P2P - Libdatachannel/RTC Library Files

libdatachannel/LICENSE

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+MIT License
+
+Copyright (c) 2024 RandomityGuy
+
+Permission is hereby granted, free of charge, to any person obtaining a copy
+of this software and associated documentation files (the "Software"), to deal
+in the Software without restriction, including without limitation the rights
+to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+copies of the Software, and to permit persons to whom the Software is
+furnished to do so, subject to the following conditions:
+
+The above copyright notice and this permission notice shall be included in all
+copies or substantial portions of the Software.
+
+THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+SOFTWARE.

libdatachannel/include/rtc/av1rtppacketizer.hpp

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+/**
+ * Copyright (c) 2023 Paul-Louis Ageneau
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+#ifndef RTC_AV1_RTP_PACKETIZER_H
+#define RTC_AV1_RTP_PACKETIZER_H
+
+#if RTC_ENABLE_MEDIA
+
+#include "mediahandler.hpp"
+#include "nalunit.hpp"
+#include "rtppacketizer.hpp"
+
+namespace rtc {
+
+// RTP packetization of AV1 payload
+class RTC_CPP_EXPORT AV1RtpPacketizer final : public RtpPacketizer {
+public:
+	inline static const uint32_t ClockRate = VideoClockRate;
+	[[deprecated("Use ClockRate")]] inline static const uint32_t defaultClockRate = ClockRate;
+
+	// Define how OBUs are seperated in a AV1 Sample
+	enum class Packetization {
+		Obu = RTC_OBU_PACKETIZED_OBU,
+		TemporalUnit = RTC_OBU_PACKETIZED_TEMPORAL_UNIT,
+	};
+
+	// Constructs AV1 payload packetizer with given RTP configuration.
+	// @note RTP configuration is used in packetization process which may change some configuration
+	// properties such as sequence number.
+	AV1RtpPacketizer(Packetization packetization, shared_ptr<RtpPacketizationConfig> rtpConfig,
+	                 size_t maxFragmentSize = DefaultMaxFragmentSize);
+
+private:
+	static std::vector<binary> extractTemporalUnitObus(const binary &data);
+
+	std::vector<binary> fragment(binary data) override;
+	std::vector<binary> fragmentObu(const binary &data);
+
+	const Packetization mPacketization;
+	const size_t mMaxFragmentSize;
+
+	std::unique_ptr<binary> mSequenceHeader;
+};
+
+// For backward compatibility, do not use
+using AV1PacketizationHandler [[deprecated("Add AV1RtpPacketizer directly")]] = PacketizationHandler;
+
+} // namespace rtc
+
+#endif /* RTC_ENABLE_MEDIA */
+
+#endif /* RTC_AV1_RTP_PACKETIZER_H */

libdatachannel/include/rtc/candidate.hpp

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+/**
+ * Copyright (c) 2019 Paul-Louis Ageneau
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+#ifndef RTC_CANDIDATE_H
+#define RTC_CANDIDATE_H
+
+#include "common.hpp"
+
+#include <string>
+
+namespace rtc {
+
+class RTC_CPP_EXPORT Candidate {
+public:
+	enum class Family { Unresolved, Ipv4, Ipv6 };
+	enum class Type { Unknown, Host, ServerReflexive, PeerReflexive, Relayed };
+	enum class TransportType { Unknown, Udp, TcpActive, TcpPassive, TcpSo, TcpUnknown };
+
+	Candidate();
+	Candidate(string candidate);
+	Candidate(string candidate, string mid);
+
+	void hintMid(string mid);
+	void changeAddress(string addr);
+	void changeAddress(string addr, uint16_t port);
+	void changeAddress(string addr, string service);
+
+	enum class ResolveMode { Simple, Lookup };
+	bool resolve(ResolveMode mode = ResolveMode::Simple);
+
+	Type type() const;
+	TransportType transportType() const;
+	uint32_t priority() const;
+	string candidate() const;
+	string mid() const;
+	operator string() const;
+
+	bool operator==(const Candidate &other) const;
+	bool operator!=(const Candidate &other) const;
+
+	bool isResolved() const;
+	Family family() const;
+	optional<string> address() const;
+	optional<uint16_t> port() const;
+
+private:
+	void parse(string candidate);
+
+	string mFoundation;
+	uint32_t mComponent, mPriority;
+	string mTypeString, mTransportString;
+	Type mType;
+	TransportType mTransportType;
+	string mNode, mService;
+	string mTail;
+
+	optional<string> mMid;
+
+	// Extracted on resolution
+	Family mFamily;
+	string mAddress;
+	uint16_t mPort;
+};
+
+RTC_CPP_EXPORT std::ostream &operator<<(std::ostream &out, const Candidate &candidate);
+RTC_CPP_EXPORT std::ostream &operator<<(std::ostream &out, const Candidate::Type &type);
+RTC_CPP_EXPORT std::ostream &operator<<(std::ostream &out,
+                                        const Candidate::TransportType &transportType);
+
+} // namespace rtc
+
+#endif

libdatachannel/include/rtc/channel.hpp

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+/**
+ * Copyright (c) 2019-2021 Paul-Louis Ageneau
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+#ifndef RTC_CHANNEL_H
+#define RTC_CHANNEL_H
+
+#include "common.hpp"
+
+#include <atomic>
+#include <functional>
+
+namespace rtc {
+
+namespace impl {
+struct Channel;
+}
+
+class RTC_CPP_EXPORT Channel : private CheshireCat<impl::Channel> {
+public:
+	virtual ~Channel();
+
+	virtual void close() = 0;
+	virtual bool send(message_variant data) = 0; // returns false if buffered
+	virtual bool send(const byte *data, size_t size) = 0;
+
+	virtual bool isOpen() const = 0;
+	virtual bool isClosed() const = 0;
+	virtual size_t maxMessageSize() const; // max message size in a call to send
+	virtual size_t bufferedAmount() const; // total size buffered to send
+
+	void onOpen(std::function<void()> callback);
+	void onClosed(std::function<void()> callback);
+	void onError(std::function<void(string error)> callback);
+
+	void onMessage(std::function<void(message_variant data)> callback);
+	void onMessage(std::function<void(binary data)> binaryCallback,
+	               std::function<void(string data)> stringCallback);
+
+	void onBufferedAmountLow(std::function<void()> callback);
+	void setBufferedAmountLowThreshold(size_t amount);
+
+	void resetCallbacks();
+
+	// Extended API
+	optional<message_variant> receive(); // only if onMessage unset
+	optional<message_variant> peek();    // only if onMessage unset
+	size_t availableAmount() const;      // total size available to receive
+	void onAvailable(std::function<void()> callback);
+
+protected:
+	Channel(impl_ptr<impl::Channel> impl);
+};
+
+} // namespace rtc
+
+#endif // RTC_CHANNEL_H

libdatachannel/include/rtc/common.hpp

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+/**
+ * Copyright (c) 2019 Paul-Louis Ageneau
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+#ifndef RTC_COMMON_H
+#define RTC_COMMON_H
+
+#ifdef RTC_STATIC
+#define RTC_CPP_EXPORT
+#else // dynamic library
+#ifdef _WIN32
+#ifdef RTC_EXPORTS
+#define RTC_CPP_EXPORT __declspec(dllexport) // building the library
+#else
+#define RTC_CPP_EXPORT __declspec(dllimport) // using the library
+#endif
+#else // not WIN32
+#define RTC_CPP_EXPORT
+#endif
+#endif
+
+#ifdef _WIN32
+#ifndef _WIN32_WINNT
+#define _WIN32_WINNT 0x0602 // Windows 8
+#endif
+#ifdef _MSC_VER
+#pragma warning(disable : 4251) // disable "X needs to have dll-interface..."
+#endif
+#endif
+
+#ifndef RTC_ENABLE_WEBSOCKET
+#define RTC_ENABLE_WEBSOCKET 1
+#endif
+
+#ifndef RTC_ENABLE_MEDIA
+#define RTC_ENABLE_MEDIA 1
+#endif
+
+#include "rtc.h" // for C API defines
+
+#include "utils.hpp"
+
+#include <cstddef>
+#include <functional>
+#include <memory>
+#include <mutex>
+#include <optional>
+#include <string>
+#include <string_view>
+#include <variant>
+#include <vector>
+
+namespace rtc {
+
+using std::byte;
+using std::nullopt;
+using std::optional;
+using std::shared_ptr;
+using std::string;
+using std::string_view;
+using std::unique_ptr;
+using std::variant;
+using std::weak_ptr;
+
+using binary = std::vector<byte>;
+using message_variant = variant<binary, string>;
+
+using std::int16_t;
+using std::int32_t;
+using std::int64_t;
+using std::int8_t;
+using std::ptrdiff_t;
+using std::size_t;
+using std::uint16_t;
+using std::uint32_t;
+using std::uint64_t;
+using std::uint8_t;
+
+} // namespace rtc
+
+#endif

libdatachannel/include/rtc/configuration.hpp

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+/**
+ * Copyright (c) 2019 Paul-Louis Ageneau
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+#ifndef RTC_ICE_CONFIGURATION_H
+#define RTC_ICE_CONFIGURATION_H
+
+#include "common.hpp"
+
+#include <vector>
+
+namespace rtc {
+
+struct RTC_CPP_EXPORT IceServer {
+	enum class Type { Stun, Turn };
+	enum class RelayType { TurnUdp, TurnTcp, TurnTls };
+
+	// Any type
+	IceServer(const string &url);
+
+	// STUN
+	IceServer(string hostname_, uint16_t port_);
+	IceServer(string hostname_, string service_);
+
+	// TURN
+	IceServer(string hostname_, uint16_t port, string username_, string password_,
+	          RelayType relayType_ = RelayType::TurnUdp);
+	IceServer(string hostname_, string service_, string username_, string password_,
+	          RelayType relayType_ = RelayType::TurnUdp);
+
+	string hostname;
+	uint16_t port;
+	Type type;
+	string username;
+	string password;
+	RelayType relayType;
+};
+
+struct RTC_CPP_EXPORT ProxyServer {
+	enum class Type { Http, Socks5 };
+
+	ProxyServer(const string &url);
+
+	ProxyServer(Type type_, string hostname_, uint16_t port_);
+	ProxyServer(Type type_, string hostname_, uint16_t port_, string username_, string password_);
+
+	Type type;
+	string hostname;
+	uint16_t port;
+	optional<string> username;
+	optional<string> password;
+};
+
+enum class CertificateType {
+	Default = RTC_CERTIFICATE_DEFAULT, // ECDSA
+	Ecdsa = RTC_CERTIFICATE_ECDSA,
+	Rsa = RTC_CERTIFICATE_RSA
+};
+
+enum class TransportPolicy { All = RTC_TRANSPORT_POLICY_ALL, Relay = RTC_TRANSPORT_POLICY_RELAY };
+
+struct RTC_CPP_EXPORT Configuration {
+	// ICE settings
+	std::vector<IceServer> iceServers;
+	optional<ProxyServer> proxyServer; // libnice only
+	optional<string> bindAddress;      // libjuice only, default any
+
+	// Options
+	CertificateType certificateType = CertificateType::Default;
+	TransportPolicy iceTransportPolicy = TransportPolicy::All;
+	bool enableIceTcp = false;
+	bool enableIceUdpMux = false; // libjuice only
+	bool disableAutoNegotiation = false;
+	bool disableAutoGathering = false;
+	bool forceMediaTransport = false;
+	bool disableFingerprintVerification = false;
+
+	// Port range
+	uint16_t portRangeBegin = 1024;
+	uint16_t portRangeEnd = 65535;
+
+	// Network MTU
+	optional<size_t> mtu;
+
+	// Local maximum message size for Data Channels
+	optional<size_t> maxMessageSize;
+
+	// Certificates and private keys
+	optional<string> certificatePemFile;
+	optional<string> keyPemFile;
+	optional<string> keyPemPass;
+};
+
+#ifdef RTC_ENABLE_WEBSOCKET
+
+struct WebSocketConfiguration {
+	bool disableTlsVerification = false; // if true, don't verify the TLS certificate
+	optional<ProxyServer> proxyServer;   // only non-authenticated http supported for now
+	std::vector<string> protocols;
+	optional<std::chrono::milliseconds> tcpConnectionTimeout; // zero to disable
+	optional<std::chrono::milliseconds> connectionTimeout;    // zero to disable
+	optional<std::chrono::milliseconds> pingInterval;         // zero to disable
+	optional<int> maxOutstandingPings;
+	optional<string> caCertificatePemFile;
+	optional<string> certificatePemFile;
+	optional<string> keyPemFile;
+	optional<string> keyPemPass;
+	optional<size_t> maxMessageSize;
+};
+
+struct WebSocketServerConfiguration {
+	uint16_t port = 8080;
+	bool enableTls = false;
+	optional<string> certificatePemFile;
+	optional<string> keyPemFile;
+	optional<string> keyPemPass;
+	optional<string> bindAddress;
+	optional<std::chrono::milliseconds> connectionTimeout;
+	optional<size_t> maxMessageSize;
+};
+
+#endif
+
+} // namespace rtc
+
+#endif

libdatachannel/include/rtc/datachannel.hpp

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+/**
+ * Copyright (c) 2019 Paul-Louis Ageneau
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+#ifndef RTC_DATA_CHANNEL_H
+#define RTC_DATA_CHANNEL_H
+
+#include "channel.hpp"
+#include "common.hpp"
+#include "reliability.hpp"
+
+#include <type_traits>
+
+namespace rtc {
+
+namespace impl {
+
+struct DataChannel;
+struct PeerConnection;
+
+} // namespace impl
+
+class RTC_CPP_EXPORT DataChannel final : private CheshireCat<impl::DataChannel>, public Channel {
+public:
+	DataChannel(impl_ptr<impl::DataChannel> impl);
+	~DataChannel() override;
+
+	optional<uint16_t> stream() const;
+	optional<uint16_t> id() const;
+	string label() const;
+	string protocol() const;
+	Reliability reliability() const;
+
+	bool isOpen(void) const override;
+	bool isClosed(void) const override;
+	size_t maxMessageSize() const override;
+
+	void close(void) override;
+	bool send(message_variant data) override;
+	bool send(const byte *data, size_t size) override;
+	template <typename Buffer> bool sendBuffer(const Buffer &buf);
+	template <typename Iterator> bool sendBuffer(Iterator first, Iterator last);
+
+private:
+	using CheshireCat<impl::DataChannel>::impl;
+};
+
+template <typename Buffer> std::pair<const byte *, size_t> to_bytes(const Buffer &buf) {
+	using T = typename std::remove_pointer<decltype(buf.data())>::type;
+	using E = typename std::conditional<std::is_void<T>::value, byte, T>::type;
+	return std::make_pair(static_cast<const byte *>(static_cast<const void *>(buf.data())),
+	                      buf.size() * sizeof(E));
+}
+
+template <typename Buffer> bool DataChannel::sendBuffer(const Buffer &buf) {
+	auto [bytes, size] = to_bytes(buf);
+	return send(bytes, size);
+}
+
+template <typename Iterator> bool DataChannel::sendBuffer(Iterator first, Iterator last) {
+	size_t size = 0;
+	for (Iterator it = first; it != last; ++it)
+		size += it->size();
+
+	binary buffer(size);
+	byte *pos = buffer.data();
+	for (Iterator it = first; it != last; ++it) {
+		auto [bytes, len] = to_bytes(*it);
+		pos = std::copy(bytes, bytes + len, pos);
+	}
+	return send(std::move(buffer));
+}
+
+} // namespace rtc
+
+#endif

libdatachannel/include/rtc/dependencydescriptor.hpp

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+/**
+ * Copyright (c) 2024 Shigemasa Watanabe (Wandbox)
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+#ifndef RTC_DEPENDENCY_DESCRIPTOR_H
+#define RTC_DEPENDENCY_DESCRIPTOR_H
+
+#include "common.hpp"
+
+#include <bitset>
+
+namespace rtc {
+
+struct BitWriter {
+	static BitWriter fromSizeBits(byte *buf, size_t offsetBits, size_t sizeBits);
+	static BitWriter fromNull();
+
+	size_t getWrittenBits() const;
+
+	bool write(uint64_t v, size_t bits);
+	// Write non-symmetric unsigned encoded integer
+	// ref: https://aomediacodec.github.io/av1-rtp-spec/#a82-syntax
+	bool writeNonSymmetric(uint64_t v, uint64_t n);
+
+private:
+	size_t writePartialByte(uint8_t *p, size_t offset, uint64_t v, size_t bits);
+
+private:
+	byte *mBuf = nullptr;
+	size_t mInitialOffset = 0;
+	size_t mOffset = 0;
+	size_t mSize = 0;
+};
+
+enum class DecodeTargetIndication {
+	NotPresent = 0,
+	Discardable = 1,
+	Switch = 2,
+	Required = 3,
+};
+
+struct RenderResolution {
+	int width = 0;
+	int height = 0;
+};
+
+struct FrameDependencyTemplate {
+	int spatialId = 0;
+	int temporalId = 0;
+	std::vector<DecodeTargetIndication> decodeTargetIndications;
+	std::vector<int> frameDiffs;
+	std::vector<int> chainDiffs;
+};
+
+struct FrameDependencyStructure {
+	int templateIdOffset = 0;
+	int decodeTargetCount = 0;
+	int chainCount = 0;
+	std::vector<int> decodeTargetProtectedBy;
+	std::vector<RenderResolution> resolutions;
+	std::vector<FrameDependencyTemplate> templates;
+};
+
+struct DependencyDescriptor {
+	bool startOfFrame = true;
+	bool endOfFrame = true;
+	int frameNumber = 0;
+	FrameDependencyTemplate dependencyTemplate;
+	std::optional<RenderResolution> resolution;
+	std::optional<uint32_t> activeDecodeTargetsBitmask;
+	bool structureAttached;
+};
+
+struct DependencyDescriptorContext {
+	DependencyDescriptor descriptor;
+	std::bitset<32> activeChains;
+	FrameDependencyStructure structure;
+};
+
+// Write dependency descriptor to RTP Header Extension
+// Dependency descriptor specification is here:
+// https://aomediacodec.github.io/av1-rtp-spec/#dependency-descriptor-rtp-header-extension
+class DependencyDescriptorWriter {
+public:
+	explicit DependencyDescriptorWriter(const DependencyDescriptorContext& context);
+	size_t getSizeBits() const;
+	size_t getSize() const;
+	void writeTo(byte *buf, size_t sizeBytes) const;
+
+private:
+	void doWriteTo(BitWriter &writer) const;
+	void writeBits(BitWriter &writer, uint64_t v, size_t bits) const;
+	void writeNonSymmetric(BitWriter &writer, uint64_t v, uint64_t n) const;
+
+private:
+	const FrameDependencyStructure &mStructure;
+	std::bitset<32> mActiveChains;
+	const DependencyDescriptor &mDescriptor;
+};
+
+} // namespace rtc
+
+#endif

