460,6 +460,7 @@ public:
 	} \
 	while (false)
 
+#ifdef MS_EXECUTABLE
 #define MS_ABORT(desc, ...) \
 	do \
 	{ \
@@ -468,6 +469,18 @@ public:
 		std::abort(); \
 	} \
 	while (false)
+#else
+#define MS_ABORT(desc, ...) \
+	do \
+	{ \
+		std::fprintf(stderr, "(ABORT) " _MS_LOG_STR_DESC desc _MS_LOG_SEPARATOR_CHAR_STD, _MS_LOG_ARG, ##__VA_ARGS__); \
+		std::fflush(stderr); \
+		char abortMessage[Logger::BufferSize]; \
+		std::snprintf(abortMessage, Logger::BufferSize, "(ABORT) " _MS_LOG_STR_DESC desc _MS_LOG_SEPARATOR_CHAR_STD, _MS_LOG_ARG, ##__VA_ARGS__); \
+		throw std::runtime_error(abortMessage); \
+	} \
+	while (false)
+#endif
 
 #define MS_ASSERT(condition, desc, ...) \
 	if (!(condition)) \

package/worker/include/RTC/PipeConsumer.hpp

@@ -4,6 +4,7 @@
 #include "RTC/Consumer.hpp"
 #include "RTC/SeqManager.hpp"
 #include "RTC/Shared.hpp"
+#include <map>
 
 namespace RTC
 {
@@ -59,6 +60,11 @@ namespace RTC
 		void UserOnResumed() override;
 		void CreateRtpStreams();
 		void RequestKeyFrame();
+		void StorePacketInTargetLayerRetransmissionBuffer(
+		  std::map<uint16_t, std::shared_ptr<RTC::RtpPacket>, RTC::SeqManager<uint16_t>::SeqLowerThan>&
+		    targetLayerRetransmissionBuffer,
+		  RTC::RtpPacket* packet,
+		  std::shared_ptr<RTC::RtpPacket>& sharedPacket);
 
 		/* Pure virtual methods inherited from RtpStreamSend::Listener. */
 	public:
@@ -75,6 +81,12 @@ namespace RTC
 		absl::flat_hash_map<RTC::RtpStreamSend*, bool> mapRtpStreamSyncRequired;
 		absl::flat_hash_map<RTC::RtpStreamSend*, std::unique_ptr<RTC::SeqManager<uint16_t>>>
 		  mapRtpStreamRtpSeqManager;
+		// Buffers to store packets that arrive earlier than the first packet of the
+		// video key frame.
+		absl::flat_hash_map<
+		  RTC::RtpStreamSend*,
+		  std::map<uint16_t, std::shared_ptr<RTC::RtpPacket>, RTC::SeqManager<uint16_t>::SeqLowerThan>>
+		  mapRtpStreamTargetLayerRetransmissionBuffer;
 	};
 } // namespace RTC
 

package/worker/include/RTC/RtcLogger.hpp

@@ -11,7 +11,7 @@ namespace RTC
 		class RtpPacket
 		{
 		public:
-			enum class DropReason : uint8_t
+			enum class DiscardReason : uint8_t
 			{
 				NONE = 0,
 				PRODUCER_NOT_FOUND,
@@ -23,18 +23,18 @@ namespace RTC
 				NOT_A_KEYFRAME,
 				EMPTY_PAYLOAD,
 				SPATIAL_LAYER_MISMATCH,
-				TOO_HIGH_TIMESTAMP_EXTRA_NEEDED,
 				PACKET_PREVIOUS_TO_SPATIAL_LAYER_SWITCH,
 				DROPPED_BY_CODEC,
-				SEND_RTP_STREAM_DISCARDED,
+				TOO_HIGH_TIMESTAMP_EXTRA_NEEDED,
+				SEND_RTP_STREAM_DISCARDED
 			};
 
-			static absl::flat_hash_map<DropReason, std::string> dropReason2String;
+			static absl::flat_hash_map<DiscardReason, std::string> discardReason2String;
 
 			RtpPacket()  = default;
 			~RtpPacket() = default;
 			void Sent();
-			void Dropped(DropReason dropReason);
+			void Discarded(DiscardReason discardReason);
 
 		private:
 			void Log() const;
@@ -51,8 +51,8 @@ namespace RTC
 			uint32_t sendRtpTimestamp{};
 			uint16_t recvSeqNumber{};
 			uint16_t sendSeqNumber{};
-			bool dropped{};
-			DropReason dropReason{ DropReason::NONE };
+			bool discarded{};
+			DiscardReason discardReason{ DiscardReason::NONE };
 		};
 	}; // namespace RtcLogger
 } // namespace RTC

package/worker/include/RTC/SimpleConsumer.hpp

@@ -5,6 +5,7 @@
 #include "RTC/Consumer.hpp"
 #include "RTC/SeqManager.hpp"
 #include "RTC/Shared.hpp"
+#include <map>
 
 namespace RTC
 {
@@ -72,6 +73,8 @@ namespace RTC
 		void UserOnResumed() override;
 		void CreateRtpStream();
 		void RequestKeyFrame();
+		void StorePacketInTargetLayerRetransmissionBuffer(
+		  RTC::RtpPacket* packet, std::shared_ptr<RTC::RtpPacket>& sharedPacket);
 		void EmitScore() const;
 
 		/* Pure virtual methods inherited from RtpStreamSend::Listener. */
@@ -90,6 +93,10 @@ namespace RTC
 		std::unique_ptr<RTC::SeqManager<uint16_t>> rtpSeqManager;
 		bool managingBitrate{ false };
 		std::unique_ptr<RTC::Codecs::EncodingContext> encodingContext;
+		// Buffer to store packets that arrive earlier than the first packet of the
+		// video key frame.
+		std::map<uint16_t, std::shared_ptr<RTC::RtpPacket>, RTC::SeqManager<uint16_t>::SeqLowerThan>
+		  targetLayerRetransmissionBuffer;
 	};
 } // namespace RTC
 

package/worker/include/RTC/SimulcastConsumer.hpp

@@ -6,6 +6,7 @@
 #include "RTC/Consumer.hpp"
 #include "RTC/SeqManager.hpp"
 #include "RTC/Shared.hpp"
+#include <map>
 
 namespace RTC
 {
@@ -95,6 +96,8 @@ namespace RTC
 		void UpdateTargetLayers(int16_t newTargetSpatialLayer, int16_t newTargetTemporalLayer);
 		bool CanSwitchToSpatialLayer(int16_t spatialLayer) const;
 		void EmitScore() const;
+		void StorePacketInTargetLayerRetransmissionBuffer(
+		  RTC::RtpPacket* packet, std::shared_ptr<RTC::RtpPacket>& sharedPacket);
 		void EmitLayersChange() const;
 		RTC::RtpStreamRecv* GetProducerCurrentRtpStream() const;
 		RTC::RtpStreamRecv* GetProducerTargetRtpStream() const;
@@ -129,7 +132,12 @@ namespace RTC
 		std::unique_ptr<RTC::Codecs::EncodingContext> encodingContext;
 		uint32_t tsOffset{ 0u }; // RTP Timestamp offset.
 		bool keyFrameForTsOffsetRequested{ false };
-		uint64_t lastBweDowngradeAtMs{ 0u }; // Last time we moved to lower spatial layer due to BWE.
+		// Last time we moved to lower spatial layer due to BWE.
+		uint64_t lastBweDowngradeAtMs{ 0u };
+		// Buffer to store packets that arrive earlier than the first packet of the
+		// video key frame.
+		std::map<uint16_t, std::shared_ptr<RTC::RtpPacket>, RTC::SeqManager<uint16_t>::SeqLowerThan>
+		  targetLayerRetransmissionBuffer;
 	};
 } // namespace RTC
 

package/worker/include/RTC/SvcConsumer.hpp

@@ -86,6 +86,8 @@ namespace RTC
 		bool RecalculateTargetLayers(int16_t& newTargetSpatialLayer, int16_t& newTargetTemporalLayer) const;
 		void UpdateTargetLayers(int16_t newTargetSpatialLayer, int16_t newTargetTemporalLayer);
 		void EmitScore() const;
+		void StorePacketInTargetLayerRetransmissionBuffer(
+		  RTC::RtpPacket* packet, std::shared_ptr<RTC::RtpPacket>& sharedPacket);
 		void EmitLayersChange() const;
 