libdatachannel/include/rtc/description.hpp

@@ -0,0 +1,401 @@
+/**
+ * Copyright (c) 2019-2020 Paul-Louis Ageneau
+ * Copyright (c) 2020 Staz Modrzynski
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+#ifndef RTC_DESCRIPTION_H
+#define RTC_DESCRIPTION_H
+
+#include "candidate.hpp"
+#include "common.hpp"
+#include "rtp.hpp"
+
+#include <iostream>
+#include <map>
+#include <vector>
+
+namespace rtc {
+
+const string DEFAULT_OPUS_AUDIO_PROFILE =
+    "minptime=10;maxaveragebitrate=96000;stereo=1;sprop-stereo=1;useinbandfec=1";
+
+// Use Constrained Baseline profile Level 3.1 (necessary for Firefox)
+// https://developer.mozilla.org/en-US/docs/Web/Media/Formats/WebRTC_codecs#Supported_video_codecs
+// TODO: Should be 42E0 but 42C0 appears to be more compatible. Investigate this.
+const string DEFAULT_H264_VIDEO_PROFILE =
+    "profile-level-id=42e01f;packetization-mode=1;level-asymmetry-allowed=1";
+
+struct CertificateFingerprint {
+	enum class Algorithm { Sha1, Sha224, Sha256, Sha384, Sha512 };
+	static string AlgorithmIdentifier(Algorithm algorithm);
+	static size_t AlgorithmSize(Algorithm algorithm);
+
+	bool isValid() const;
+
+	Algorithm algorithm;
+	string value;
+};
+
+class RTC_CPP_EXPORT Description {
+public:
+	enum class Type { Unspec, Offer, Answer, Pranswer, Rollback };
+	enum class Role { ActPass, Passive, Active };
+
+	enum class Direction {
+		SendOnly = RTC_DIRECTION_SENDONLY,
+		RecvOnly = RTC_DIRECTION_RECVONLY,
+		SendRecv = RTC_DIRECTION_SENDRECV,
+		Inactive = RTC_DIRECTION_INACTIVE,
+		Unknown = RTC_DIRECTION_UNKNOWN
+	};
+
+	Description(const string &sdp, Type type = Type::Unspec, Role role = Role::ActPass);
+	Description(const string &sdp, string typeString);
+
+	Type type() const;
+	string typeString() const;
+	Role role() const;
+	string bundleMid() const;
+	std::vector<string> iceOptions() const;
+	optional<string> iceUfrag() const;
+	optional<string> icePwd() const;
+	optional<CertificateFingerprint> fingerprint() const;
+	bool ended() const;
+
+	void hintType(Type type);
+	void addIceOption(string option);
+	void removeIceOption(const string &option);
+	void setIceAttribute(string ufrag, string pwd);
+	void setFingerprint(CertificateFingerprint f);
+
+	std::vector<string> attributes() const;
+	void addAttribute(string attr);
+	void removeAttribute(const string &attr);
+
+	std::vector<Candidate> candidates() const;
+	std::vector<Candidate> extractCandidates();
+	bool hasCandidate(const Candidate &candidate) const;
+	void addCandidate(Candidate candidate);
+	void addCandidates(std::vector<Candidate> candidates);
+	void endCandidates();
+
+	operator string() const;
+	string generateSdp(string_view eol = "\r\n") const;
+	string generateApplicationSdp(string_view eol = "\r\n") const;
+
+	class RTC_CPP_EXPORT RidAttribute {
+	public:
+		RidAttribute(string name, string value);
+
+		[[nodiscard]] const string &name() const;
+		[[nodiscard]] const string &value() const;
+
+		void value(string value);
+
+	private:
+		string mName;
+		string mValue;
+	};
+
+	class RTC_CPP_EXPORT Rid {
+	public:
+		explicit Rid(string rid);
+		Rid(string rid, std::vector<RidAttribute> attributes);
+
+		[[nodiscard]] const string &rid() const;
+		[[nodiscard]] const std::vector<RidAttribute> &attributes() const;
+
+	private:
+		string mRid;
+		std::vector<RidAttribute> mAttributes;
+	};
+
+	class RTC_CPP_EXPORT RidBuilder {
+	public:
+		explicit RidBuilder(string rid);
+
+		// https://datatracker.ietf.org/doc/html/rfc8851#name-arid-restrictions
+		RidBuilder& max_width(uint32_t value);
+		RidBuilder& max_height(uint32_t value);
+		RidBuilder& max_fps(uint32_t value);
+		RidBuilder& max_br(uint32_t value);
+
+		RidBuilder& custom(string key, string value);
+
+		// One time use only
+		[[nodiscard]] Rid build();
+
+	private:
+		string mRid;
+		std::vector<RidAttribute> mAttributes;
+
+		RidBuilder& saveAttribute(string key, string value);
+	};
+
+	class RTC_CPP_EXPORT Entry {
+	public:
+		virtual ~Entry() = default;
+
+		virtual string type() const;
+		virtual string protocol() const;
+		virtual string description() const;
+		virtual string mid() const;
+
+		Direction direction() const;
+		void setDirection(Direction dir);
+
+		bool isRemoved() const;
+		void markRemoved();
+
+		std::vector<string> attributes() const;
+		void addAttribute(string attr);
+		void removeAttribute(const string &attr);
+		void addRid(string rid);	// Just the name
+		void addRid(Rid rid);		// With RFC 8851 attributes
+
+		std::vector<Rid> rids() const;
+
+		struct RTC_CPP_EXPORT ExtMap {
+			static int parseId(string_view description);
+
+			ExtMap(int id, string uri, Direction direction = Direction::Unknown);
+			ExtMap(string_view description);
+
+			void setDescription(string_view description);
+
+			int id;
+			string uri;
+			string attributes;
+			Direction direction = Direction::Unknown;
+		};
+
+		std::vector<int> extIds();
+		ExtMap *extMap(int id);
+		const ExtMap *extMap(int id) const;
+		void addExtMap(ExtMap map);
+		void removeExtMap(int id);
+
+		operator string() const;
+		string generateSdp(string_view eol = "\r\n", string_view addr = "0.0.0.0",
+		                   uint16_t port = 9) const;
+
+		virtual void parseSdpLine(string_view line);
+
+	protected:
+		Entry(const string &mline, string mid, Direction dir = Direction::Unknown);
+
+		virtual string generateSdpLines(string_view eol) const;
+
+		std::vector<string> mAttributes;
+		std::map<int, ExtMap> mExtMaps;
+
+	private:
+		string mType;
+		string mProtocol;
+		string mDescription;
+		string mMid;
+		std::vector<Rid> mRids;
+		Direction mDirection;
+		bool mIsRemoved;
+	};
+
+	struct RTC_CPP_EXPORT Application : public Entry {
+	public:
+		Application(string mid = "data");
+		Application(const string &mline, string mid);
+		virtual ~Application() = default;
+
+		Application reciprocate() const;
+
+		void setSctpPort(uint16_t port);
+		void hintSctpPort(uint16_t port);
+		void setMaxMessageSize(size_t size);
+
+		optional<uint16_t> sctpPort() const;
+		optional<size_t> maxMessageSize() const;
+
+		virtual void parseSdpLine(string_view line) override;
+
+	private:
+		virtual string generateSdpLines(string_view eol) const override;
+
+		optional<uint16_t> mSctpPort;
+		optional<size_t> mMaxMessageSize;
+	};
+
+	// Media (non-data)
+	class RTC_CPP_EXPORT Media : public Entry {
+	public:
+		Media(const string &mline, string mid, Direction dir = Direction::SendOnly);
+		Media(const string &sdp);
+		virtual ~Media() = default;
+
+		string description() const override;
+		Media reciprocate() const;
+
+		void addSSRC(uint32_t ssrc, optional<string> name, optional<string> msid = nullopt,
+		             optional<string> trackId = nullopt);
+		void removeSSRC(uint32_t ssrc);
+		void replaceSSRC(uint32_t old, uint32_t ssrc, optional<string> name,
+		                 optional<string> msid = nullopt, optional<string> trackID = nullopt);
+		bool hasSSRC(uint32_t ssrc) const;
+		void clearSSRCs();
+		std::vector<uint32_t> getSSRCs() const;
+		optional<std::string> getCNameForSsrc(uint32_t ssrc) const;
+
+		void addRtxSSRC(SSRC primarySsrc, SSRC rtxSsrc, optional<string> cname = nullopt);
+		void removeRtxSSRC(SSRC primarySsrc);
+		optional<SSRC> getRtxSsrcForSsrc(SSRC primarySsrc) const;
+		optional<SSRC> getSsrcForRtxSsrc(SSRC rtxSsrc) const;
+
+		int bitrate() const;
+		void setBitrate(int bitrate);
+
+		struct RTC_CPP_EXPORT RtpMap {
+			static int parsePayloadType(string_view description);
+
+			explicit RtpMap(int payloadType);
+			RtpMap(string_view description);
+
+			void setDescription(string_view description);
+
+			void addFeedback(string fb);
+			void removeFeedback(const string &str);
+			bool hasFeedback(const string &str) const;
+
+			void addParameter(string p);
+			void removeParameter(const string &str);
+
+			int payloadType;
+			string format;
+			int clockRate;
+			string encParams;
+
+			std::vector<string> rtcpFbs;
+			std::vector<string> fmtps;
+		};
+
+		bool hasPayloadType(int payloadType) const;
+		std::vector<int> payloadTypes() const;
+		RtpMap *rtpMap(int payloadType);
+		const RtpMap *rtpMap(int payloadType) const;
+		void addRtpMap(RtpMap map);
+		void removeRtpMap(int payloadType);
+		void removeFormat(const string &format);
+
+		void addRtxCodec(int payloadType, int origPayloadType, unsigned int clockRate);
+
+		optional<int> getRtxPayloadType(int primaryPayloadType) const;
+
+		bool isRtxEnabled() const;
+		void disableRtx();
+
+		virtual void parseSdpLine(string_view line) override;
+
+	private:
+		virtual string generateSdpLines(string_view eol) const override;
+
+		int mBas = -1;
+
+		std::vector<int> mOrderedPayloadTypes;
+		std::map<int, RtpMap> mRtpMaps;
+		std::vector<uint32_t> mSsrcs;
+		std::map<uint32_t, string> mCNameMap;
+		std::map<SSRC, SSRC> mSsrcToRtxSsrc;  // primary_ssrc -> rtx_ssrc
+	};
+
+	class RTC_CPP_EXPORT Audio : public Media {
+	public:
+		Audio(string mid = "audio", Direction dir = Direction::SendOnly);
+
+		void addAudioCodec(int payloadType, string codec, optional<string> profile = std::nullopt);
+		void addOpusCodec(int payloadType, optional<string> profile = DEFAULT_OPUS_AUDIO_PROFILE);
+		void addPCMACodec(int payloadType, optional<string> profile = std::nullopt);
+		void addPCMUCodec(int payloadType, optional<string> profile = std::nullopt);
+		void addAACCodec(int payloadType, optional<string> profile = std::nullopt);
+		void addG722Codec(int payloadType, optional<string> profile = std::nullopt);
+
+		[[deprecated("Use addAACCodec")]] inline void
+		addAacCodec(int payloadType, optional<string> profile = std::nullopt) {
+			addAACCodec(payloadType, std::move(profile));
+		};
+	};
+
+	class RTC_CPP_EXPORT Video : public Media {
+	public:
+		Video(string mid = "video", Direction dir = Direction::SendOnly);
+
+		void addVideoCodec(int payloadType, string codec, optional<string> profile = std::nullopt);
+
+		void addH264Codec(int payloadType, optional<string> profile = DEFAULT_H264_VIDEO_PROFILE);
+		void addH265Codec(int payloadType, optional<string> profile = std::nullopt);
+		void addVP8Codec(int payloadType, optional<string> profile = std::nullopt);
+		void addVP9Codec(int payloadType, optional<string> profile = std::nullopt);
+		void addAV1Codec(int payloadType, optional<string> profile = std::nullopt);
+	};
+
+	// Adds RTX payload type mappings for SDP negotiation (codec level only).
+	// For per-stream RTX, also call Media::addRtxSSRC() once SSRCs are known.
+	// Set audio=true to also apply RTX to audio media streams.
+	void addRtx(optional<unsigned int> clockRate = nullopt, bool audio = false);
+
+	bool hasApplication() const;
+	bool hasAudioOrVideo() const;
+	bool hasMid(string_view mid) const;
+
+	int addMedia(Media media);
+	int addMedia(Application application);
+	int addApplication(string mid = "data");
+	int addVideo(string mid = "video", Direction dir = Direction::SendOnly);
+	int addAudio(string mid = "audio", Direction dir = Direction::SendOnly);
+	void clearMedia();
+
+	variant<Media *, Application *> media(int index);
+	variant<const Media *, const Application *> media(int index) const;
+	int mediaCount() const;
+
+	const Application *application() const;
+	Application *application();
+
+	static Type stringToType(const string &typeString);
+	static string typeToString(Type type);
+
+	string sessionId() const;
+
+private:
+	optional<Candidate> defaultCandidate() const;
+	shared_ptr<Entry> createEntry(string mline, string mid, Direction dir);
+	void removeApplication();
+
+	Type mType;
+
+	// Session-level attributes
+	Role mRole;
+	string mUsername;
+	string mSessionId;
+	std::vector<string> mIceOptions;
+	optional<string> mIceUfrag, mIcePwd;
+	optional<CertificateFingerprint> mFingerprint;
+	std::vector<string> mAttributes; // other attributes
+
+	// Entries
+	std::vector<shared_ptr<Entry>> mEntries;
+	shared_ptr<Application> mApplication;
+
+	// Candidates
+	std::vector<Candidate> mCandidates;
+	bool mEnded = false;
+};
+
+RTC_CPP_EXPORT std::ostream &operator<<(std::ostream &out, const Description &description);
+RTC_CPP_EXPORT std::ostream &operator<<(std::ostream &out, Description::Type type);
+RTC_CPP_EXPORT std::ostream &operator<<(std::ostream &out, Description::Role role);
+RTC_CPP_EXPORT std::ostream &operator<<(std::ostream &out, const Description::Direction &direction);
+
+} // namespace rtc
+
+#endif

libdatachannel/include/rtc/frameinfo.hpp

@@ -0,0 +1,44 @@
+/**
+ * Copyright (c) 2019-2020 Paul-Louis Ageneau
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+#ifndef RTC_FRAMEINFO_H
+#define RTC_FRAMEINFO_H
+
+#include "common.hpp"
+
+#include <chrono>
+
+namespace rtc {
+
+struct RTC_CPP_EXPORT FrameInfo {
+	FrameInfo(uint32_t timestamp) : timestamp(timestamp) {};
+	template<typename Period = std::ratio<1>> FrameInfo(std::chrono::duration<double, Period> timestamp) : timestampSeconds(timestamp) {}
+
+	[[deprecated]] FrameInfo(uint8_t payloadType, uint32_t timestamp) : timestamp(timestamp), payloadType(payloadType) {};
+
+	uint32_t timestamp = 0;
+	uint8_t payloadType = 0;
+
+	optional<std::chrono::duration<double>> timestampSeconds;
+
+	bool isKeyFrame = false;	// Set by the application
+
+	/// Absolute capture timestamp in NTP-format (high 32 bits seconds since
+	/// the NTP epoch 1900-01-01 UTC; low 32 bits fractional seconds). When
+	/// set alongside an RtpPacketizationConfig with absCaptureTimeId > 0,
+	/// the RtpPacketizer writes the 8-byte form of the abs-capture-time RTP
+	/// header extension on every packet of this frame. Receivers use this
+	/// to recover the source's capture wallclock and compute glass-to-glass
+	/// latency.
+	/// https://webrtc.googlesource.com/src/+/refs/heads/main/docs/native-code/rtp-hdrext/abs-capture-time
+	optional<uint64_t> absCaptureTimeNtp;
+};
+
+} // namespace rtc
+
+#endif // RTC_FRAMEINFO_H

libdatachannel/include/rtc/global.hpp

@@ -0,0 +1,66 @@
+/**
+ * Copyright (c) 2020-2021 Paul-Louis Ageneau
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+#ifndef RTC_GLOBAL_H
+#define RTC_GLOBAL_H
+
+#include "common.hpp"
+
+#include <chrono>
+#include <future>
+#include <iostream>
+
+namespace rtc {
+
+#ifdef None
+#undef None
+#endif
+
+enum class LogLevel { // Don't change, it must match plog severity
+	None = 0,
+	Fatal = 1,
+	Error = 2,
+	Warning = 3,
+	Info = 4,
+	Debug = 5,
+	Verbose = 6
+};
+
+typedef std::function<void(LogLevel level, string message)> LogCallback;
+
+RTC_CPP_EXPORT void InitLogger(LogLevel level, LogCallback callback = nullptr);
+
+RTC_CPP_EXPORT void SetThreadPoolSize(unsigned int count); // 0: hardware concurrency
+
+struct SctpSettings {
+	// For the following settings, not set means optimized default
+	optional<size_t> recvBufferSize;                // in bytes
+	optional<size_t> sendBufferSize;                // in bytes
+	optional<size_t> maxChunksOnQueue;              // in chunks
+	optional<size_t> initialCongestionWindow;       // in MTUs
+	optional<size_t> maxBurst;                      // in MTUs
+	optional<unsigned int> congestionControlModule; // 0: RFC2581, 1: HSTCP, 2: H-TCP, 3: RTCC
+	optional<std::chrono::milliseconds> delayedSackTime;
+	optional<std::chrono::milliseconds> minRetransmitTimeout;
+	optional<std::chrono::milliseconds> maxRetransmitTimeout;
+	optional<std::chrono::milliseconds> initialRetransmitTimeout;
+	optional<unsigned int> maxRetransmitAttempts;
+	optional<std::chrono::milliseconds> heartbeatInterval;
+};
+
+RTC_CPP_EXPORT void SetSctpSettings(SctpSettings s);
+
+// Optional global preload and cleanup
+RTC_CPP_EXPORT bool Preload();
+RTC_CPP_EXPORT std::shared_future<void> Cleanup();
+
+RTC_CPP_EXPORT std::ostream &operator<<(std::ostream &out, LogLevel level);
+
+} // namespace rtc
+
+#endif

libdatachannel/include/rtc/h264rtpdepacketizer.hpp

@@ -0,0 +1,42 @@
+/**
+ * Copyright (c) 2020 Staz Modrzynski
+ * Copyright (c) 2020 Paul-Louis Ageneau
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+#ifndef RTC_H264_RTP_DEPACKETIZER_H
+#define RTC_H264_RTP_DEPACKETIZER_H
+
+#if RTC_ENABLE_MEDIA
+
+#include "common.hpp"
+#include "message.hpp"
+#include "nalunit.hpp"
+#include "rtp.hpp"
+#include "rtpdepacketizer.hpp"
+
+namespace rtc {
+
+/// RTP depacketization for H264
+class RTC_CPP_EXPORT H264RtpDepacketizer final : public VideoRtpDepacketizer {
+public:
+	using Separator = NalUnit::Separator;
+
+	H264RtpDepacketizer(Separator separator = Separator::StartSequence);
+	~H264RtpDepacketizer();
+
+private:
+	message_ptr reassemble(message_buffer &buffer) override;
+	void addSeparator(binary &frame);
+
+	const NalUnit::Separator mSeparator;
+};
+
+} // namespace rtc
+
+#endif // RTC_ENABLE_MEDIA
+
+#endif /* RTC_H264_RTP_DEPACKETIZER_H */