 		/* Pure virtual methods inherited from RtpStreamSend::Listener. */
@@ -106,7 +108,12 @@ namespace RTC
 		int16_t provisionalTargetSpatialLayer{ -1 };
 		int16_t provisionalTargetTemporalLayer{ -1 };
 		std::unique_ptr<RTC::Codecs::EncodingContext> encodingContext;
-		uint64_t lastBweDowngradeAtMs{ 0u }; // Last time we moved to lower spatial layer due to BWE.
+		// Last time we moved to lower spatial layer due to BWE.
+		uint64_t lastBweDowngradeAtMs{ 0u };
+		// Buffer to store packets that arrive earlier than the first packet of the
+		// video key frame.
+		std::map<uint16_t, std::shared_ptr<RTC::RtpPacket>, RTC::SeqManager<uint16_t>::SeqLowerThan>
+		  targetLayerRetransmissionBuffer;
 	};
 } // namespace RTC
 

package/worker/meson.build

@@ -259,8 +259,8 @@ libuv_proj = subproject(
   ],
 )
 
-libsrtp2_proj = subproject(
-  'libsrtp2',
+libsrtp3_proj = subproject(
+  'libsrtp3',
   default_options: [
     'warning_level=0',
     'crypto-library=openssl',
@@ -310,7 +310,7 @@ dependencies = [
   abseil_cpp_proj.get_variable('absl_container_dep'),
   openssl_proj.get_variable('openssl_dep'),
   libuv_proj.get_variable('libuv_dep'),
-  libsrtp2_proj.get_variable('libsrtp2_dep'),
+  libsrtp3_proj.get_variable('libsrtp3_dep'),
   usrsctp_proj.get_variable('usrsctp_dep'),
   flatbuffers_proj.get_variable('flatbuffers_dep'),
   flatbuffers_generator_dep,
@@ -322,7 +322,7 @@ link_whole = [
   openssl_proj.get_variable('libcrypto_lib'),
   openssl_proj.get_variable('libssl_lib'),
   libuv_proj.get_variable('libuv'),
-  libsrtp2_proj.get_variable('libsrtp2'),
+  libsrtp3_proj.get_variable('libsrtp3'),
   usrsctp_proj.get_variable('usrsctp'),
   flatbuffers_proj.get_variable('flatbuffers_lib'),
   libwebrtc,

package/worker/meson_options.txt

@@ -1,5 +1,5 @@
 option('ms_log_trace', type : 'boolean', value : false, description : 'When set to true, logs the current method/function if current log level is "debug"')
 option('ms_log_file_line', type : 'boolean', value : false, description : 'When set to true, all the logging macros print more verbose information, including current file and line')
-option('ms_rtc_logger_rtp', type : 'boolean', value : false, description : 'When set to true, prints a line with information for each RTP packet')
+option('ms_rtc_logger_rtp', type : 'boolean', value : false, description : 'When set to true, prints a line with information for each processed RTP packet')
 option('ms_disable_liburing', type : 'boolean', value : false, description : 'When set to true, disables liburing integration despite current host supports it')
 option('ms_sctp_stack', type : 'boolean', value : false, description : 'When set to true, uses mediasoup SCTP stack instead of usrsctp library')

package/worker/scripts/package-lock.json

@@ -121,9 +121,10 @@
 			"integrity": "sha512-3oSeUO0TMV67hN1AmbXsK4yaqU7tjiHlbxRDZOpH0KW9+CeX4bRAaX0Anxt0tx2MrpRpWwQaPwIlISEJhYU5Pw=="
 		},
 		"node_modules/brace-expansion": {
-			"version": "2.0.1",
-			"resolved": "https://registry.npmjs.org/brace-expansion/-/brace-expansion-2.0.1.tgz",
-			"integrity": "sha512-XnAIvQ8eM+kC6aULx6wuQiwVsnzsi9d3WxzV3FpWTGA19F621kwdbsAcFKXgKUHZWsy+mY6iL1sHTxWEFCytDA==",
+			"version": "2.0.2",
+			"resolved": "https://registry.npmjs.org/brace-expansion/-/brace-expansion-2.0.2.tgz",
+			"integrity": "sha512-Jt0vHyM+jmUBqojB7E1NIYadt0vI0Qxjxd2TErW94wDz+E2LAm5vKMXXwg6ZZBTHPuUlDgQHKXvjGBdfcF1ZDQ==",
+			"license": "MIT",
 			"dependencies": {
 				"balanced-match": "^1.0.0"
 			}
@@ -144,9 +145,10 @@
 			}
 		},
 		"node_modules/clang-format/node_modules/brace-expansion": {
-			"version": "1.1.11",
-			"resolved": "https://registry.npmjs.org/brace-expansion/-/brace-expansion-1.1.11.tgz",
-			"integrity": "sha512-iCuPHDFgrHX7H2vEI/5xpz07zSHB00TpugqhmYtVmMO6518mCuRMoOYFldEBl0g187ufozdaHgWKcYFb61qGiA==",
+			"version": "1.1.12",
+			"resolved": "https://registry.npmjs.org/brace-expansion/-/brace-expansion-1.1.12.tgz",
+			"integrity": "sha512-9T9UjW3r0UW5c1Q7GTwllptXwhvYmEzFhzMfZ9H7FQWt+uZePjZPjBP/W1ZEyZ1twGWom5/56TF4lPcqjnDHcg==",
+			"license": "MIT",
 			"dependencies": {
 				"balanced-match": "^1.0.0",
 				"concat-map": "0.0.1"
@@ -674,9 +676,9 @@
 			"integrity": "sha512-3oSeUO0TMV67hN1AmbXsK4yaqU7tjiHlbxRDZOpH0KW9+CeX4bRAaX0Anxt0tx2MrpRpWwQaPwIlISEJhYU5Pw=="
 		},
 		"brace-expansion": {
-			"version": "2.0.1",
-			"resolved": "https://registry.npmjs.org/brace-expansion/-/brace-expansion-2.0.1.tgz",
-			"integrity": "sha512-XnAIvQ8eM+kC6aULx6wuQiwVsnzsi9d3WxzV3FpWTGA19F621kwdbsAcFKXgKUHZWsy+mY6iL1sHTxWEFCytDA==",
+			"version": "2.0.2",
+			"resolved": "https://registry.npmjs.org/brace-expansion/-/brace-expansion-2.0.2.tgz",
+			"integrity": "sha512-Jt0vHyM+jmUBqojB7E1NIYadt0vI0Qxjxd2TErW94wDz+E2LAm5vKMXXwg6ZZBTHPuUlDgQHKXvjGBdfcF1ZDQ==",
 			"requires": {
 				"balanced-match": "^1.0.0"
 			}
@@ -692,9 +694,9 @@
 			},
 			"dependencies": {
 				"brace-expansion": {
-					"version": "1.1.11",
-					"resolved": "https://registry.npmjs.org/brace-expansion/-/brace-expansion-1.1.11.tgz",
-					"integrity": "sha512-iCuPHDFgrHX7H2vEI/5xpz07zSHB00TpugqhmYtVmMO6518mCuRMoOYFldEBl0g187ufozdaHgWKcYFb61qGiA==",
+					"version": "1.1.12",
+					"resolved": "https://registry.npmjs.org/brace-expansion/-/brace-expansion-1.1.12.tgz",
+					"integrity": "sha512-9T9UjW3r0UW5c1Q7GTwllptXwhvYmEzFhzMfZ9H7FQWt+uZePjZPjBP/W1ZEyZ1twGWom5/56TF4lPcqjnDHcg==",
 					"requires": {
 						"balanced-match": "^1.0.0",
 						"concat-map": "0.0.1"

package/worker/src/RTC/PipeConsumer.cpp

@@ -6,9 +6,16 @@
 #include "MediaSoupErrors.hpp"
 #include "Utils.hpp"
 #include "RTC/Codecs/Tools.hpp"
+#ifdef MS_RTC_LOGGER_RTP
+#include "RTC/RtcLogger.hpp"
+#endif
 
 namespace RTC
 {
+	/* Static. */
+
+	static constexpr size_t TargetLayerRetransmissionBufferSize{ 15u };
+
 	/* Instance methods. */
 
 	PipeConsumer::PipeConsumer(
@@ -52,9 +59,13 @@ namespace RTC
 		{
 			delete rtpStream;
 		}
+
 		this->rtpStreams.clear();
 		this->mapMappedSsrcSsrc.clear();
 		this->mapSsrcRtpStream.clear();
+		this->mapRtpStreamSyncRequired.clear();
+		this->mapRtpStreamRtpSeqManager.clear();
+		this->mapRtpStreamTargetLayerRetransmissionBuffer.clear();
 	}
 
 	flatbuffers::Offset<FBS::Consumer::DumpResponse> PipeConsumer::FillBuffer(
@@ -222,67 +233,95 @@ namespace RTC
 		packet->logger.consumerId = this->id;
 #endif
 