libdatachannel/include/rtc/h264rtppacketizer.hpp

@@ -0,0 +1,57 @@
+/**
+ * Copyright (c) 2020 Filip Klembara (in2core)
+ * Copyright (c) 2023 Paul-Louis Ageneau
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+#ifndef RTC_H264_RTP_PACKETIZER_H
+#define RTC_H264_RTP_PACKETIZER_H
+
+#if RTC_ENABLE_MEDIA
+
+#include "nalunit.hpp"
+#include "rtppacketizer.hpp"
+
+namespace rtc {
+
+/// RTP packetization for H264
+class RTC_CPP_EXPORT H264RtpPacketizer final : public RtpPacketizer {
+public:
+	using Separator = NalUnit::Separator;
+
+	inline static const uint32_t ClockRate = VideoClockRate;
+	[[deprecated("Use ClockRate")]] inline static const uint32_t defaultClockRate = ClockRate;
+
+	/// Constructs h264 payload packetizer with given RTP configuration.
+	/// @note RTP configuration is used in packetization process which may change some configuration
+	/// properties such as sequence number.
+	/// @param separator NAL unit separator
+	/// @param rtpConfig RTP configuration
+	/// @param maxFragmentSize maximum size of one NALU fragment
+	H264RtpPacketizer(Separator separator, shared_ptr<RtpPacketizationConfig> rtpConfig,
+	                  size_t maxFragmentSize = DefaultMaxFragmentSize);
+
+	// For backward compatibility, do not use
+	[[deprecated]] H264RtpPacketizer(
+	    shared_ptr<RtpPacketizationConfig> rtpConfig,
+	    size_t maxFragmentSize = DefaultMaxFragmentSize);
+
+private:
+	std::vector<binary> fragment(binary data) override;
+	std::vector<NalUnit> splitFrame(const binary &frame);
+
+	const Separator mSeparator;
+	const size_t mMaxFragmentSize;
+};
+
+// For backward compatibility, do not use
+using H264PacketizationHandler [[deprecated("Add H264RtpPacketizer directly")]] = PacketizationHandler;
+
+} // namespace rtc
+
+#endif /* RTC_ENABLE_MEDIA */
+
+#endif /* RTC_H264_RTP_PACKETIZER_H */

libdatachannel/include/rtc/h265nalunit.hpp

@@ -0,0 +1,194 @@
+/**
+ * Copyright (c) 2023 Zita Liao (Dolby)
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+#ifndef RTC_H265_NAL_UNIT_H
+#define RTC_H265_NAL_UNIT_H
+
+#if RTC_ENABLE_MEDIA
+
+#include "common.hpp"
+#include "nalunit.hpp"
+
+#include <cassert>
+#include <vector>
+
+namespace rtc {
+
+#pragma pack(push, 1)
+
+#define H265_FU_HEADER_SIZE 1
+/// Nalu header
+struct RTC_CPP_EXPORT H265NalUnitHeader {
+	/*
+	* nal_unit_header( ) {
+	* forbidden_zero_bit	f(1)
+	* nal_unit_type			u(6)
+	* nuh_layer_id			u(6)
+	* nuh_temporal_id_plus1	u(3)
+	}
+	*/
+	uint8_t _first = 0;  // high byte of header
+	uint8_t _second = 0; // low byte of header
+
+	bool forbiddenBit() const { return _first >> 7; }
+	uint8_t unitType() const { return (_first & 0b0111'1110) >> 1; }
+	uint8_t nuhLayerId() const { return ((_first & 0x1) << 5) | ((_second & 0b1111'1000) >> 3); }
+	uint8_t nuhTempIdPlus1() const { return _second & 0b111; }
+
+	void setForbiddenBit(bool isSet) { _first = (_first & 0x7F) | (isSet << 7); }
+	void setUnitType(uint8_t type) { _first = (_first & 0b1000'0001) | ((type & 0b11'1111) << 1); }
+	void setNuhLayerId(uint8_t nuhLayerId) {
+		_first = (_first & 0b1111'1110) | ((nuhLayerId & 0b10'0000) >> 5);
+		_second = (_second & 0b0000'0111) | ((nuhLayerId & 0b01'1111) << 3);
+	}
+	void setNuhTempIdPlus1(uint8_t nuhTempIdPlus1) {
+		_second = (_second & 0b1111'1000) | (nuhTempIdPlus1 & 0b111);
+	}
+};
+
+/// Nalu fragment header
+struct RTC_CPP_EXPORT H265NalUnitFragmentHeader {
+	/*
+	 * +---------------+
+	 * |0|1|2|3|4|5|6|7|
+	 * +-+-+-+-+-+-+-+-+
+	 * |S|E|  FuType   |
+	 * +---------------+
+	 */
+	uint8_t _first = 0;
+
+	bool isStart() const { return _first >> 7; }
+	bool isEnd() const { return (_first >> 6) & 0x01; }
+	uint8_t unitType() const { return _first & 0b11'1111; }
+
+	void setStart(bool isSet) { _first = (_first & 0x7F) | (isSet << 7); }
+	void setEnd(bool isSet) { _first = (_first & 0b1011'1111) | (isSet << 6); }
+	void setUnitType(uint8_t type) { _first = (_first & 0b1100'0000) | (type & 0b11'1111); }
+};
+
+#pragma pack(pop)
+
+struct H265NalUnitFragment;
+
+/// NAL unit
+struct RTC_CPP_EXPORT H265NalUnit : NalUnit {
+	static std::vector<binary> GenerateFragments(const std::vector<H265NalUnit> &nalus,
+	                                             size_t maxFragmentSize);
+
+	H265NalUnit(const H265NalUnit &unit) = default;
+	H265NalUnit(size_t size, bool includingHeader = true)
+	    : NalUnit(size, includingHeader, NalUnit::Type::H265) {}
+	H265NalUnit(binary &&data) : NalUnit(std::move(data)) {}
+	H265NalUnit() : NalUnit(NalUnit::Type::H265) {}
+
+	template <typename Iterator>
+	H265NalUnit(Iterator begin_, Iterator end_) : NalUnit(begin_, end_) {}
+
+	bool forbiddenBit() const { return header()->forbiddenBit(); }
+	uint8_t unitType() const { return header()->unitType(); }
+	uint8_t nuhLayerId() const { return header()->nuhLayerId(); }
+	uint8_t nuhTempIdPlus1() const { return header()->nuhTempIdPlus1(); }
+
+	binary payload() const {
+		assert(size() >= H265_NAL_HEADER_SIZE);
+		return {begin() + H265_NAL_HEADER_SIZE, end()};
+	}
+
+	void setForbiddenBit(bool isSet) { header()->setForbiddenBit(isSet); }
+	void setUnitType(uint8_t type) { header()->setUnitType(type); }
+	void setNuhLayerId(uint8_t nuhLayerId) { header()->setNuhLayerId(nuhLayerId); }
+	void setNuhTempIdPlus1(uint8_t nuhTempIdPlus1) { header()->setNuhTempIdPlus1(nuhTempIdPlus1); }
+
+	void setPayload(binary payload) {
+		assert(size() >= H265_NAL_HEADER_SIZE);
+		erase(begin() + H265_NAL_HEADER_SIZE, end());
+		insert(end(), payload.begin(), payload.end());
+	}
+
+	std::vector<H265NalUnitFragment> generateFragments(size_t maxFragmentSize) const;
+
+protected:
+	const H265NalUnitHeader *header() const {
+		assert(size() >= H265_NAL_HEADER_SIZE);
+		return reinterpret_cast<const H265NalUnitHeader *>(data());
+	}
+
+	H265NalUnitHeader *header() {
+		assert(size() >= H265_NAL_HEADER_SIZE);
+		return reinterpret_cast<H265NalUnitHeader *>(data());
+	}
+};
+
+/// NAL unit fragment
+struct RTC_CPP_EXPORT H265NalUnitFragment : H265NalUnit {
+	[[deprecated]] static std::vector<shared_ptr<H265NalUnitFragment>> fragmentsFrom(shared_ptr<H265NalUnit> nalu,
+	                                                                  uint16_t maxFragmentSize);
+
+	enum class FragmentType { Start, Middle, End };
+
+	H265NalUnitFragment(FragmentType type, bool forbiddenBit, uint8_t nuhLayerId,
+	                    uint8_t nuhTempIdPlus1, uint8_t unitType, binary data);
+
+	uint8_t unitType() const { return fragmentHeader()->unitType(); }
+
+	binary payload() const {
+		assert(size() >= H265_NAL_HEADER_SIZE + H265_FU_HEADER_SIZE);
+		return {begin() + H265_NAL_HEADER_SIZE + H265_FU_HEADER_SIZE, end()};
+	}
+
+	FragmentType type() const {
+		if (fragmentHeader()->isStart()) {
+			return FragmentType::Start;
+		} else if (fragmentHeader()->isEnd()) {
+			return FragmentType::End;
+		} else {
+			return FragmentType::Middle;
+		}
+	}
+
+	void setUnitType(uint8_t type) { fragmentHeader()->setUnitType(type); }
+
+	void setPayload(binary payload) {
+		assert(size() >= H265_NAL_HEADER_SIZE + H265_FU_HEADER_SIZE);
+		erase(begin() + H265_NAL_HEADER_SIZE + H265_FU_HEADER_SIZE, end());
+		insert(end(), payload.begin(), payload.end());
+	}
+
+	void setFragmentType(FragmentType type);
+
+protected:
+	const uint8_t nal_type_fu = 49;
+
+	H265NalUnitHeader *fragmentIndicator() { return reinterpret_cast<H265NalUnitHeader *>(data()); }
+
+	const H265NalUnitHeader *fragmentIndicator() const {
+		return reinterpret_cast<const H265NalUnitHeader *>(data());
+	}
+
+	H265NalUnitFragmentHeader *fragmentHeader() {
+		return reinterpret_cast<H265NalUnitFragmentHeader *>(data() + H265_NAL_HEADER_SIZE);
+	}
+
+	const H265NalUnitFragmentHeader *fragmentHeader() const {
+		return reinterpret_cast<const H265NalUnitFragmentHeader *>(data() + H265_NAL_HEADER_SIZE);
+	}
+};
+
+class [[deprecated]] RTC_CPP_EXPORT H265NalUnits : public std::vector<shared_ptr<H265NalUnit>> {
+public:
+	static const uint16_t defaultMaximumFragmentSize =
+	    uint16_t(RTC_DEFAULT_MTU - 12 - 8 - 40); // SRTP/UDP/IPv6
+
+	std::vector<shared_ptr<binary>> generateFragments(uint16_t maxFragmentSize);
+};
+
+} // namespace rtc
+
+#endif /* RTC_ENABLE_MEDIA */
+
+#endif /* RTC_NAL_UNIT_H */

libdatachannel/include/rtc/h265rtpdepacketizer.hpp

@@ -0,0 +1,45 @@
+/**
+ * Copyright (c) 2020 Staz Modrzynski
+ * Copyright (c) 2020-2024 Paul-Louis Ageneau
+ * Copyright (c) 2024 Robert Edmonds
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+#ifndef RTC_H265_RTP_DEPACKETIZER_H
+#define RTC_H265_RTP_DEPACKETIZER_H
+
+#if RTC_ENABLE_MEDIA
+
+#include "common.hpp"
+#include "h265nalunit.hpp"
+#include "message.hpp"
+#include "rtp.hpp"
+#include "rtpdepacketizer.hpp"
+
+#include <set>
+
+namespace rtc {
+
+/// RTP depacketization for H265
+class RTC_CPP_EXPORT H265RtpDepacketizer final : public VideoRtpDepacketizer {
+public:
+	using Separator = NalUnit::Separator;
+
+	H265RtpDepacketizer(Separator separator = Separator::StartSequence);
+	~H265RtpDepacketizer();
+
+private:
+	message_ptr reassemble(message_buffer &buffer);
+	void addSeparator(binary &frame);
+
+	const NalUnit::Separator mSeparator;
+};
+
+} // namespace rtc
+
+#endif // RTC_ENABLE_MEDIA
+
+#endif // RTC_H265_RTP_DEPACKETIZER_H

libdatachannel/include/rtc/h265rtppacketizer.hpp

@@ -0,0 +1,55 @@
+/**
+ * Copyright (c) 2023 Zita Liao (Dolby)
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+#ifndef RTC_H265_RTP_PACKETIZER_H
+#define RTC_H265_RTP_PACKETIZER_H
+
+#if RTC_ENABLE_MEDIA
+
+#include "h265nalunit.hpp"
+#include "rtppacketizer.hpp"
+
+namespace rtc {
+
+// RTP packetization for H265
+class RTC_CPP_EXPORT H265RtpPacketizer final : public RtpPacketizer {
+public:
+	using Separator = NalUnit::Separator;
+
+	inline static const uint32_t ClockRate = VideoClockRate;
+	[[deprecated("Use ClockRate")]] inline static const uint32_t defaultClockRate = ClockRate;
+
+	// Constructs h265 payload packetizer with given RTP configuration.
+	// @note RTP configuration is used in packetization process which may change some configuration
+	// properties such as sequence number.
+	// @param separator NAL unit separator
+	// @param rtpConfig  RTP configuration
+	// @param maxFragmentSize maximum size of one NALU fragment
+	H265RtpPacketizer(Separator separator, shared_ptr<RtpPacketizationConfig> rtpConfig,
+	                  size_t maxFragmentSize = DefaultMaxFragmentSize);
+
+	// For backward compatibility, do not use
+	[[deprecated]] H265RtpPacketizer(shared_ptr<RtpPacketizationConfig> rtpConfig,
+	                  size_t maxFragmentSize = DefaultMaxFragmentSize);
+
+private:
+	std::vector<binary> fragment(binary data) override;
+	std::vector<H265NalUnit> splitFrame(const binary &frame);
+
+	const NalUnit::Separator mSeparator;
+	const size_t mMaxFragmentSize;
+};
+
+// For backward compatibility, do not use
+using H265PacketizationHandler [[deprecated("Add H265RtpPacketizer directly")]] = PacketizationHandler;
+
+} // namespace rtc
+
+#endif /* RTC_ENABLE_MEDIA */
+
+#endif /* RTC_H265_RTP_PACKETIZER_H */

libdatachannel/include/rtc/iceudpmuxlistener.hpp

@@ -0,0 +1,47 @@
+/**
+ * Copyright (c) 2025 Alex Potsides
+ * Copyright (c) 2025 Paul-Louis Ageneau
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+#ifndef RTC_ICE_UDP_MUX_LISTENER_H
+#define RTC_ICE_UDP_MUX_LISTENER_H
+
+#include "common.hpp"
+
+namespace rtc {
+
+namespace impl {
+
+struct IceUdpMuxListener;
+
+} // namespace impl
+
+struct IceUdpMuxRequest { // TODO change name
+	string localUfrag;
+	string remoteUfrag;
+	string remoteAddress;
+	uint16_t remotePort;
+};
+
+class RTC_CPP_EXPORT IceUdpMuxListener final : private CheshireCat<impl::IceUdpMuxListener> {
+public:
+	IceUdpMuxListener(uint16_t port, optional<string> bindAddress = nullopt);
+	~IceUdpMuxListener();
+
+	void stop();
+
+	uint16_t port() const;
+
+	void OnUnhandledStunRequest(std::function<void(IceUdpMuxRequest)> callback);
+
+private:
+	using CheshireCat<impl::IceUdpMuxListener>::impl;
+};
+
+} // namespace rtc
+
+#endif

libdatachannel/include/rtc/mediahandler.hpp

@@ -0,0 +1,60 @@
+/**
+ * Copyright (c) 2020 Staz Modrzynski
+ * Copyright (c) 2020 Paul-Louis Ageneau
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+#ifndef RTC_MEDIA_HANDLER_H
+#define RTC_MEDIA_HANDLER_H
+
+#include "common.hpp"
+#include "description.hpp"
+#include "message.hpp"
+
+#include <vector>
+
+namespace rtc {
+
+class RTC_CPP_EXPORT MediaHandler : public std::enable_shared_from_this<MediaHandler> {
+public:
+	MediaHandler();
+	virtual ~MediaHandler();
+
+	/// Called when a media is added or updated
+	/// @param desc Description of the media
+	virtual void media([[maybe_unused]] const Description::Media &desc) {}
+
+	/// Called when there is traffic coming from the peer
+	/// @param messages Incoming messages from the peer, can be modified by the handler
+	/// @param send Send callback to send messages back to the peer
+	virtual void incoming([[maybe_unused]] message_vector &messages, [[maybe_unused]] const message_callback &send) {}
+
+	/// Called when there is traffic that needs to be sent to the peer
+	/// @param messages Outgoing messages to the peer, can be modified by the handler
+	/// @param send Send callback to send messages back to the peer
+	virtual void outgoing([[maybe_unused]] message_vector &messages, [[maybe_unused]] const message_callback &send) {}
+
+	virtual bool requestKeyframe(const std::vector<SSRC>& targetSSRCs, bool retransmit, const message_callback &send);
+	virtual bool requestBitrate(unsigned int bitrate, const message_callback &send);
+
+	void addToChain(shared_ptr<MediaHandler> handler);
+	void setNext(shared_ptr<MediaHandler> handler);
+	shared_ptr<MediaHandler> next();
+	shared_ptr<const MediaHandler> next() const;
+	shared_ptr<MediaHandler> last();             // never null
+	shared_ptr<const MediaHandler> last() const; // never null
+
+	void mediaChain(const Description::Media &desc);
+	void incomingChain(message_vector &messages, const message_callback &send);
+	void outgoingChain(message_vector &messages, const message_callback &send);
+
+private:
+	shared_ptr<MediaHandler> mNext;
+};
+
+} // namespace rtc
+
+#endif // RTC_MEDIA_HANDLER_H