+		auto ssrc           = this->mapMappedSsrcSsrc.at(packet->GetSsrc());
+		auto* rtpStream     = this->mapSsrcRtpStream.at(ssrc);
+		auto& syncRequired  = this->mapRtpStreamSyncRequired.at(rtpStream);
+		auto& rtpSeqManager = this->mapRtpStreamRtpSeqManager.at(rtpStream);
+		auto& targetLayerRetransmissionBuffer =
+		  this->mapRtpStreamTargetLayerRetransmissionBuffer.at(rtpStream);
+
 		if (!IsActive())
 		{
 #ifdef MS_RTC_LOGGER_RTP
-			packet->logger.Dropped(RtcLogger::RtpPacket::DropReason::CONSUMER_INACTIVE);
+			packet->logger.Discarded(RtcLogger::RtpPacket::DiscardReason::CONSUMER_INACTIVE);
 #endif
 
+			rtpSeqManager->Drop(packet->GetSequenceNumber());
+
 			return;
 		}
 
-		auto payloadType = packet->GetPayloadType();
-
-		// NOTE: This may happen if this Consumer supports just some codecs of those
-		// in the corresponding Producer.
-		if (!this->supportedCodecPayloadTypes[payloadType])
+		// If we need to sync, support key frames and this is not a key frame, ignore
+		// the packet.
+		if (syncRequired && this->keyFrameSupported && !packet->IsKeyFrame())
 		{
-			MS_DEBUG_DEV("payload type not supported [payloadType:%" PRIu8 "]", payloadType);
-
 #ifdef MS_RTC_LOGGER_RTP
-			packet->logger.Dropped(RtcLogger::RtpPacket::DropReason::UNSUPPORTED_PAYLOAD_TYPE);
+			packet->logger.Discarded(RtcLogger::RtpPacket::DiscardReason::NOT_A_KEYFRAME);
 #endif
 
+			// NOTE: No need to drop the packet in the RTP sequence manager since here
+			// we are blocking all packets but the key frame that would trigger sync
+			// below.
+
+			// Store the packet for the scenario in which this packet is part of the
+			// key frame and it arrived before the first packet of the key frame.
+			// TODO: Uncomment once this issue is fixed:
+			// https://github.com/versatica/mediasoup/issues/1554
+			// StorePacketInTargetLayerRetransmissionBuffer(
+			//   targetLayerRetransmissionBuffer, packet, sharedPacket);
+
 			return;
 		}
 
-		auto ssrc           = this->mapMappedSsrcSsrc.at(packet->GetSsrc());
-		auto* rtpStream     = this->mapSsrcRtpStream.at(ssrc);
-		auto& syncRequired  = this->mapRtpStreamSyncRequired.at(rtpStream);
-		auto& rtpSeqManager = this->mapRtpStreamRtpSeqManager.at(rtpStream);
+		auto payloadType = packet->GetPayloadType();
 
-		// If we need to sync, support key frames and this is not a key frame, ignore
-		// the packet.
-		if (syncRequired && this->keyFrameSupported && !packet->IsKeyFrame())
+		// NOTE: This may happen if this Consumer supports just some codecs of those
+		// in the corresponding Producer.
+		if (!this->supportedCodecPayloadTypes[payloadType])
 		{
+			MS_WARN_DEV("payload type not supported [payloadType:%" PRIu8 "]", payloadType);
+
 #ifdef MS_RTC_LOGGER_RTP
-			packet->logger.Dropped(RtcLogger::RtpPacket::DropReason::NOT_A_KEYFRAME);
+			packet->logger.Discarded(RtcLogger::RtpPacket::DiscardReason::UNSUPPORTED_PAYLOAD_TYPE);
 #endif
 
+			rtpSeqManager->Drop(packet->GetSequenceNumber());
+
 			return;
 		}
 
 		// Packets with only padding are not forwarded.
 		if (packet->GetPayloadLength() == 0)
 		{
-			rtpSeqManager->Drop(packet->GetSequenceNumber());
-
 #ifdef MS_RTC_LOGGER_RTP
-			packet->logger.Dropped(RtcLogger::RtpPacket::DropReason::EMPTY_PAYLOAD);
+			packet->logger.Discarded(RtcLogger::RtpPacket::DiscardReason::EMPTY_PAYLOAD);
 #endif
 