libdatachannel/include/rtc/message.hpp

@@ -0,0 +1,100 @@
+/**
+ * Copyright (c) 2019-2020 Paul-Louis Ageneau
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+#ifndef RTC_MESSAGE_H
+#define RTC_MESSAGE_H
+
+#include "common.hpp"
+#include "frameinfo.hpp"
+#include "reliability.hpp"
+
+#include <functional>
+
+namespace rtc {
+
+struct RTC_CPP_EXPORT Message : binary {
+	enum Type { Binary, String, Control, Reset };
+
+	Message(const Message &message) = default;
+	Message(size_t size, Type type_ = Binary) : binary(size), type(type_) {}
+
+	template <typename Iterator>
+	Message(Iterator begin_, Iterator end_, Type type_ = Binary)
+	    : binary(begin_, end_), type(type_) {}
+
+	Message(binary &&data, Type type_ = Binary) : binary(std::move(data)), type(type_) {}
+
+	Type type;
+	unsigned int stream = 0; // Stream id (SCTP stream or SSRC)
+	unsigned int dscp = 0;   // Differentiated Services Code Point
+	shared_ptr<Reliability> reliability;
+	shared_ptr<FrameInfo> frameInfo;
+};
+
+using message_ptr = shared_ptr<Message>;
+using message_callback = std::function<void(message_ptr message)>;
+using message_vector = std::vector<message_ptr>;
+
+inline size_t message_size_func(const message_ptr &m) {
+	return m->type == Message::Binary || m->type == Message::String ? m->size() : 0;
+}
+
+template <typename Iterator>
+message_ptr make_message(Iterator begin, Iterator end, Message::Type type = Message::Binary,
+                         unsigned int stream = 0, shared_ptr<Reliability> reliability = nullptr) {
+	auto message = std::make_shared<Message>(begin, end, type);
+	message->stream = stream;
+	message->reliability = reliability;
+	return message;
+}
+
+template <typename Iterator>
+message_ptr make_message(Iterator begin, Iterator end, shared_ptr<FrameInfo> frameInfo) {
+	auto message = std::make_shared<Message>(begin, end);
+	message->frameInfo = frameInfo;
+	return message;
+}
+
+// For backward compatibiity, do not use
+template <typename Iterator>
+[[deprecated]] message_ptr make_message(Iterator begin, Iterator end, Message::Type type,
+                         unsigned int stream, shared_ptr<FrameInfo> frameInfo) {
+	auto message = std::make_shared<Message>(begin, end, type);
+	message->stream = stream;
+	message->frameInfo = frameInfo;
+	return message;
+}
+
+RTC_CPP_EXPORT message_ptr make_message(size_t size, Message::Type type = Message::Binary,
+                                        unsigned int stream = 0,
+                                        shared_ptr<Reliability> reliability = nullptr);
+
+RTC_CPP_EXPORT message_ptr make_message(binary &&data, Message::Type type = Message::Binary,
+                                        unsigned int stream = 0,
+                                        shared_ptr<Reliability> reliability = nullptr);
+
+RTC_CPP_EXPORT message_ptr make_message(binary &&data, shared_ptr<FrameInfo> frameInfo);
+
+RTC_CPP_EXPORT message_ptr make_message(size_t size, message_ptr orig);
+
+RTC_CPP_EXPORT message_ptr make_message(message_variant data);
+
+#if RTC_ENABLE_MEDIA
+
+// Reconstructs a message_ptr from an opaque rtcMessage pointer that
+// was allocated by rtcCreateOpaqueMessage().
+message_ptr make_message_from_opaque_ptr(rtcMessage *&&message);
+
+#endif
+
+RTC_CPP_EXPORT message_variant to_variant(Message &&message);
+RTC_CPP_EXPORT message_variant to_variant(const Message &message);
+
+} // namespace rtc
+
+#endif

libdatachannel/include/rtc/nalunit.hpp

@@ -0,0 +1,197 @@
+/**
+ * Copyright (c) 2020 Filip Klembara (in2core)
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+#ifndef RTC_NAL_UNIT_H
+#define RTC_NAL_UNIT_H
+
+#if RTC_ENABLE_MEDIA
+
+#include "common.hpp"
+
+#include <vector>
+#include <cassert>
+
+namespace rtc {
+
+#pragma pack(push, 1)
+
+/// Nalu header
+struct RTC_CPP_EXPORT NalUnitHeader {
+	uint8_t _first = 0;
+
+	bool forbiddenBit() const { return _first >> 7; }
+	uint8_t nri() const { return _first >> 5 & 0x03; }
+	uint8_t idc() const { return _first & 0x60; }
+	uint8_t unitType() const { return _first & 0x1F; }
+
+	void setForbiddenBit(bool isSet) { _first = (_first & 0x7F) | (isSet << 7); }
+	void setNRI(uint8_t nri) { _first = (_first & 0x9F) | ((nri & 0x03) << 5); }
+	void setUnitType(uint8_t type) { _first = (_first & 0xE0) | (type & 0x1F); }
+};
+
+/// Nalu fragment header
+struct RTC_CPP_EXPORT NalUnitFragmentHeader {
+	uint8_t _first = 0;
+
+	bool isStart() const { return _first >> 7; }
+	bool reservedBit6() const { return (_first >> 5) & 0x01; }
+	bool isEnd() const { return (_first >> 6) & 0x01; }
+	uint8_t unitType() const { return _first & 0x1F; }
+
+	void setStart(bool isSet) { _first = (_first & 0x7F) | (isSet << 7); }
+	void setEnd(bool isSet) { _first = (_first & 0xBF) | (isSet << 6); }
+	void setReservedBit6(bool isSet) { _first = (_first & 0xDF) | (isSet << 5); }
+	void setUnitType(uint8_t type) { _first = (_first & 0xE0) | (type & 0x1F); }
+};
+
+#pragma pack(pop)
+
+enum NalUnitStartSequenceMatch {
+	NUSM_noMatch,
+	NUSM_firstZero,
+	NUSM_secondZero,
+	NUSM_thirdZero,
+	NUSM_shortMatch,
+	NUSM_longMatch
+};
+
+static const size_t H264_NAL_HEADER_SIZE = 1;
+static const size_t H265_NAL_HEADER_SIZE = 2;
+
+struct NalUnitFragmentA;
+
+/// NAL unit
+struct RTC_CPP_EXPORT NalUnit : binary {
+	static std::vector<binary> GenerateFragments(const std::vector<NalUnit> &nalus,
+	                                             size_t maxFragmentSize);
+
+	enum class Separator {
+		Length = RTC_NAL_SEPARATOR_LENGTH, // first 4 bytes are NAL unit length
+		LongStartSequence = RTC_NAL_SEPARATOR_LONG_START_SEQUENCE,   // 0x00, 0x00, 0x00, 0x01
+		ShortStartSequence = RTC_NAL_SEPARATOR_SHORT_START_SEQUENCE, // 0x00, 0x00, 0x01
+		StartSequence = RTC_NAL_SEPARATOR_START_SEQUENCE, // LongStartSequence or ShortStartSequence
+	};
+
+	static NalUnitStartSequenceMatch StartSequenceMatchSucc(NalUnitStartSequenceMatch match,
+	                                                 std::byte _byte, Separator separator);
+
+	enum class Type { H264, H265 };
+
+	NalUnit(const NalUnit &unit) = default;
+	NalUnit(size_t size, bool includingHeader = true, Type type = Type::H264)
+	    : binary(size + (includingHeader ? 0
+	                                     : (type == Type::H264 ? H264_NAL_HEADER_SIZE
+	                                                           : H265_NAL_HEADER_SIZE))) {}
+	NalUnit(binary &&data) : binary(std::move(data)) {}
+	NalUnit(Type type = Type::H264)
+	    : binary(type == Type::H264 ? H264_NAL_HEADER_SIZE : H265_NAL_HEADER_SIZE) {}
+	template <typename Iterator> NalUnit(Iterator begin_, Iterator end_) : binary(begin_, end_) {}
+
+	bool forbiddenBit() const { return header()->forbiddenBit(); }
+	uint8_t nri() const { return header()->nri(); }
+	uint8_t unitType() const { return header()->unitType(); }
+
+	binary payload() const {
+		assert(size() >= 1);
+		return {begin() + 1, end()};
+	}
+
+	void setForbiddenBit(bool isSet) { header()->setForbiddenBit(isSet); }
+	void setNRI(uint8_t nri) { header()->setNRI(nri); }
+	void setUnitType(uint8_t type) { header()->setUnitType(type); }
+
+	void setPayload(binary payload) {
+		assert(size() >= 1);
+		erase(begin() + 1, end());
+		insert(end(), payload.begin(), payload.end());
+	}
+
+	std::vector<NalUnitFragmentA> generateFragments(size_t maxFragmentSize) const;
+
+protected:
+	const NalUnitHeader *header() const {
+		assert(size() >= 1);
+		return reinterpret_cast<const NalUnitHeader *>(data());
+	}
+
+	NalUnitHeader *header() {
+		assert(size() >= 1);
+		return reinterpret_cast<NalUnitHeader *>(data());
+	}
+};
+
+/// Nal unit fragment A
+struct RTC_CPP_EXPORT NalUnitFragmentA : NalUnit {
+	// For backward compatibility, do not use
+	[[deprecated]] static std::vector<shared_ptr<NalUnitFragmentA>>
+	fragmentsFrom(shared_ptr<NalUnit> nalu, uint16_t maxFragmentSize);
+
+	enum class FragmentType { Start, Middle, End };
+
+	NalUnitFragmentA(FragmentType type, bool forbiddenBit, uint8_t nri, uint8_t unitType,
+	                 binary data);
+
+	uint8_t unitType() const { return fragmentHeader()->unitType(); }
+
+	binary payload() const {
+		assert(size() >= 2);
+		return {begin() + 2, end()};
+	}
+
+	FragmentType type() const {
+		if (fragmentHeader()->isStart()) {
+			return FragmentType::Start;
+		} else if (fragmentHeader()->isEnd()) {
+			return FragmentType::End;
+		} else {
+			return FragmentType::Middle;
+		}
+	}
+
+	void setUnitType(uint8_t type) { fragmentHeader()->setUnitType(type); }
+
+	void setPayload(binary payload) {
+		assert(size() >= 2);
+		erase(begin() + 2, end());
+		insert(end(), payload.begin(), payload.end());
+	}
+
+	void setFragmentType(FragmentType type);
+
+protected:
+	const uint8_t nal_type_fu_A = 28;
+
+	NalUnitHeader *fragmentIndicator() { return reinterpret_cast<NalUnitHeader *>(data()); }
+
+	const NalUnitHeader *fragmentIndicator() const {
+		return reinterpret_cast<const NalUnitHeader *>(data());
+	}
+
+	NalUnitFragmentHeader *fragmentHeader() {
+		return reinterpret_cast<NalUnitFragmentHeader *>(fragmentIndicator() + 1);
+	}
+
+	const NalUnitFragmentHeader *fragmentHeader() const {
+		return reinterpret_cast<const NalUnitFragmentHeader *>(fragmentIndicator() + 1);
+	}
+};
+
+// For backward compatibility, do not use
+class [[deprecated]] RTC_CPP_EXPORT NalUnits : public std::vector<shared_ptr<NalUnit>> {
+public:
+	static const uint16_t defaultMaximumFragmentSize =
+	    uint16_t(RTC_DEFAULT_MTU - 12 - 8 - 40); // SRTP/UDP/IPv6
+
+	std::vector<shared_ptr<binary>> generateFragments(uint16_t maxFragmentSize);
+};
+
+} // namespace rtc
+
+#endif /* RTC_ENABLE_MEDIA */
+
+#endif /* RTC_NAL_UNIT_H */

libdatachannel/include/rtc/pacinghandler.hpp

@@ -0,0 +1,61 @@
+/**
+ * Copyright (c) 2024 Sean DuBois <sean@siobud.com>
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+#ifndef RTC_PACING_HANDLER_H
+#define RTC_PACING_HANDLER_H
+
+#if RTC_ENABLE_MEDIA
+
+#include "mediahandler.hpp"
+#include "utils.hpp"
+
+#include <atomic>
+#include <queue>
+
+namespace rtc {
+
+// Paced sending of RTP packets. It takes a stream of RTP packets that can have an uneven bitrate
+// and delivers them in a smoother manner by sending a fixed size of them on an interval
+class RTC_CPP_EXPORT PacingHandler : public MediaHandler {
+public:
+	PacingHandler(double bitsPerSecond, std::chrono::milliseconds sendInterval, size_t maxQueueSize=0);
+
+	void setBitrate(double bitsPerSecond);
+
+	void setMaxQueueAmount(size_t maxQueueAmount);
+
+	void onOverflow(std::function<void()> callback);
+
+	void outgoing(message_vector &messages, const message_callback &send) override;
+
+private:
+	std::atomic<bool> mHaveScheduled = false;
+
+	double mBytesPerSecond;
+	double mBudget;
+
+	std::chrono::milliseconds mSendInterval;
+	std::chrono::time_point<std::chrono::high_resolution_clock> mLastRun;
+
+	size_t mMaxQueueAmount;
+
+	std::mutex mParamsMutex;
+	std::mutex mMutex;
+	std::queue<message_ptr> mRtpBuffer;
+
+	synchronized_callback<> mOverflowCallback;
+
+	void schedule(const message_callback &send, std::chrono::milliseconds scheduleInterval);
+	void run(const message_callback &send);
+};
+
+} // namespace rtc
+
+#endif // RTC_ENABLE_MEDIA
+
+#endif // RTC_PACING_HANDLER_H

libdatachannel/include/rtc/peerconnection.hpp

@@ -0,0 +1,142 @@
+/**
+ * Copyright (c) 2019 Paul-Louis Ageneau
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+#ifndef RTC_PEER_CONNECTION_H
+#define RTC_PEER_CONNECTION_H
+
+#include "candidate.hpp"
+#include "common.hpp"
+#include "configuration.hpp"
+#include "datachannel.hpp"
+#include "description.hpp"
+#include "reliability.hpp"
+#include "track.hpp"
+
+#include <chrono>
+#include <functional>
+
+namespace rtc {
+
+namespace impl {
+
+struct PeerConnection;
+
+}
+
+struct RTC_CPP_EXPORT DataChannelInit {
+	Reliability reliability = {};
+	bool negotiated = false;
+	optional<uint16_t> id = nullopt;
+	string protocol = "";
+};
+
+struct RTC_CPP_EXPORT LocalDescriptionInit {
+    optional<string> iceUfrag;
+    optional<string> icePwd;
+};
+
+class RTC_CPP_EXPORT PeerConnection final : CheshireCat<impl::PeerConnection> {
+public:
+	enum class State : int {
+		New = RTC_NEW,
+		Connecting = RTC_CONNECTING,
+		Connected = RTC_CONNECTED,
+		Disconnected = RTC_DISCONNECTED,
+		Failed = RTC_FAILED,
+		Closed = RTC_CLOSED
+	};
+
+	enum class IceState : int {
+		New = RTC_ICE_NEW,
+		Checking = RTC_ICE_CHECKING,
+		Connected = RTC_ICE_CONNECTED,
+		Completed = RTC_ICE_COMPLETED,
+		Failed = RTC_ICE_FAILED,
+		Disconnected = RTC_ICE_DISCONNECTED,
+		Closed = RTC_ICE_CLOSED
+	};
+
+	enum class GatheringState : int {
+		New = RTC_GATHERING_NEW,
+		InProgress = RTC_GATHERING_INPROGRESS,
+		Complete = RTC_GATHERING_COMPLETE
+	};
+
+	enum class SignalingState : int {
+		Stable = RTC_SIGNALING_STABLE,
+		HaveLocalOffer = RTC_SIGNALING_HAVE_LOCAL_OFFER,
+		HaveRemoteOffer = RTC_SIGNALING_HAVE_REMOTE_OFFER,
+		HaveLocalPranswer = RTC_SIGNALING_HAVE_LOCAL_PRANSWER,
+		HaveRemotePranswer = RTC_SIGNALING_HAVE_REMOTE_PRANSWER,
+	};
+
+	PeerConnection();
+	PeerConnection(Configuration config);
+	~PeerConnection();
+
+	void close();
+
+	const Configuration *config() const;
+	State state() const;
+	IceState iceState() const;
+	GatheringState gatheringState() const;
+	SignalingState signalingState() const;
+	bool negotiationNeeded() const;
+	bool hasMedia() const;
+	optional<Description> localDescription() const;
+	optional<Description> remoteDescription() const;
+	size_t remoteMaxMessageSize() const;
+	optional<string> localAddress() const;
+	optional<string> remoteAddress() const;
+	uint16_t maxDataChannelId() const;
+	bool getSelectedCandidatePair(Candidate *local, Candidate *remote);
+
+	void setLocalDescription(Description::Type type = Description::Type::Unspec, LocalDescriptionInit init = {});
+	void gatherLocalCandidates(std::vector<IceServer> additionalIceServers = {});
+	void setRemoteDescription(Description description);
+	void addRemoteCandidate(Candidate candidate);
+
+	// For specific use cases only
+	Description createOffer();
+	Description createAnswer();
+
+	void setMediaHandler(shared_ptr<MediaHandler> handler);
+	shared_ptr<MediaHandler> getMediaHandler();
+
+	[[nodiscard]] shared_ptr<DataChannel> createDataChannel(string label,
+	                                                        DataChannelInit init = {});
+	void onDataChannel(std::function<void(std::shared_ptr<DataChannel> dataChannel)> callback);
+
+	[[nodiscard]] shared_ptr<Track> addTrack(Description::Media description);
+	void onTrack(std::function<void(std::shared_ptr<Track> track)> callback);
+
+	void onLocalDescription(std::function<void(Description description)> callback);
+	void onLocalCandidate(std::function<void(Candidate candidate)> callback);
+	void onStateChange(std::function<void(State state)> callback);
+	void onIceStateChange(std::function<void(IceState state)> callback);
+	void onGatheringStateChange(std::function<void(GatheringState state)> callback);
+	void onSignalingStateChange(std::function<void(SignalingState state)> callback);
+
+	void resetCallbacks();
+	CertificateFingerprint remoteFingerprint();
+
+	// Stats
+	void clearStats();
+	size_t bytesSent();
+	size_t bytesReceived();
+	optional<std::chrono::milliseconds> rtt();
+};
+
+RTC_CPP_EXPORT std::ostream &operator<<(std::ostream &out, PeerConnection::State state);
+RTC_CPP_EXPORT std::ostream &operator<<(std::ostream &out, PeerConnection::IceState state);
+RTC_CPP_EXPORT std::ostream &operator<<(std::ostream &out, PeerConnection::GatheringState state);
+RTC_CPP_EXPORT std::ostream &operator<<(std::ostream &out, PeerConnection::SignalingState state);
+
+} // namespace rtc
+
+#endif

libdatachannel/include/rtc/plihandler.hpp

@@ -0,0 +1,41 @@
+/**
+ * Copyright (c) 2023 Arda Cinar
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+#ifndef RTC_PLI_RESPONDER_H
+#define RTC_PLI_RESPONDER_H
+
+#if RTC_ENABLE_MEDIA
+
+#include "mediahandler.hpp"
+#include "utils.hpp"
+
+#include <unordered_map>
+
+namespace rtc {
+
+/// Responds to PLI and FIR messages sent by the receiver. The sender should respond to these
+/// messages by sending an intra.
+class RTC_CPP_EXPORT PliHandler final : public MediaHandler {
+    rtc::synchronized_callback<> mOnPli;
+
+public:
+	/// Constructs the PLIResponder object to notify whenever a new intra frame is requested
+	/// @param onPli The callback that gets called whenever an intra frame is requested by the receiver
+    PliHandler(std::function<void(void)> onPli);
+
+	void incoming(message_vector &messages, const message_callback &send) override;
+
+protected:
+	std::unordered_map<uint64_t, uint8_t> mFirSSRCSeqNumberMap;
+};
+
+}
+
+#endif // RTC_ENABLE_MEDIA
+
+#endif // RTC_PLI_RESPONDER_H

libdatachannel/include/rtc/reliability.hpp

@@ -0,0 +1,43 @@
+/**
+ * Copyright (c) 2019 Paul-Louis Ageneau
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+#ifndef RTC_RELIABILITY_H
+#define RTC_RELIABILITY_H
+
+#include "common.hpp"
+
+#include <chrono>
+
+namespace rtc {
+
+struct Reliability {
+	// It true, the channel does not enforce message ordering and out-of-order delivery is allowed
+	bool unordered = false;
+
+	// If both maxPacketLifeTime or maxRetransmits are unset, the channel is reliable.
+	// If either maxPacketLifeTime or maxRetransmits is set, the channel is unreliable.
+	// (The settings are exclusive so both maxPacketLifetime and maxRetransmits must not be set.)
+
+	// Time window during which transmissions and retransmissions may occur
+	optional<std::chrono::milliseconds> maxPacketLifeTime;
+
+	// Maximum number of retransmissions that are attempted
+	optional<unsigned int> maxRetransmits;
+
+	// For backward compatibility, do not use
+	enum class Type { Reliable = 0, Rexmit, Timed };
+	union {
+		Type typeDeprecated = Type::Reliable;
+		[[deprecated("Use maxPacketLifeTime or maxRetransmits")]] Type type;
+	};
+	variant<int, std::chrono::milliseconds> rexmit = 0;
+};
+
+} // namespace rtc
+
+#endif

libdatachannel/include/rtc/rembhandler.hpp

@@ -0,0 +1,35 @@
+/**
+ * Copyright (c) 2024 Vladimir Voronin
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+#ifndef RTC_REMB_RESPONDER_H
+#define RTC_REMB_RESPONDER_H
+
+#if RTC_ENABLE_MEDIA
+
+#include "mediahandler.hpp"
+#include "utils.hpp"
+
+namespace rtc {
+
+/// Responds to REMB messages sent by the receiver.
+class RTC_CPP_EXPORT RembHandler final : public MediaHandler {
+    rtc::synchronized_callback<unsigned int> mOnRemb;
+
+public:
+	/// Constructs the RembResponder object to notify whenever a bitrate
+	/// @param onRemb The callback that gets called whenever a bitrate by the receiver
+    RembHandler(std::function<void(unsigned int)> onRemb);
+
+	void incoming(message_vector &messages, const message_callback &send) override;
+};
+
+}
+
+#endif // RTC_ENABLE_MEDIA
+
+#endif // RTC_REMB_RESPONDER_H

libdatachannel/include/rtc/rtc.h

@@ -0,0 +1,581 @@
+/**
+ * Copyright (c) 2019-2021 Paul-Louis Ageneau
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+#ifndef RTC_C_API
+#define RTC_C_API
+
+#include "version.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#include <stdbool.h>
+#include <stdint.h>
+
+#ifdef RTC_STATIC
+#define RTC_C_EXPORT
+#else // dynamic library
+#ifdef _WIN32
+#ifdef RTC_EXPORTS
+#define RTC_C_EXPORT __declspec(dllexport) // building the library
+#else
+#define RTC_C_EXPORT __declspec(dllimport) // using the library
+#endif
+#else // not WIN32
+#define RTC_C_EXPORT
+#endif
+#endif
+
+#ifndef RTC_ENABLE_WEBSOCKET
+#define RTC_ENABLE_WEBSOCKET 1
+#endif
+
+#ifndef RTC_ENABLE_MEDIA
+#define RTC_ENABLE_MEDIA 1
+#endif
+
+#define RTC_DEFAULT_MTU 1280 // IPv6 minimum guaranteed MTU
+
+#if RTC_ENABLE_MEDIA
+#define RTC_DEFAULT_MAX_FRAGMENT_SIZE ((uint16_t)(RTC_DEFAULT_MTU - 12 - 8 - 40)) // SRTP/UDP/IPv6
+#define RTC_DEFAULT_MAX_STORED_PACKET_COUNT 512
+// Deprecated, do not use
+#define RTC_DEFAULT_MAXIMUM_FRAGMENT_SIZE RTC_DEFAULT_MAX_FRAGMENT_SIZE
+#define RTC_DEFAULT_MAXIMUM_PACKET_COUNT_FOR_NACK_CACHE RTC_DEFAULT_MAX_STORED_PACKET_COUNT
+#endif
+
+#ifdef _WIN32
+#ifdef CAPI_STDCALL
+#define RTC_API __stdcall
+#else
+#define RTC_API
+#endif
+#else // not WIN32
+#define RTC_API
+#endif
+
+#if defined(__GNUC__) || defined(__clang__)
+#define RTC_DEPRECATED __attribute__((deprecated))
+#elif defined(_MSC_VER)
+#define RTC_DEPRECATED __declspec(deprecated)
+#else
+#define DEPRECATED
+#endif
+
+// libdatachannel C API
+
+typedef enum {
+	RTC_NEW = 0,
+	RTC_CONNECTING = 1,
+	RTC_CONNECTED = 2,
+	RTC_DISCONNECTED = 3,
+	RTC_FAILED = 4,
+	RTC_CLOSED = 5
+} rtcState;
+
+typedef enum {
+	RTC_ICE_NEW = 0,
+	RTC_ICE_CHECKING = 1,
+	RTC_ICE_CONNECTED = 2,
+	RTC_ICE_COMPLETED = 3,
+	RTC_ICE_FAILED = 4,
+	RTC_ICE_DISCONNECTED = 5,
+	RTC_ICE_CLOSED = 6
+} rtcIceState;
+
+typedef enum {
+	RTC_GATHERING_NEW = 0,
+	RTC_GATHERING_INPROGRESS = 1,
+	RTC_GATHERING_COMPLETE = 2
+} rtcGatheringState;
+
+typedef enum {
+	RTC_SIGNALING_STABLE = 0,
+	RTC_SIGNALING_HAVE_LOCAL_OFFER = 1,
+	RTC_SIGNALING_HAVE_REMOTE_OFFER = 2,
+	RTC_SIGNALING_HAVE_LOCAL_PRANSWER = 3,
+	RTC_SIGNALING_HAVE_REMOTE_PRANSWER = 4,
+} rtcSignalingState;
+
+typedef enum { // Don't change, it must match plog severity
+	RTC_LOG_NONE = 0,
+	RTC_LOG_FATAL = 1,
+	RTC_LOG_ERROR = 2,
+	RTC_LOG_WARNING = 3,
+	RTC_LOG_INFO = 4,
+	RTC_LOG_DEBUG = 5,
+	RTC_LOG_VERBOSE = 6
+} rtcLogLevel;
+
+typedef enum {
+	RTC_CERTIFICATE_DEFAULT = 0, // ECDSA
+	RTC_CERTIFICATE_ECDSA = 1,
+	RTC_CERTIFICATE_RSA = 2,
+} rtcCertificateType;
+
+typedef enum {
+	// video
+	RTC_CODEC_H264 = 0,
+	RTC_CODEC_VP8 = 1,
+	RTC_CODEC_VP9 = 2,
+	RTC_CODEC_H265 = 3,
+	RTC_CODEC_AV1 = 4,
+
+	// audio
+	RTC_CODEC_OPUS = 128,
+	RTC_CODEC_PCMU = 129,
+	RTC_CODEC_PCMA = 130,
+	RTC_CODEC_AAC = 131,
+	RTC_CODEC_G722 = 132,
+} rtcCodec;
+
+typedef enum {
+	RTC_DIRECTION_UNKNOWN = 0,
+	RTC_DIRECTION_SENDONLY = 1,
+	RTC_DIRECTION_RECVONLY = 2,
+	RTC_DIRECTION_SENDRECV = 3,
+	RTC_DIRECTION_INACTIVE = 4
+} rtcDirection;
+
+typedef enum { RTC_TRANSPORT_POLICY_ALL = 0, RTC_TRANSPORT_POLICY_RELAY = 1 } rtcTransportPolicy;
+
+#define RTC_ERR_SUCCESS 0
+#define RTC_ERR_INVALID -1   // invalid argument
+#define RTC_ERR_FAILURE -2   // runtime error
+#define RTC_ERR_NOT_AVAIL -3 // element not available
+#define RTC_ERR_TOO_SMALL -4 // buffer too small
+
+typedef void(RTC_API *rtcLogCallbackFunc)(rtcLogLevel level, const char *message);
+typedef void(RTC_API *rtcDescriptionCallbackFunc)(int pc, const char *sdp, const char *type,
+                                                  void *ptr);
+typedef void(RTC_API *rtcCandidateCallbackFunc)(int pc, const char *cand, const char *mid,
+                                                void *ptr);
+typedef void(RTC_API *rtcStateChangeCallbackFunc)(int pc, rtcState state, void *ptr);
+typedef void(RTC_API *rtcIceStateChangeCallbackFunc)(int pc, rtcIceState state, void *ptr);
+typedef void(RTC_API *rtcGatheringStateCallbackFunc)(int pc, rtcGatheringState state, void *ptr);
+typedef void(RTC_API *rtcSignalingStateCallbackFunc)(int pc, rtcSignalingState state, void *ptr);
+typedef void(RTC_API *rtcDataChannelCallbackFunc)(int pc, int dc, void *ptr);
+typedef void(RTC_API *rtcTrackCallbackFunc)(int pc, int tr, void *ptr);
+typedef void(RTC_API *rtcOpenCallbackFunc)(int id, void *ptr);
+typedef void(RTC_API *rtcClosedCallbackFunc)(int id, void *ptr);
+typedef void(RTC_API *rtcErrorCallbackFunc)(int id, const char *error, void *ptr);
+typedef void(RTC_API *rtcMessageCallbackFunc)(int id, const char *message, int size, void *ptr);
+typedef void *(RTC_API *rtcInterceptorCallbackFunc)(int pc, const char *message, int size,
+                                                    void *ptr);
+typedef void(RTC_API *rtcBufferedAmountLowCallbackFunc)(int id, void *ptr);
+typedef void(RTC_API *rtcAvailableCallbackFunc)(int id, void *ptr);
+typedef void(RTC_API *rtcPliHandlerCallbackFunc)(int tr, void *ptr);
+typedef void(RTC_API *rtcRembHandlerCallbackFunc)(int tr, unsigned int bitrate, void *ptr);
+
+typedef struct {
+	uint32_t timestamp;
+	uint8_t payloadType;
+	double timestampSeconds; // negative means not available
+} rtcFrameInfo;
+
+typedef void(RTC_API *rtcFrameCallbackFunc)(int tr, const char *data, int size,
+                                            const rtcFrameInfo *info, void *ptr);
+
+// Log
+
+// NULL cb on the first call will log to stdout
+RTC_C_EXPORT void rtcInitLogger(rtcLogLevel level, rtcLogCallbackFunc cb);
+
+// User pointer
+RTC_C_EXPORT void rtcSetUserPointer(int id, void *ptr);
+RTC_C_EXPORT void *rtcGetUserPointer(int i);
+
+// PeerConnection
+
+typedef struct {
+	const char **iceServers;
+	int iceServersCount;
+	const char *proxyServer; // libnice only
+	const char *bindAddress; // libjuice only, NULL means any
+	rtcCertificateType certificateType;
+	const char *certificatePemFile; // NULL for autogenerated certificate
+	const char *keyPemFile;         // NULL for autogenerated certificate
+	const char *keyPemPass;         // NULL if no pass
+	rtcTransportPolicy iceTransportPolicy;
+	bool enableIceTcp;
+	bool enableIceUdpMux; // libjuice only
+	bool disableAutoNegotiation;
+	bool forceMediaTransport;
+	uint16_t portRangeBegin; // 0 means automatic
+	uint16_t portRangeEnd;   // 0 means automatic
+	int mtu;                 // <= 0 means automatic
+	int maxMessageSize;      // <= 0 means default
+} rtcConfiguration;
+
+RTC_C_EXPORT int rtcCreatePeerConnection(const rtcConfiguration *config); // returns pc id
+RTC_C_EXPORT int rtcClosePeerConnection(int pc);
+RTC_C_EXPORT int rtcDeletePeerConnection(int pc);
+
+RTC_C_EXPORT int rtcSetLocalDescriptionCallback(int pc, rtcDescriptionCallbackFunc cb);
+RTC_C_EXPORT int rtcSetLocalCandidateCallback(int pc, rtcCandidateCallbackFunc cb);
+RTC_C_EXPORT int rtcSetStateChangeCallback(int pc, rtcStateChangeCallbackFunc cb);
+RTC_C_EXPORT int rtcSetIceStateChangeCallback(int pc, rtcIceStateChangeCallbackFunc cb);
+RTC_C_EXPORT int rtcSetGatheringStateChangeCallback(int pc, rtcGatheringStateCallbackFunc cb);
+RTC_C_EXPORT int rtcSetSignalingStateChangeCallback(int pc, rtcSignalingStateCallbackFunc cb);
+
+RTC_C_EXPORT int rtcSetLocalDescription(int pc, const char *type); // type may be NULL
+RTC_C_EXPORT int rtcSetRemoteDescription(int pc, const char *sdp, const char *type);
+RTC_C_EXPORT int rtcAddRemoteCandidate(int pc, const char *cand, const char *mid);
+
+RTC_C_EXPORT int rtcGetLocalDescription(int pc, char *buffer, int size);
+RTC_C_EXPORT int rtcGetRemoteDescription(int pc, char *buffer, int size);
+
+RTC_C_EXPORT int rtcGetLocalDescriptionType(int pc, char *buffer, int size);
+RTC_C_EXPORT int rtcGetRemoteDescriptionType(int pc, char *buffer, int size);
+
+// For specific use cases only
+RTC_C_EXPORT int rtcCreateOffer(int pc, char *buffer, int size);
+RTC_C_EXPORT int rtcCreateAnswer(int pc, char *buffer, int size);
+
+RTC_C_EXPORT int rtcGetLocalAddress(int pc, char *buffer, int size);
+RTC_C_EXPORT int rtcGetRemoteAddress(int pc, char *buffer, int size);
+
+RTC_C_EXPORT int rtcGetSelectedCandidatePair(int pc, char *local, int localSize, char *remote,
+                                             int remoteSize);
+
+RTC_C_EXPORT bool rtcIsNegotiationNeeded(int pc);
+
+RTC_C_EXPORT int rtcGetMaxDataChannelStream(int pc);
+RTC_C_EXPORT int rtcGetRemoteMaxMessageSize(int pc);
+
+// DataChannel, Track, and WebSocket common API
+
+RTC_C_EXPORT int rtcSetOpenCallback(int id, rtcOpenCallbackFunc cb);
+RTC_C_EXPORT int rtcSetClosedCallback(int id, rtcClosedCallbackFunc cb);
+RTC_C_EXPORT int rtcSetErrorCallback(int id, rtcErrorCallbackFunc cb);
+RTC_C_EXPORT int rtcSetMessageCallback(int id, rtcMessageCallbackFunc cb);
+RTC_C_EXPORT int rtcSendMessage(int id, const char *data, int size);
+RTC_C_EXPORT int rtcClose(int id);
+RTC_C_EXPORT int rtcDelete(int id);
+RTC_C_EXPORT bool rtcIsOpen(int id);
+RTC_C_EXPORT bool rtcIsClosed(int id);
+
+RTC_C_EXPORT int rtcMaxMessageSize(int id);
+RTC_C_EXPORT int rtcGetBufferedAmount(int id); // total size buffered to send
+RTC_C_EXPORT int rtcSetBufferedAmountLowThreshold(int id, int amount);
+RTC_C_EXPORT int rtcSetBufferedAmountLowCallback(int id, rtcBufferedAmountLowCallbackFunc cb);
+
+// DataChannel, Track, and WebSocket common extended API
+
+RTC_C_EXPORT int rtcGetAvailableAmount(int id); // total size available to receive
+RTC_C_EXPORT int rtcSetAvailableCallback(int id, rtcAvailableCallbackFunc cb);
+RTC_C_EXPORT int rtcReceiveMessage(int id, char *buffer, int *size);
+
+// DataChannel
+
+typedef struct {
+	bool unordered;
+	bool unreliable;
+	unsigned int maxPacketLifeTime; // ignored if reliable
+	unsigned int maxRetransmits;    // ignored if reliable
+} rtcReliability;
+
+typedef struct {
+	rtcReliability reliability;
+	const char *protocol; // empty string if NULL
+	bool negotiated;
+	bool manualStream;
+	uint16_t stream; // numeric ID 0-65534, ignored if manualStream is false
+} rtcDataChannelInit;
+
+RTC_C_EXPORT int rtcSetDataChannelCallback(int pc, rtcDataChannelCallbackFunc cb);
+RTC_C_EXPORT int rtcCreateDataChannel(int pc, const char *label); // returns dc id
+RTC_C_EXPORT int rtcCreateDataChannelEx(int pc, const char *label,
+                                        const rtcDataChannelInit *init); // returns dc id
+RTC_C_EXPORT int rtcDeleteDataChannel(int dc);
+
+RTC_C_EXPORT int rtcGetDataChannelStream(int dc);
+RTC_C_EXPORT int rtcGetDataChannelLabel(int dc, char *buffer, int size);
+RTC_C_EXPORT int rtcGetDataChannelProtocol(int dc, char *buffer, int size);
+RTC_C_EXPORT int rtcGetDataChannelReliability(int dc, rtcReliability *reliability);
+
+// Track
+
+typedef struct {
+	rtcDirection direction;
+	rtcCodec codec;
+	int payloadType;
+	uint32_t ssrc;
+	const char *mid;
+	const char *name;    // optional
+	const char *msid;    // optional
+	const char *trackId; // optional, track ID used in MSID
+	const char *profile; // optional, codec profile
+} rtcTrackInit;
+
+RTC_C_EXPORT int rtcSetTrackCallback(int pc, rtcTrackCallbackFunc cb);
+RTC_C_EXPORT int rtcAddTrack(int pc, const char *mediaDescriptionSdp); // returns tr id
+RTC_C_EXPORT int rtcAddTrackEx(int pc, const rtcTrackInit *init);      // returns tr id
+RTC_C_EXPORT int rtcDeleteTrack(int tr);
+
+RTC_C_EXPORT int rtcGetTrackDescription(int tr, char *buffer, int size);
+RTC_C_EXPORT int rtcGetTrackMid(int tr, char *buffer, int size);
+RTC_C_EXPORT int rtcGetTrackDirection(int tr, rtcDirection *direction);
+
+RTC_C_EXPORT int rtcRequestKeyframe(int tr);
+RTC_C_EXPORT int rtcRequestBitrate(int tr, unsigned int bitrate);
+
+RTC_C_EXPORT int rtcSetFrameCallback(int tr, rtcFrameCallbackFunc cb);
+
+#if RTC_ENABLE_MEDIA
+
+// Media
+
+// Define how OBUs are packetizied in a AV1 Sample
+typedef enum {
+	RTC_OBU_PACKETIZED_OBU = 0,
+	RTC_OBU_PACKETIZED_TEMPORAL_UNIT = 1,
+} rtcObuPacketization;
+
+// Define how NAL units are separated in a H264/H265 sample
+typedef enum {
+	RTC_NAL_SEPARATOR_LENGTH = 0,               // first 4 bytes are NAL unit length
+	RTC_NAL_SEPARATOR_LONG_START_SEQUENCE = 1,  // 0x00, 0x00, 0x00, 0x01
+	RTC_NAL_SEPARATOR_SHORT_START_SEQUENCE = 2, // 0x00, 0x00, 0x01
+	RTC_NAL_SEPARATOR_START_SEQUENCE = 3,       // long or short start sequence
+} rtcNalUnitSeparator;
+
+typedef struct {
+	uint32_t ssrc;
+	const char *cname;
+	uint8_t payloadType;
+	uint32_t clockRate;
+	uint16_t sequenceNumber;
+	uint32_t timestamp;
+
+	// H264, H265, AV1
+	uint16_t maxFragmentSize; // Maximum fragment size, 0 means default
+
+	// H264/H265 only
+	rtcNalUnitSeparator nalSeparator; // NAL unit separator
+
+	// AV1 only
+	rtcObuPacketization obuPacketization; // OBU paketization for AV1 samples
+
+	uint8_t playoutDelayId;
+	uint16_t playoutDelayMin;
+	uint16_t playoutDelayMax;
+
+	uint8_t colorSpaceId;
+	uint8_t colorChromaSitingHorz;
+	uint8_t colorChromaSitingVert;
+	uint8_t colorRange;
+	uint8_t colorPrimaries;
+	uint8_t colorTransfer;
+	uint8_t colorMatrix;
+} rtcPacketizerInit;
+
+// Deprecated, do not use
+typedef rtcPacketizerInit rtcPacketizationHandlerInit;
+
+typedef struct {
+	uint32_t ssrc;
+	const char *name;    // optional
+	const char *msid;    // optional
+	const char *trackId; // optional, track ID used in MSID
+} rtcSsrcForTypeInit;
+
+// Opaque type used (via rtcMessage*) to reference an rtc::Message
+typedef void *rtcMessage;
+
+// Allocate a new opaque message.
+// Must be explicitly freed by rtcDeleteOpaqueMessage() unless
+// explicitly returned by a media interceptor callback;
+RTC_C_EXPORT rtcMessage *rtcCreateOpaqueMessage(void *data, int size);
+RTC_C_EXPORT void rtcDeleteOpaqueMessage(rtcMessage *msg);
+
+// Set MediaInterceptor on peer connection
+RTC_C_EXPORT int rtcSetMediaInterceptorCallback(int id, rtcInterceptorCallbackFunc cb);
+
+// Set a packetizer on track
+RTC_C_EXPORT int rtcSetH264Packetizer(int tr, const rtcPacketizerInit *init);
+RTC_C_EXPORT int rtcSetH265Packetizer(int tr, const rtcPacketizerInit *init);
+RTC_C_EXPORT int rtcSetAV1Packetizer(int tr, const rtcPacketizerInit *init);
+RTC_C_EXPORT int rtcSetVP8Packetizer(int tr, const rtcPacketizerInit *init);
+RTC_C_EXPORT int rtcSetVP9Packetizer(int tr, const rtcPacketizerInit *init);
+RTC_C_EXPORT int rtcSetOpusPacketizer(int tr, const rtcPacketizerInit *init);
+RTC_C_EXPORT int rtcSetAACPacketizer(int tr, const rtcPacketizerInit *init);
+RTC_C_EXPORT int rtcSetPCMUPacketizer(int tr, const rtcPacketizerInit *init);
+RTC_C_EXPORT int rtcSetPCMAPacketizer(int tr, const rtcPacketizerInit *init);
+RTC_C_EXPORT int rtcSetG722Packetizer(int tr, const rtcPacketizerInit *init);
+
+// Set a depacketizer on track
+RTC_C_EXPORT int rtcSetH264Depacketizer(int tr, rtcNalUnitSeparator nalSeparator);
+RTC_C_EXPORT int rtcSetH265Depacketizer(int tr, rtcNalUnitSeparator nalSeparator);
+RTC_C_EXPORT int rtcSetVP8Depacketizer(int tr);
+RTC_C_EXPORT int rtcSetVP9Depacketizer(int tr);
+RTC_C_EXPORT int rtcSetOpusDepacketizer(int tr);
+RTC_C_EXPORT int rtcSetAACDepacketizer(int tr);
+RTC_C_EXPORT int rtcSetPCMUDepacketizer(int tr);
+RTC_C_EXPORT int rtcSetPCMADepacketizer(int tr);
+RTC_C_EXPORT int rtcSetG722Depacketizer(int tr);
+
+// Deprecated, do not use
+RTC_DEPRECATED static inline int
+rtcSetH264PacketizationHandler(int tr, const rtcPacketizationHandlerInit *init) {
+	return rtcSetH264Packetizer(tr, init);
+}
+RTC_DEPRECATED static inline int
+rtcSetH265PacketizationHandler(int tr, const rtcPacketizationHandlerInit *init) {
+	return rtcSetH265Packetizer(tr, init);
+}
+RTC_DEPRECATED static inline int
+rtcSetAV1PacketizationHandler(int tr, const rtcPacketizationHandlerInit *init) {
+	return rtcSetAV1Packetizer(tr, init);
+}
+RTC_DEPRECATED static inline int
+rtcSetOpusPacketizationHandler(int tr, const rtcPacketizationHandlerInit *init) {
+	return rtcSetOpusPacketizer(tr, init);
+}
+RTC_DEPRECATED static inline int
+rtcSetAACPacketizationHandler(int tr, const rtcPacketizationHandlerInit *init) {
+	return rtcSetAACPacketizer(tr, init);
+}
+
+// Chain RtcpReceivingSession on track
+RTC_C_EXPORT int rtcChainRtcpReceivingSession(int tr);
+
+// Chain RtcpSrReporter on track
+RTC_C_EXPORT int rtcChainRtcpSrReporter(int tr);
+
+// Chain RtcpNackResponder on track
+RTC_C_EXPORT int rtcChainRtcpNackResponder(int tr, unsigned int maxStoredPacketsCount);
+
+// Chain PliHandler on track
+RTC_C_EXPORT int rtcChainPliHandler(int tr, rtcPliHandlerCallbackFunc cb);
+
+// Chain RembHandler on track
+RTC_C_EXPORT int rtcChainRembHandler(int tr, rtcRembHandlerCallbackFunc cb);
+
+// Chain PacingHandler on track
+RTC_C_EXPORT int rtcChainPacingHandler(int tr, double bitsPerSecond, int sendIntervalMs);
+
+// Transform seconds to timestamp using track's clock rate, result is written to timestamp
+RTC_C_EXPORT int rtcTransformSecondsToTimestamp(int id, double seconds, uint32_t *timestamp);
+
+// Transform timestamp to seconds using track's clock rate, result is written to seconds
+RTC_C_EXPORT int rtcTransformTimestampToSeconds(int id, uint32_t timestamp, double *seconds);
+
+// Get current timestamp, result is written to timestamp
+RTC_C_EXPORT int rtcGetCurrentTrackTimestamp(int id, uint32_t *timestamp);
+
+// Set RTP timestamp for track identified by given id
+RTC_C_EXPORT int rtcSetTrackRtpTimestamp(int id, uint32_t timestamp);
+
+// Get timestamp of last RTCP SR, result is written to timestamp
+RTC_C_EXPORT int rtcGetLastTrackSenderReportTimestamp(int id, uint32_t *timestamp);
+
+// Get sync timestamps from receiving RTCP session
+RTC_C_EXPORT int rtcGetTrackRtcpSyncTimestamps(int tr, uint64_t *rtpTimestamp, uint64_t *ntpTimestamp);
+
+// Get all available payload types for given codec and stores them in buffer, does nothing if
+// buffer is NULL
+RTC_C_EXPORT int rtcGetTrackPayloadTypesForCodec(int tr, const char *ccodec, int *buffer, int size);
+
+// Get all SSRCs for given track
+RTC_C_EXPORT int rtcGetSsrcsForTrack(int tr, uint32_t *buffer, int count);
+
+// Get CName for SSRC
+RTC_C_EXPORT int rtcGetCNameForSsrc(int tr, uint32_t ssrc, char *cname, int cnameSize);
+
+// Get all SSRCs for given media type in given SDP
+RTC_C_EXPORT int rtcGetSsrcsForType(const char *mediaType, const char *sdp, uint32_t *buffer, int bufferSize);
+
+// Set SSRC for given media type in given SDP
+RTC_C_EXPORT int rtcSetSsrcForType(const char *mediaType, const char *sdp, char *buffer, const int bufferSize,
+                                   rtcSsrcForTypeInit *init);
+
+// For backward compatibility, do not use
+RTC_C_EXPORT RTC_DEPRECATED int rtcSetNeedsToSendRtcpSr(int id);
+
+#endif // RTC_ENABLE_MEDIA
+
+#if RTC_ENABLE_WEBSOCKET
+
+// WebSocket
+
+typedef struct {
+	bool disableTlsVerification; // if true, don't verify the TLS certificate
+	const char *proxyServer;     // only non-authenticated http supported for now
+	const char **protocols;
+	int protocolsCount;
+	int tcpConnectionTimeoutMs; // in milliseconds, 0 means default, < 0 means disabled
+	int connectionTimeoutMs;    // in milliseconds, 0 means default, < 0 means disabled
+	int pingIntervalMs;         // in milliseconds, 0 means default, < 0 means disabled
+	int maxOutstandingPings;    // 0 means default, < 0 means disabled
+	int maxMessageSize;         // <= 0 means default
+} rtcWsConfiguration;
+
+RTC_C_EXPORT int rtcCreateWebSocket(const char *url); // returns ws id
+RTC_C_EXPORT int rtcCreateWebSocketEx(const char *url, const rtcWsConfiguration *config);
+RTC_C_EXPORT int rtcDeleteWebSocket(int ws);
+
+RTC_C_EXPORT int rtcGetWebSocketRemoteAddress(int ws, char *buffer, int size);
+RTC_C_EXPORT int rtcGetWebSocketPath(int ws, char *buffer, int size);
+
+// WebSocketServer
+
+typedef void(RTC_API *rtcWebSocketClientCallbackFunc)(int wsserver, int ws, void *ptr);
+
+typedef struct {
+	uint16_t port;                  // 0 means automatic selection
+	bool enableTls;                 // if true, enable TLS (WSS)
+	const char *certificatePemFile; // NULL for autogenerated certificate
+	const char *keyPemFile;         // NULL for autogenerated certificate
+	const char *keyPemPass;         // NULL if no pass
+	const char *bindAddress;        // NULL for any
+	int connectionTimeoutMs;        // in milliseconds, 0 means default, < 0 means disabled
+	int maxMessageSize;             // <= 0 means default
+} rtcWsServerConfiguration;
+
+RTC_C_EXPORT int rtcCreateWebSocketServer(const rtcWsServerConfiguration *config,
+                                          rtcWebSocketClientCallbackFunc cb); // returns wsserver id
+RTC_C_EXPORT int rtcDeleteWebSocketServer(int wsserver);
+
+RTC_C_EXPORT int rtcGetWebSocketServerPort(int wsserver);
+
+#endif
+
+// Global settings
+
+// Note: Applied when threads are spawned
+RTC_C_EXPORT int rtcSetThreadPoolSize(unsigned int count);
+
+typedef struct {
+	int recvBufferSize;          // in bytes, <= 0 means optimized default
+	int sendBufferSize;          // in bytes, <= 0 means optimized default
+	int maxChunksOnQueue;        // in chunks, <= 0 means optimized default
+	int initialCongestionWindow; // in MTUs, <= 0 means optimized default
+	int maxBurst;                // in MTUs, 0 means optimized default, < 0 means disabled
+	int congestionControlModule; // 0: RFC2581 (default), 1: HSTCP, 2: H-TCP, 3: RTCC
+	int delayedSackTimeMs;       // in milliseconds, 0 means optimized default, < 0 means disabled
+	int minRetransmitTimeoutMs;  // in milliseconds, <= 0 means optimized default
+	int maxRetransmitTimeoutMs;  // in milliseconds, <= 0 means optimized default
+	int initialRetransmitTimeoutMs; // in milliseconds, <= 0 means optimized default
+	int maxRetransmitAttempts;      // number of retransmissions, <= 0 means optimized default
+	int heartbeatIntervalMs;        // in milliseconds, <= 0 means optimized default
+} rtcSctpSettings;
+
+// Note: SCTP settings apply to newly-created PeerConnections only
+RTC_C_EXPORT int rtcSetSctpSettings(const rtcSctpSettings *settings);
+
+// Optional global preload and cleanup
+RTC_C_EXPORT bool rtcPreload(void);
+RTC_C_EXPORT void rtcCleanup(void);
+
+#ifdef __cplusplus
+} // extern "C"
+#endif
+
+#endif