+			rtpSeqManager->Drop(packet->GetSequenceNumber());
+
 			return;
 		}
 
 		// Whether this is the first packet after re-sync.
 		const bool isSyncPacket = syncRequired;
 
+		// Whether packets stored in the target layer retransmission buffer must be
+		// sent once this packet is sent.
+		bool sendPacketsInTargetLayerRetransmissionBuffer{ false };
+
 		// Sync sequence number and timestamp if required.
 		if (isSyncPacket)
 		{
 			if (packet->IsKeyFrame())
 			{
-				MS_DEBUG_TAG(rtp, "sync key frame received");
+				MS_DEBUG_TAG(
+				  rtp,
+				  "sync key frame received [ssrc:%" PRIu32 ", seq:%" PRIu16 ", ts:%" PRIu32 "]",
+				  packet->GetSsrc(),
+				  packet->GetSequenceNumber(),
+				  packet->GetTimestamp());
+
+				sendPacketsInTargetLayerRetransmissionBuffer = true;
 			}
 
 			rtpSeqManager->Sync(packet->GetSequenceNumber() - 1);
@@ -321,8 +360,9 @@ namespace RTC
 			  origSeq);
 		}
 
-		// Process the packet.
-		if (rtpStream->ReceivePacket(packet, sharedPacket))
+		const bool sent = rtpStream->ReceivePacket(packet, sharedPacket);
+
+		if (sent)
 		{
 			// Send the packet.
 			this->listener->OnConsumerSendRtpPacket(this, packet);
@@ -341,11 +381,46 @@ namespace RTC
 			  packet->GetTimestamp(),
 			  origSsrc,
 			  origSeq);
+
+#ifdef MS_RTC_LOGGER_RTP
+			packet->logger.Discarded(RtcLogger::RtpPacket::DiscardReason::SEND_RTP_STREAM_DISCARDED);
+#endif
 		}
 
 		// Restore packet fields.
 		packet->SetSsrc(origSsrc);
 		packet->SetSequenceNumber(origSeq);
+
+		// If sent packet was the first packet of a key frame, let's send buffered
+		// packets belonging to the same key frame that arrived earlier due to
+		// packet misorder.
+		if (sendPacketsInTargetLayerRetransmissionBuffer)
+		{
+			// NOTE: Only send buffered packets if the first packet containing the key
+			// frame was sent.
+			if (sent)
+			{
+				for (auto& kv : targetLayerRetransmissionBuffer)
+				{
+					auto& bufferedSharedPacket = kv.second;
+					auto* bufferedPacket       = bufferedSharedPacket.get();
+
+					if (bufferedPacket->GetSequenceNumber() > origSeq)
+					{
+						MS_DEBUG_DEV(
+						  "sending packet buffered in the target layer retransmission buffer [ssrc:%" PRIu32
+						  ", seq:%" PRIu16 ", ts:%" PRIu32 "]",
+						  bufferedPacket->GetSsrc(),
+						  bufferedPacket->GetSequenceNumber(),
+						  bufferedPacket->GetTimestamp());
+
+						SendRtpPacket(bufferedPacket, bufferedSharedPacket);
+					}
+				}
+			}
+
+			targetLayerRetransmissionBuffer.clear();
+		}
 	}
 
 	bool PipeConsumer::GetRtcp(RTC::RTCP::CompoundPacket* packet, uint64_t nowMs)
@@ -557,6 +632,13 @@ namespace RTC
 		{
 			rtpStream->Pause();
 		}
+
+		for (auto& kv : this->mapRtpStreamTargetLayerRetransmissionBuffer)
+		{
+			auto& targetLayerRetransmissionBuffer = kv.second;
+
+			targetLayerRetransmissionBuffer.clear();
+		}
 	}
 
 	void PipeConsumer::UserOnPaused()
@@ -567,6 +649,13 @@ namespace RTC
 		{
 			rtpStream->Pause();
 		}
+
+		for (auto& kv : this->mapRtpStreamTargetLayerRetransmissionBuffer)
+		{
+			auto& targetLayerRetransmissionBuffer = kv.second;
+
+			targetLayerRetransmissionBuffer.clear();
+		}
 	}
 
 	void PipeConsumer::UserOnResumed()
@@ -690,6 +779,8 @@ namespace RTC
 
 			this->mapRtpStreamRtpSeqManager[rtpStream].reset(
 			  new RTC::SeqManager<uint16_t>(initialOutputSeq));
+
+			this->mapRtpStreamTargetLayerRetransmissionBuffer[rtpStream];
 		}
 	}
 