libdatachannel/include/rtc/rtc.hpp

@@ -0,0 +1,53 @@
+/**
+ * Copyright (c) 2019 Paul-Louis Ageneau
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+// C API
+#include "rtc.h"
+
+// C++ API
+#include "common.hpp"
+#include "global.hpp"
+//
+#include "datachannel.hpp"
+#include "iceudpmuxlistener.hpp"
+#include "peerconnection.hpp"
+#include "track.hpp"
+
+#if RTC_ENABLE_WEBSOCKET
+
+// WebSocket
+#include "websocket.hpp"
+#include "websocketserver.hpp"
+
+#endif // RTC_ENABLE_WEBSOCKET
+
+#if RTC_ENABLE_MEDIA
+
+// Media
+#include "av1rtppacketizer.hpp"
+#include "dependencydescriptor.hpp"
+#include "rtppacketizer.hpp"
+#include "rtpdepacketizer.hpp"
+#include "h264rtppacketizer.hpp"
+#include "h264rtpdepacketizer.hpp"
+#include "h265rtppacketizer.hpp"
+#include "h265rtpdepacketizer.hpp"
+#include "vp8rtppacketizer.hpp"
+#include "vp8rtpdepacketizer.hpp"
+#include "vp9rtppacketizer.hpp"
+#include "vp9rtpdepacketizer.hpp"
+#include "mediahandler.hpp"
+#include "rtcpapphandler.hpp"
+#include "plihandler.hpp"
+#include "rembhandler.hpp"
+#include "pacinghandler.hpp"
+#include "rtcpnackresponder.hpp"
+#include "rtcpreceivingsession.hpp"
+#include "rtcpsrreporter.hpp"
+
+#endif // RTC_ENABLE_MEDIA

libdatachannel/include/rtc/rtcpapphandler.hpp

@@ -0,0 +1,36 @@
+/**
+ * Copyright (c) 2026 Apple Inc
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+#ifndef RTC_RTCP_APP_HANDLER_H
+#define RTC_RTCP_APP_HANDLER_H
+
+#if RTC_ENABLE_MEDIA
+
+#include "mediahandler.hpp"
+#include "rtp.hpp"
+#include "utils.hpp"
+
+namespace rtc {
+
+/// Handles RTCP APP packets (payload type 204) as defined in RFC 3550 section 6.7.
+class RTC_CPP_EXPORT RtcpAppHandler final : public MediaHandler {
+	rtc::synchronized_callback<const RtcpAppName, uint8_t, binary> mOnApp;
+
+public:
+	/// Constructs the RtcpAppHandler to notify when RTCP APP packets are received.
+	/// @param onApp Callback invoked with (name, subtype, application-dependent data)
+	RtcpAppHandler(std::function<void(const RtcpAppName name, uint8_t subtype, binary data)> onApp);
+
+	void incoming(message_vector &messages, const message_callback &send) override;
+};
+
+} // namespace rtc
+
+#endif // RTC_ENABLE_MEDIA
+
+#endif // RTC_RTCP_APP_HANDLER_H

libdatachannel/include/rtc/rtcpnackresponder.hpp

@@ -0,0 +1,89 @@
+/**
+ * Copyright (c) 2020 Filip Klembara (in2core)
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+#ifndef RTC_RTCP_NACK_RESPONDER_H
+#define RTC_RTCP_NACK_RESPONDER_H
+
+#if RTC_ENABLE_MEDIA
+
+#include "mediahandler.hpp"
+#include "description.hpp"
+#include "rtp.hpp"
+
+#include <mutex>
+#include <queue>
+#include <unordered_map>
+
+namespace rtc {
+
+class RTC_CPP_EXPORT RtcpNackResponder final : public MediaHandler {
+public:
+	static const size_t DefaultMaxSize = 512;
+
+	RtcpNackResponder(size_t maxSize = DefaultMaxSize);
+
+	void media(const Description::Media &desc) override;
+	void incoming(message_vector &messages, const message_callback &send) override;
+	void outgoing(message_vector &messages, const message_callback &send) override;
+
+private:
+	message_ptr wrapInRtx(const message_ptr &original);
+
+	// RTX state populated by media() from SDP inspection
+	optional<SSRC> mRtxSsrc;
+	std::unordered_map<int, int> mRtxPayloadTypeMap; // original PT -> RTX PT
+	uint16_t mRtxSequenceNumber = 0;
+	bool mRtxEnabled = false;
+	std::mutex mMutex;
+
+	// Packet storage
+	class RTC_CPP_EXPORT Storage {
+
+		/// Packet storage element
+		struct RTC_CPP_EXPORT Element {
+			Element(message_ptr packet, uint16_t sequenceNumber, shared_ptr<Element> next = nullptr);
+			const message_ptr packet;
+			const uint16_t sequenceNumber;
+			/// Pointer to newer element
+			shared_ptr<Element> next = nullptr;
+		};
+
+	private:
+		/// Oldest packet in storage
+		shared_ptr<Element> oldest = nullptr;
+		/// Newest packet in storage
+		shared_ptr<Element> newest = nullptr;
+		/// Inner storage
+		std::unordered_map<uint16_t, shared_ptr<Element>> storage{};
+		std::mutex mutex;
+
+		/// Maximum storage size
+		const size_t maxSize;
+
+		/// Returns current size
+		size_t size();
+
+	public:
+		Storage(size_t _maxSize);
+
+		/// Returns packet with given sequence number
+		message_ptr get(uint16_t sequenceNumber);
+
+		/// Stores packet
+		/// @param packet Packet
+		void store(message_ptr packet);
+	};
+
+	const shared_ptr<Storage> mStorage;
+};
+
+} // namespace rtc
+
+#endif /* RTC_ENABLE_MEDIA */
+
+#endif /* RTC_RTCP_NACK_RESPONDER_H */

libdatachannel/include/rtc/rtcpreceivingsession.hpp

@@ -0,0 +1,95 @@
+/**
+ * Copyright (c) 2020 Staz Modrzynski
+ * Copyright (c) 2020 Paul-Louis Ageneau
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+#ifndef RTC_RTCP_RECEIVING_SESSION_H
+#define RTC_RTCP_RECEIVING_SESSION_H
+
+#if RTC_ENABLE_MEDIA
+
+#include "common.hpp"
+#include "description.hpp"
+#include "mediahandler.hpp"
+#include "message.hpp"
+#include "rtp.hpp"
+
+#include <atomic>
+#include <mutex>
+#include <unordered_map>
+#include <vector>
+
+#define RTP_SEQ_MOD (1<<16)
+
+namespace rtc {
+
+// An RtcpSession can be plugged into a Track to handle the whole RTCP session
+class RTC_CPP_EXPORT RtcpReceivingSession : public MediaHandler {
+public:
+	RtcpReceivingSession() = default;
+	virtual ~RtcpReceivingSession() = default;
+
+	void media(const Description::Media &desc) override;
+	void incoming(message_vector &messages, const message_callback &send) override;
+	bool requestKeyframe(const std::vector<SSRC>& targetSSRCs, bool retransmit, const message_callback &send) override;
+	bool requestBitrate(unsigned int bitrate, const message_callback &send) override;
+
+	// For backward compatibility
+	[[deprecated("Use Track::requestKeyframe()")]] inline bool requestKeyframe() { return false; };
+	[[deprecated("Use Track::requestBitrate()")]] inline void requestBitrate(unsigned int) {};
+
+	struct SyncTimestamps {
+		uint64_t rtpTimestamp;
+		uint64_t ntpTimestamp;
+	};
+
+	SyncTimestamps getSyncTimestamps();
+
+protected:
+	void pushREMB(const message_callback &send, unsigned int bitrate);
+	void pushRR(const message_callback &send,unsigned int lastSrDelay);
+	void pushPLI(const message_callback &send);
+	void pushFIR(const message_callback &send, const std::vector<SSRC>& targetSSRCs, bool retransmit);
+
+	void initSeq(uint16_t seq);
+	bool updateSeq(uint16_t seq);
+	
+	SSRC mSsrc = 0;
+	uint32_t mGreatestSeqNo = 0;
+    uint16_t mMaxSeq = 0;			// highest seq. number seen
+    uint32_t mCycles = 0;			// shifted count of seq. number cycles
+	uint32_t mBaseSeq = 0;			// base seq number
+    uint32_t mBadSeq = 0;			// last 'bad' seq number + 1
+	uint32_t mProbation = 0;		// sequ. packets till source is valid
+    uint32_t mReceived = 0;			// packets received
+    uint32_t mExpectedPrior = 0;	// packet expected at last interval
+    uint32_t mReceivedPrior = 0;	// packet received at last interval
+	uint32_t mTransit = 0;			// relative trans time for prev pkt
+	uint32_t mJitter = 0;	
+	
+	SyncTimestamps mSyncTimestamps{0,0};
+
+	std::atomic<unsigned int> mRequestedBitrate = 0;
+	std::mutex mSyncMutex;
+	std::mutex mMutex;
+
+	message_ptr unwrapRtx(const message_ptr &rtxPacket);
+
+	// RTX state populated by media() from SDP inspection
+	std::unordered_map<uint8_t, uint8_t> mRtxToPrimaryPtMap; // RTX PT -> primary PT
+	SSRC mRtxPrimarySsrc = 0;
+	bool mRtxEnabled = false;
+
+	bool mSupportsRfc5104Fir = false;
+	std::unordered_map<SSRC, uint32_t> mRfc5104FirCmdNums; // SSRC -> seq no mapping
+};
+
+} // namespace rtc
+
+#endif // RTC_ENABLE_MEDIA
+
+#endif // RTC_RTCP_RECEIVING_SESSION_H

libdatachannel/include/rtc/rtcpsrreporter.hpp

@@ -0,0 +1,49 @@
+/**
+ * Copyright (c) 2020 Filip Klembara (in2core)
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+#ifndef RTC_RTCP_SR_REPORTER_H
+#define RTC_RTCP_SR_REPORTER_H
+
+#if RTC_ENABLE_MEDIA
+
+#include "mediahandler.hpp"
+#include "rtp.hpp"
+#include "rtppacketizationconfig.hpp"
+
+#include <chrono>
+
+namespace rtc {
+
+class RTC_CPP_EXPORT RtcpSrReporter final : public MediaHandler {
+public:
+	RtcpSrReporter(shared_ptr<RtpPacketizationConfig> rtpConfig);
+	~RtcpSrReporter();
+
+	uint32_t lastReportedTimestamp() const;
+	[[deprecated]] void setNeedsToReport();
+
+	void outgoing(message_vector &messages, const message_callback &send) override;
+
+	// TODO: remove this
+	const shared_ptr<RtpPacketizationConfig> rtpConfig;
+
+private:
+	void addToReport(RtpHeader *header, size_t size);
+	message_ptr getSenderReport(uint32_t timestamp);
+
+	uint32_t mPacketCount = 0;
+	uint32_t mPayloadOctets = 0;
+	uint32_t mLastReportedTimestamp = 0;
+	std::chrono::steady_clock::time_point mLastReportTime;
+};
+
+} // namespace rtc
+
+#endif /* RTC_ENABLE_MEDIA */
+
+#endif /* RTC_RTCP_SR_REPORTER_H */

libdatachannel/include/rtc/rtp.hpp

@@ -0,0 +1,407 @@
+/**
+ * Copyright (c) 2020 Staz Modrzynski
+ * Copyright (c) 2020 Paul-Louis Ageneau
+ * Copyright (c) 2020 Filip Klembara (in2core)
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+#ifndef RTC_RTP_HPP
+#define RTC_RTP_HPP
+
+#include "common.hpp"
+
+#include <array>
+#include <vector>
+
+namespace rtc {
+
+typedef uint32_t SSRC;
+
+RTC_CPP_EXPORT bool IsRtcp(const binary &data);
+
+#pragma pack(push, 1)
+
+struct RTC_CPP_EXPORT RtpExtensionHeader {
+	uint16_t _profileSpecificId;
+	uint16_t _headerLength;
+
+	[[nodiscard]] uint16_t profileSpecificId() const;
+	[[nodiscard]] uint16_t headerLength() const;
+
+	[[nodiscard]] size_t getSize() const;
+	[[nodiscard]] const char *getBody() const;
+	[[nodiscard]] char *getBody();
+
+	void setProfileSpecificId(uint16_t profileSpecificId);
+	void setHeaderLength(uint16_t headerLength);
+
+	void clearBody();
+	size_t writeCurrentVideoOrientation(bool twoByteHeader, size_t offset, uint8_t id,
+	                                    uint8_t value);
+	size_t writeOneByteHeader(size_t offset, uint8_t id, const byte *value, size_t size);
+	size_t writeTwoByteHeader(size_t offset, uint8_t id, const byte *value, size_t size);
+	size_t writeHeader(bool twoByteHeader, size_t offset, uint8_t id, const byte *value,
+	                   size_t size);
+};
+
+struct RTC_CPP_EXPORT RtpHeader {
+	uint8_t _first;
+	uint8_t _payloadType;
+	uint16_t _seqNumber;
+	uint32_t _timestamp;
+	SSRC _ssrc;
+	// The following field is SSRC _csrc[]
+
+	[[nodiscard]] uint8_t version() const;
+	[[nodiscard]] bool padding() const;
+	[[nodiscard]] bool extension() const;
+	[[nodiscard]] uint8_t csrcCount() const;
+	[[nodiscard]] uint8_t marker() const;
+	[[nodiscard]] uint8_t payloadType() const;
+	[[nodiscard]] uint16_t seqNumber() const;
+	[[nodiscard]] uint32_t timestamp() const;
+	[[nodiscard]] uint32_t ssrc() const;
+
+	[[nodiscard]] size_t getSize() const;
+	[[nodiscard]] size_t getExtensionHeaderSize() const;
+	[[nodiscard]] const RtpExtensionHeader *getExtensionHeader() const;
+	[[nodiscard]] RtpExtensionHeader *getExtensionHeader();
+	[[nodiscard]] const char *getBody() const;
+	[[nodiscard]] char *getBody();
+
+	void log() const;
+
+	void preparePacket();
+	void setSeqNumber(uint16_t newSeqNo);
+	void setPayloadType(uint8_t newPayloadType);
+	void setSsrc(uint32_t in_ssrc);
+	void setMarker(bool marker);
+	void setTimestamp(uint32_t i);
+	void setExtension(bool extension);
+};
+
+struct RTC_CPP_EXPORT RtcpReportBlock {
+	SSRC _ssrc;
+	uint32_t _fractionLostAndPacketsLost; // fraction lost is 8-bit, packets lost is 24-bit
+	uint16_t _seqNoCycles;
+	uint16_t _highestSeqNo;
+	uint32_t _jitter;
+	uint32_t _lastReport;
+	uint32_t _delaySinceLastReport;
+
+	[[nodiscard]] uint16_t seqNoCycles() const;
+	[[nodiscard]] uint16_t highestSeqNo() const;
+	[[nodiscard]] uint32_t extendedHighestSeqNo() const;
+	[[nodiscard]] uint32_t jitter() const;
+	[[nodiscard]] uint32_t delaySinceSR() const;
+
+	[[nodiscard]] SSRC getSSRC() const;
+	[[nodiscard]] uint32_t getNTPOfSR() const;
+	[[nodiscard]] uint8_t getFractionLost() const;
+	[[nodiscard]] unsigned int getPacketsLostCount() const;
+
+	void preparePacket(SSRC in_ssrc, uint8_t fraction, unsigned int totalPacketsLost,
+	                   uint16_t highestSeqNo, uint16_t seqNoCycles, uint32_t jitter,
+	                   uint64_t lastSR_NTP, uint64_t lastSR_DELAY);
+	void setSSRC(SSRC in_ssrc);
+	void setPacketsLost(uint8_t fractionLost, unsigned int packetsLostCount);
+	void setSeqNo(uint16_t highestSeqNo, uint16_t seqNoCycles);
+	void setJitter(uint32_t jitter);
+	void setNTPOfSR(uint64_t ntp);
+	void setDelaySinceSR(uint32_t sr);
+
+	void log() const;
+};
+
+struct RTC_CPP_EXPORT RtcpHeader {
+	uint8_t _first;
+	uint8_t _payloadType;
+	uint16_t _length;
+
+	[[nodiscard]] uint8_t version() const;
+	[[nodiscard]] bool padding() const;
+	[[nodiscard]] uint8_t reportCount() const;
+	[[nodiscard]] uint8_t payloadType() const;
+	[[nodiscard]] uint16_t length() const;
+	[[nodiscard]] size_t lengthInBytes() const;
+
+	void prepareHeader(uint8_t payloadType, uint8_t reportCount, uint16_t length);
+	void setPayloadType(uint8_t type);
+	void setReportCount(uint8_t count);
+	void setLength(uint16_t length);
+
+	void log() const;
+};
+
+struct RTC_CPP_EXPORT RtcpFbHeader {
+	RtcpHeader header;
+
+	SSRC _packetSender;
+	SSRC _mediaSource;
+
+	[[nodiscard]] SSRC packetSenderSSRC() const;
+	[[nodiscard]] SSRC mediaSourceSSRC() const;
+
+	void setPacketSenderSSRC(SSRC ssrc);
+	void setMediaSourceSSRC(SSRC ssrc);
+
+	void log() const;
+};
+
+struct RTC_CPP_EXPORT RtcpSr {
+	RtcpHeader header;
+
+	SSRC _senderSSRC;
+	uint64_t _ntpTimestamp;
+	uint32_t _rtpTimestamp;
+	uint32_t _packetCount;
+	uint32_t _octetCount;
+
+	RtcpReportBlock _reportBlocks;
+
+	[[nodiscard]] static unsigned int Size(unsigned int reportCount);
+
+	[[nodiscard]] uint64_t ntpTimestamp() const;
+	[[nodiscard]] uint32_t rtpTimestamp() const;
+	[[nodiscard]] uint32_t packetCount() const;
+	[[nodiscard]] uint32_t octetCount() const;
+	[[nodiscard]] uint32_t senderSSRC() const;
+
+	[[nodiscard]] const RtcpReportBlock *getReportBlock(int num) const; // nullptr if out-of-bounds
+	[[nodiscard]] RtcpReportBlock *getReportBlock(int num); // nullptr if out-of-bounds
+	[[nodiscard]] unsigned int size(unsigned int reportCount);
+	[[nodiscard]] size_t getSize() const;
+
+	void preparePacket(SSRC senderSSRC, uint8_t reportCount);
+	void setNtpTimestamp(uint64_t ts);
+	void setRtpTimestamp(uint32_t ts);
+	void setOctetCount(uint32_t ts);
+	void setPacketCount(uint32_t ts);
+
+	void log() const;
+};
+
+struct RTC_CPP_EXPORT RtcpSdesItem {
+	uint8_t type;
+
+	uint8_t _length;
+	char _text[1];
+
+	[[nodiscard]] static unsigned int Size(uint8_t textLength);
+
+	[[nodiscard]] string text() const;
+	[[nodiscard]] uint8_t length() const;
+
+	void setText(string text);
+};
+
+struct RTC_CPP_EXPORT RtcpSdesChunk {
+	SSRC _ssrc;
+	RtcpSdesItem _items;
+
+	[[nodiscard]] static unsigned int Size(const std::vector<uint8_t> textLengths);
+
+	[[nodiscard]] SSRC ssrc() const;
+
+	void setSSRC(SSRC ssrc);
+
+	// Get item at given index
+	// All items with index < num must be valid, otherwise this function has undefined behaviour
+	// (use safelyCountChunkSize() to check if chunk is valid).
+	[[nodiscard]] const RtcpSdesItem *getItem(int num) const;
+	[[nodiscard]] RtcpSdesItem *getItem(int num);
+
+	// Get size of chunk
+	// All items must be valid, otherwise this function has undefined behaviour (use
+	// safelyCountChunkSize() to check if chunk is valid)
+	[[nodiscard]] unsigned int getSize() const;
+
+	long safelyCountChunkSize(size_t maxChunkSize) const;
+};
+
+struct RTC_CPP_EXPORT RtcpSdes {
+	RtcpHeader header;
+	RtcpSdesChunk _chunks;
+
+	[[nodiscard]] static unsigned int Size(const std::vector<std::vector<uint8_t>> lengths);
+
+	bool isValid() const;
+
+	// Returns number of chunks in this packet
+	// Returns 0 if packet is invalid
+	unsigned int chunksCount() const;
+
+	// Get chunk at given index
+	// All chunks (and their items) with index < `num` must be valid, otherwise this function has
+	// undefined behaviour (use `isValid` to check if chunk is valid).
+	const RtcpSdesChunk *getChunk(int num) const;
+	RtcpSdesChunk *getChunk(int num);
+
+	void preparePacket(uint8_t chunkCount);
+};
+
+struct RTC_CPP_EXPORT RtcpRr {
+	RtcpHeader header;
+
+	SSRC _senderSSRC;
+	RtcpReportBlock _reportBlocks;
+
+	[[nodiscard]] static size_t SizeWithReportBlocks(uint8_t reportCount);
+
+	SSRC senderSSRC() const;
+	bool isSenderReport();
+	bool isReceiverReport();
+
+	[[nodiscard]] RtcpReportBlock *getReportBlock(int num); // nullptr if out-of-bounds
+	[[nodiscard]] const RtcpReportBlock *getReportBlock(int num) const; // nullptr if out-of-bounds
+	[[nodiscard]] size_t getSize() const;
+
+	void preparePacket(SSRC senderSSRC, uint8_t reportCount);
+	void setSenderSSRC(SSRC ssrc);
+
+	void log() const;
+};
+
+struct RTC_CPP_EXPORT RtcpRemb {
+	RtcpFbHeader header;
+
+	char _id[4];       // Unique identifier ('R' 'E' 'M' 'B')
+	uint32_t _bitrate; // Num SSRC, Br Exp, Br Mantissa (bit mask)
+	SSRC _ssrcs[1];
+
+	[[nodiscard]] static size_t SizeWithSSRCs(int count);
+
+	[[nodiscard]] unsigned int getSize() const;
+
+	void preparePacket(SSRC senderSSRC, unsigned int numSSRC, unsigned int in_bitrate);
+	void setBitrate(unsigned int numSSRC, unsigned int in_bitrate);
+	SSRC getSSRC(int num) const;
+	void setSSRC(int num, SSRC newSsrc);
+	bool hasValidId() const;
+	int getSSRCCount() const;
+	unsigned int getBitrate() const;
+
+	// Deprecated
+	[[deprecated("use setSSRC")]] void setSsrc(int num, SSRC newSssrc);
+	[[deprecated("use getSSRCCount")]] unsigned int getNumSSRC() const;
+	[[deprecated("use getSSRCCount")]] unsigned int getNumSSRC();
+	unsigned int getBitrate();
+};
+
+struct RTC_CPP_EXPORT RtcpPli {
+	RtcpFbHeader header;
+
+	[[nodiscard]] static unsigned int Size();
+
+	void preparePacket(SSRC messageSSRC);
+
+	void log() const;
+};
+
+struct RTC_CPP_EXPORT RtcpFirFci {
+	uint32_t _ssrc;
+	uint8_t _seqNo;
+	uint8_t _dummy1;
+	uint16_t _dummy2;
+
+	void preparePacket(SSRC ssrc, uint8_t seqNo);
+
+	[[nodiscard]] SSRC getSSRC() const;
+	[[nodiscard]] uint8_t getSeqNo() const;
+};
+
+struct RTC_CPP_EXPORT RtcpFir {
+	RtcpFbHeader header;
+	RtcpFirFci _fcis[1];
+
+	[[nodiscard]] static size_t SizeWithFcis(int count);
+
+	int getFciCount() const;
+	[[nodiscard]] const RtcpFirFci *getFci(int num) const; // nullptr if out-of-bounds
+	void preparePacket(SSRC senderSSRC, const std::vector<RtcpFirFci> &fcis);
+
+	void log() const;
+};
+
+struct RTC_CPP_EXPORT RtcpNackPart {
+	uint16_t _pid;
+	uint16_t _blp;
+
+	uint16_t pid();
+	uint16_t blp();
+
+	void setPid(uint16_t pid);
+	void setBlp(uint16_t blp);
+
+	std::vector<uint16_t> getSequenceNumbers();
+};
+
+struct RTC_CPP_EXPORT RtcpNack {
+	RtcpFbHeader header;
+	RtcpNackPart parts[1];
+
+	[[nodiscard]] static unsigned int Size(unsigned int discreteSeqNoCount);
+
+	[[nodiscard]] unsigned int getSeqNoCount();
+
+	void preparePacket(SSRC ssrc, unsigned int discreteSeqNoCount);
+
+	/**
+	 * Add a packet to the list of missing packets.
+	 * @param fciCount The number of FCI fields that are present in this packet.
+	 *                  Let the number start at zero and let this function grow the number.
+	 * @param fciPID The seq no of the active FCI. It will be initialized automatically, and will
+	 * change automatically.
+	 * @param missingPacket The seq no of the missing packet. This will be added to the queue.
+	 * @return true if the packet has grown, false otherwise.
+	 */
+	bool addMissingPacket(unsigned int *fciCount, uint16_t *fciPID, uint16_t missingPacket);
+};
+
+typedef std::array<char, 4> RtcpAppName;
+
+struct RTC_CPP_EXPORT RtcpApp {
+	RtcpHeader header;
+
+	SSRC _ssrc;
+	char _name[4];
+	char _data[1];
+
+	[[nodiscard]] static size_t SizeWithData(size_t dataLength);
+
+	[[nodiscard]] SSRC ssrc() const;
+	[[nodiscard]] uint8_t subtype() const;
+	[[nodiscard]] RtcpAppName name() const;
+	[[nodiscard]] const char *data() const;
+	[[nodiscard]] size_t dataSize() const;
+
+	void preparePacket(SSRC ssrc, const RtcpAppName &name, uint8_t subtype, size_t dataLength);
+	void setSSRC(SSRC ssrc);
+	void setName(const RtcpAppName &name);
+
+	void log() const;
+};
+
+struct RTC_CPP_EXPORT RtpRtx {
+	RtpHeader header;
+
+	[[nodiscard]] const char *getBody() const;
+	[[nodiscard]] char *getBody();
+	[[nodiscard]] size_t getBodySize(size_t totalSize) const;
+	[[nodiscard]] size_t getSize() const;
+	[[nodiscard]] uint16_t getOriginalSeqNo() const;
+
+	// Returns the new size of the packet
+	size_t normalizePacket(size_t totalSize, SSRC originalSSRC, uint8_t originalPayloadType);
+
+	size_t copyTo(RtpHeader *dest, size_t totalSize, uint8_t originalPayloadType);
+};
+
+#pragma pack(pop)
+
+} // namespace rtc
+
+#endif