@@ -710,7 +801,37 @@ namespace RTC
 		}
 	}
 
-	inline void PipeConsumer::OnRtpStreamScore(
+	void PipeConsumer::StorePacketInTargetLayerRetransmissionBuffer(
+	  std::map<uint16_t, std::shared_ptr<RTC::RtpPacket>, RTC::SeqManager<uint16_t>::SeqLowerThan>&
+	    targetLayerRetransmissionBuffer,
+	  RTC::RtpPacket* packet,
+	  std::shared_ptr<RTC::RtpPacket>& sharedPacket)
+	{
+		MS_TRACE();
+
+		MS_DEBUG_DEV(
+		  "storing packet in target layer retransmission buffer [ssrc:%" PRIu32 ", seq:%" PRIu16
+		  ", ts:%" PRIu32 "]",
+		  packet->GetSsrc(),
+		  packet->GetSequenceNumber(),
+		  packet->GetTimestamp());
+
+		// Store original packet into the buffer. Only clone once and only if
+		// necessary.
+		if (!sharedPacket)
+		{
+			sharedPacket.reset(packet->Clone());
+		}
+
+		targetLayerRetransmissionBuffer[packet->GetSequenceNumber()] = sharedPacket;
+
+		if (targetLayerRetransmissionBuffer.size() > TargetLayerRetransmissionBufferSize)
+		{
+			targetLayerRetransmissionBuffer.erase(targetLayerRetransmissionBuffer.begin());
+		}
+	}
+
+	void PipeConsumer::OnRtpStreamScore(
 	  RTC::RtpStream* /*rtpStream*/, uint8_t /*score*/, uint8_t /*previousScore*/)
 	{
 		MS_TRACE();
@@ -718,8 +839,7 @@ namespace RTC
 		// Do nothing.
 	}
 
-	inline void PipeConsumer::OnRtpStreamRetransmitRtpPacket(
-	  RTC::RtpStreamSend* rtpStream, RTC::RtpPacket* packet)
+	void PipeConsumer::OnRtpStreamRetransmitRtpPacket(RTC::RtpStreamSend* rtpStream, RTC::RtpPacket* packet)
 	{
 		MS_TRACE();
 

package/worker/src/RTC/Producer.cpp

@@ -10,6 +10,9 @@
 #include "RTC/Consts.hpp"
 #include "RTC/RTCP/Feedback.hpp"
 #include "RTC/RTCP/XrReceiverReferenceTime.hpp"
+#ifdef MS_RTC_LOGGER_RTP
+#include "RTC/RtcLogger.hpp"
+#endif
 #include <absl/container/inlined_vector.h>
 #include <cstring> // std::memcpy()
 
@@ -592,7 +595,7 @@ namespace RTC
 			MS_WARN_TAG(rtp, "no stream found for received packet [ssrc:%" PRIu32 "]", packet->GetSsrc());
 
 #ifdef MS_RTC_LOGGER_RTP
-			packet->logger.Dropped(RtcLogger::RtpPacket::DropReason::RECV_RTP_STREAM_NOT_FOUND);
+			packet->logger.Discarded(RtcLogger::RtpPacket::DiscardReason::RECV_RTP_STREAM_NOT_FOUND);
 #endif
 
 			return ReceiveRtpPacketResult::DISCARDED;
@@ -619,7 +622,7 @@ namespace RTC
 				}
 
 #ifdef MS_RTC_LOGGER_RTP
-				packet->logger.Dropped(RtcLogger::RtpPacket::DropReason::RECV_RTP_STREAM_DISCARDED);
+				packet->logger.Discarded(RtcLogger::RtpPacket::DiscardReason::RECV_RTP_STREAM_DISCARDED);
 #endif
 
 				return result;
@@ -635,7 +638,7 @@ namespace RTC
 			if (!rtpStream->ReceiveRtxPacket(packet))
 			{
 #ifdef MS_RTC_LOGGER_RTP
-				packet->logger.Dropped(RtcLogger::RtpPacket::DropReason::RECV_RTP_STREAM_NOT_FOUND);
+				packet->logger.Discarded(RtcLogger::RtpPacket::DiscardReason::RECV_RTP_STREAM_NOT_FOUND);
 #endif
 
 				return result;