libdatachannel/include/rtc/rtpdepacketizer.hpp

@@ -0,0 +1,79 @@
+/**
+ * Copyright (c) 2024 Paul-Louis Ageneau
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+#ifndef RTC_RTP_DEPACKETIZER_H
+#define RTC_RTP_DEPACKETIZER_H
+
+#if RTC_ENABLE_MEDIA
+
+#include "mediahandler.hpp"
+#include "message.hpp"
+
+#include <set>
+
+namespace rtc {
+
+// Base RTP depacketizer class
+class RTC_CPP_EXPORT RtpDepacketizer : public MediaHandler {
+public:
+	RtpDepacketizer();
+	RtpDepacketizer(uint32_t clockRate);
+	virtual ~RtpDepacketizer();
+
+	virtual void incoming(message_vector &messages, const message_callback &send) override;
+
+protected:
+	shared_ptr<FrameInfo> createFrameInfo(uint32_t timestamp, uint8_t payloadType) const;
+
+private:
+	const uint32_t mClockRate;
+};
+
+// Base class for video RTP depacketizer
+class RTC_CPP_EXPORT VideoRtpDepacketizer : public RtpDepacketizer {
+public:
+	inline static const uint32_t ClockRate = 90000;
+
+	VideoRtpDepacketizer();
+	virtual ~VideoRtpDepacketizer();
+
+protected:
+	struct sequence_cmp {
+		bool operator()(message_ptr a, message_ptr b) const;
+	};
+	using message_buffer = std::set<message_ptr, sequence_cmp>;
+
+	virtual message_ptr reassemble(message_buffer &messages) = 0;
+
+private:
+	void incoming(message_vector &messages, const message_callback &send) override;
+
+	message_buffer mBuffer;
+};
+
+// Generic audio RTP depacketizer
+template <uint32_t DEFAULT_CLOCK_RATE>
+class RTC_CPP_EXPORT AudioRtpDepacketizer final : public RtpDepacketizer {
+public:
+	inline static const uint32_t DefaultClockRate = DEFAULT_CLOCK_RATE;
+
+	AudioRtpDepacketizer(uint32_t clockRate = DefaultClockRate) : RtpDepacketizer(clockRate) {}
+};
+
+// Audio RTP depacketizers
+using OpusRtpDepacketizer = AudioRtpDepacketizer<48000>;
+using AACRtpDepacketizer = AudioRtpDepacketizer<48000>;
+using PCMARtpDepacketizer = AudioRtpDepacketizer<8000>;
+using PCMURtpDepacketizer = AudioRtpDepacketizer<8000>;
+using G722RtpDepacketizer = AudioRtpDepacketizer<8000>;
+
+} // namespace rtc
+
+#endif /* RTC_ENABLE_MEDIA */
+
+#endif /* RTC_RTP_DEPACKETIZER_H */

libdatachannel/include/rtc/rtppacketizationconfig.hpp

@@ -0,0 +1,141 @@
+/**
+ * Copyright (c) 2020 Filip Klembara (in2core)
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+#ifndef RTC_RTP_PACKETIZATION_CONFIG_H
+#define RTC_RTP_PACKETIZATION_CONFIG_H
+
+#if RTC_ENABLE_MEDIA
+
+#include "dependencydescriptor.hpp"
+#include "rtp.hpp"
+
+namespace rtc {
+
+struct RTC_CPP_EXPORT VideoLayersAllocation;
+
+// RTP configuration used in packetization process
+class RTC_CPP_EXPORT RtpPacketizationConfig {
+public:
+	SSRC ssrc;
+	std::string cname;
+	uint8_t payloadType;
+	uint32_t clockRate;
+	uint8_t videoOrientationId;
+
+	// current sequence number
+	uint16_t sequenceNumber;
+
+	// current timestamp
+	uint32_t timestamp;
+
+	// start timestamp
+	uint32_t startTimestamp;
+
+	/// Current video orientation
+	///
+	/// Bit#       7  6  5  4  3  2  1  0
+	/// Definition 0  0  0  0  C  F  R1 R0
+	///
+	/// C
+	///   0 - Front-facing camera (use this if unsure)
+	///   1 - Back-facing camera
+	///
+	/// F
+	///   0 - No Flip
+	///   1 - Horizontal flip
+	///
+	/// R1 R0 - CW rotation that receiver must apply
+	///   0 - 0 degrees
+	///   1 - 90 degrees
+	///   2 - 180 degrees
+	///   3 - 270 degrees
+	uint8_t videoOrientation = 0;
+
+	// MID Extension Header
+	uint8_t midId = 0;
+	optional<std::string> mid;
+
+	// RID Extension Header
+	uint8_t ridId = 0;
+	optional<std::string> rid;
+
+	// Dependency Descriptor Extension Header
+	uint8_t dependencyDescriptorId = 0;
+
+	optional<DependencyDescriptorContext> dependencyDescriptorContext;
+	// the negotiated ID of the playout delay header extension
+	// https://webrtc.googlesource.com/src/+/main/docs/native-code/rtp-hdrext/playout-delay/README.md
+	uint8_t playoutDelayId = 0;
+
+	// Minimum/maxiumum playout delay, in 10ms intervals. A value of 10 would equal a 100ms delay
+	uint16_t playoutDelayMin = 0;
+	uint16_t playoutDelayMax = 0;
+
+	// Google Video Layers Allocation for simulcast
+	// https://webrtc.googlesource.com/src/+/refs/heads/main/docs/native-code/rtp-hdrext/video-layers-allocation00
+	//
+	// The negotiated extension id
+	uint8_t videoLayersAllocationId = 0;
+	// Stream index, unique per RID/SSRC
+	uint8_t videoLayersAllocationStreamIndex = 0;
+	// Shared data about layers
+	std::shared_ptr<const VideoLayersAllocation> videoLayersAllocationStreams;
+
+	// https://webrtc.googlesource.com/src/+/refs/heads/main/docs/native-code/rtp-hdrext/color-space/
+	uint8_t colorSpaceId = 0;               // the negotiated ID of color space header extension
+	uint8_t colorChromaSitingHorz = 0;      // unspecified
+	uint8_t colorChromaSitingVert = 0;      // unspecified
+	uint8_t colorRange = 2;                 // full range
+	uint8_t colorPrimaries = 1;             // BT.709-6
+	uint8_t colorTransfer = 1;              // BT.709-6
+	uint8_t colorMatrix = 1;                // BT.709-6
+
+	// abs-capture-time RTP header extension. When absCaptureTimeId > 0 and an
+	// outgoing FrameInfo carries an absCaptureTimeNtp value, the RtpPacketizer
+	// writes the 8-byte (shortened) form of the abs-capture-time extension into
+	// every packet for that frame. The receiver can recover the source's
+	// capture wallclock and compute true glass-to-glass latency.
+	// https://webrtc.googlesource.com/src/+/refs/heads/main/docs/native-code/rtp-hdrext/abs-capture-time
+	uint8_t absCaptureTimeId = 0;
+
+	/// Construct RTP configuration used in packetization process
+	/// @param ssrc SSRC of source
+	/// @param cname CNAME of source
+	/// @param payloadType Payload type of source
+	/// @param clockRate Clock rate of source used in timestamps
+	/// nullopt)
+	/// @param videoOrientationId Video orientation (see above)
+	RtpPacketizationConfig(SSRC ssrc, std::string cname, uint8_t payloadType, uint32_t clockRate,
+	                       uint8_t videoOrientationId = 0);
+
+	RtpPacketizationConfig(const RtpPacketizationConfig &) = delete;
+
+	/// Convert timestamp to seconds
+	/// @param timestamp Timestamp
+	/// @param clockRate Clock rate for timestamp calculation
+	static double getSecondsFromTimestamp(uint32_t timestamp, uint32_t clockRate);
+
+	/// Convert timestamp to seconds
+	/// @param timestamp Timestamp
+	double timestampToSeconds(uint32_t timestamp);
+
+	/// Convert seconds to timestamp
+	/// @param seconds Number of seconds
+	/// @param clockRate Clock rate for timestamp calculation
+	static uint32_t getTimestampFromSeconds(double seconds, uint32_t clockRate);
+
+	/// Convert seconds to timestamp
+	/// @param seconds Number of seconds
+	uint32_t secondsToTimestamp(double seconds);
+};
+
+} // namespace rtc
+
+#endif /* RTC_ENABLE_MEDIA */
+
+#endif /* RTC_RTP_PACKETIZATION_CONFIG_H */

libdatachannel/include/rtc/rtppacketizer.hpp

@@ -0,0 +1,106 @@
+/**
+ * Copyright (c) 2020 Filip Klembara (in2core)
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+#ifndef RTC_RTP_PACKETIZER_H
+#define RTC_RTP_PACKETIZER_H
+
+#if RTC_ENABLE_MEDIA
+
+#include "mediahandler.hpp"
+#include "message.hpp"
+#include "rtppacketizationconfig.hpp"
+
+namespace rtc {
+
+/// RTP packetizer
+class RTC_CPP_EXPORT RtpPacketizer : public MediaHandler {
+public:
+	/// Default maximum fragment size (for video packetizers)
+	inline static const size_t DefaultMaxFragmentSize = RTC_DEFAULT_MAX_FRAGMENT_SIZE;
+
+	/// Clock rate for video in RTP
+	inline static const uint32_t VideoClockRate = 90 * 1000;
+
+	/// Constructs packetizer with given RTP configuration
+	/// @note RTP configuration is used in packetization process which may change some configuration
+	/// properties such as sequence number.
+	/// @param rtpConfig  RTP configuration
+	RtpPacketizer(shared_ptr<RtpPacketizationConfig> rtpConfig);
+	virtual ~RtpPacketizer();
+
+	virtual void media(const Description::Media &desc) override;
+	virtual void outgoing(message_vector &messages, const message_callback &send) override;
+
+	/// RTP packetization config
+	const shared_ptr<RtpPacketizationConfig> rtpConfig;
+
+protected:
+	/// Fragment data into payloads
+	/// Default implementation returns data as a single payload
+	/// @param data Input data
+	virtual std::vector<binary> fragment(binary data);
+
+	/// Creates an RTP packet for a payload
+	/// @note This function increases the sequence number.
+	/// @param payload RTP payload
+	/// @param mark Set marker flag in RTP packet if true
+	virtual message_ptr packetize(const binary &payload, bool mark, shared_ptr<FrameInfo> frameInfo = nullptr);
+
+private:
+	static const auto RtpHeaderSize = 12;
+	static const auto RtpExtHeaderCvoSize = 8;
+
+	uint32_t videoLayersAllocationInitialPacketCount = 0;
+
+	bool shouldEmitVideoLayersAllocation(shared_ptr<FrameInfo> frameInfo);
+};
+
+// Generic audio RTP packetizer
+template <uint32_t DEFAULT_CLOCK_RATE>
+class RTC_CPP_EXPORT AudioRtpPacketizer final : public RtpPacketizer {
+public:
+	inline static const uint32_t DefaultClockRate = DEFAULT_CLOCK_RATE;
+	inline static const uint32_t defaultClockRate [[deprecated("Use DefaultClockRate")]] =
+	    DEFAULT_CLOCK_RATE; // for backward compatibility
+
+	AudioRtpPacketizer(shared_ptr<RtpPacketizationConfig> rtpConfig)
+	    : RtpPacketizer(std::move(rtpConfig)) {}
+};
+
+// Audio RTP packetizers
+using OpusRtpPacketizer = AudioRtpPacketizer<48000>;
+using AACRtpPacketizer = AudioRtpPacketizer<48000>;
+using PCMARtpPacketizer = AudioRtpPacketizer<8000>;
+using PCMURtpPacketizer = AudioRtpPacketizer<8000>;
+using G722RtpPacketizer = AudioRtpPacketizer<8000>;
+
+// Dummy wrapper for backward compatibility, do not use
+class RTC_CPP_EXPORT PacketizationHandler final : public MediaHandler {
+public:
+	PacketizationHandler(shared_ptr<RtpPacketizer> packetizer)
+	    : mPacketizer(std::move(packetizer)) {}
+
+	inline void outgoing(message_vector &messages, const message_callback &send) {
+		return mPacketizer->outgoing(messages, send);
+	}
+
+private:
+	shared_ptr<RtpPacketizer> mPacketizer;
+};
+
+// Audio packetization handlers for backward compatibility, do not use
+using OpusPacketizationHandler [[deprecated("Add OpusRtpPacketizer directly")]] =
+    PacketizationHandler;
+using AACPacketizationHandler [[deprecated("Add AACRtpPacketizer directly")]] =
+    PacketizationHandler;
+
+} // namespace rtc
+
+#endif /* RTC_ENABLE_MEDIA */
+
+#endif /* RTC_RTP_PACKETIZER_H */

libdatachannel/include/rtc/track.hpp

@@ -0,0 +1,70 @@
+/**
+ * Copyright (c) 2020 Paul-Louis Ageneau
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+#ifndef RTC_TRACK_H
+#define RTC_TRACK_H
+
+#include "channel.hpp"
+#include "common.hpp"
+#include "description.hpp"
+#include "mediahandler.hpp"
+
+#include <vector>
+
+namespace rtc {
+
+namespace impl {
+
+class Track;
+
+} // namespace impl
+
+class RTC_CPP_EXPORT Track final : private CheshireCat<impl::Track>, public Channel {
+public:
+	Track(impl_ptr<impl::Track> impl);
+	~Track() override;
+
+	string mid() const;
+	Description::Direction direction() const;
+	Description::Media description() const;
+
+	void setDescription(Description::Media description);
+
+	void close(void) override;
+	bool send(message_variant data) override;
+	bool send(const byte *data, size_t size) override;
+
+	bool isOpen(void) const override;
+	bool isClosed(void) const override;
+	size_t maxMessageSize() const override;
+
+	void sendFrame(binary data, FrameInfo info);
+	void sendFrame(const byte *data, size_t size, FrameInfo info);
+	void onFrame(std::function<void(binary data, FrameInfo info)> callback);
+
+	bool requestKeyframe(SSRC ssrc=0, bool retransmit=false);
+	bool requestKeyframe(const std::vector<SSRC>& targetSSRCs, bool retransmit=false);
+	bool requestBitrate(unsigned int bitrate);
+	bool sendRtcpApp(uint32_t ssrc, const RtcpAppName &name, uint8_t subtype,
+	                 const binary &data = binary{});
+
+	void setMediaHandler(shared_ptr<MediaHandler> handler);
+	void chainMediaHandler(shared_ptr<MediaHandler> handler);
+	shared_ptr<MediaHandler> getMediaHandler();
+
+	// Deprecated, use setMediaHandler() and getMediaHandler()
+	inline void setRtcpHandler(shared_ptr<MediaHandler> handler) { setMediaHandler(handler); }
+	inline shared_ptr<MediaHandler> getRtcpHandler() { return getMediaHandler(); }
+
+private:
+	using CheshireCat<impl::Track>::impl;
+};
+
+} // namespace rtc
+
+#endif

libdatachannel/include/rtc/utils.hpp

@@ -0,0 +1,174 @@
+/**
+ * Copyright (c) 2019-2021 Paul-Louis Ageneau
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+#ifndef RTC_UTILS_H
+#define RTC_UTILS_H
+
+#include <algorithm>
+#include <cctype>
+#include <functional>
+#include <memory>
+#include <mutex>
+#include <optional>
+#include <string>
+#include <tuple>
+#include <utility>
+
+namespace rtc {
+
+// overloaded helper
+template <class... Ts> struct overloaded : Ts... { using Ts::operator()...; };
+template <class... Ts> overloaded(Ts...) -> overloaded<Ts...>;
+
+// weak_ptr bind helper
+template <typename F, typename T, typename... Args> auto weak_bind(F &&f, T *t, Args &&..._args) {
+	return [bound = std::bind(f, t, _args...), weak_this = t->weak_from_this()](auto &&...args) {
+		if (auto shared_this = weak_this.lock())
+			return bound(args...);
+		else
+			return static_cast<decltype(bound(args...))>(false);
+	};
+}
+
+// scope_guard helper
+class scope_guard final {
+public:
+	scope_guard(std::function<void()> func) : function(std::move(func)) {}
+	scope_guard(scope_guard &&other) = delete;
+	scope_guard(const scope_guard &) = delete;
+	void operator=(const scope_guard &) = delete;
+
+	~scope_guard() {
+		if (function)
+			function();
+	}
+
+private:
+	std::function<void()> function;
+};
+
+// callback with built-in synchronization
+template <typename... Args> class synchronized_callback {
+public:
+	synchronized_callback() = default;
+	synchronized_callback(synchronized_callback &&cb) { *this = std::move(cb); }
+	synchronized_callback(const synchronized_callback &cb) { *this = cb; }
+	synchronized_callback(std::function<void(Args...)> func) { *this = std::move(func); }
+	virtual ~synchronized_callback() { *this = nullptr; }
+
+	synchronized_callback &operator=(synchronized_callback &&cb) {
+		std::scoped_lock lock(mutex, cb.mutex);
+		set(std::exchange(cb.callback, nullptr));
+		return *this;
+	}
+
+	synchronized_callback &operator=(const synchronized_callback &cb) {
+		std::scoped_lock lock(mutex, cb.mutex);
+		set(cb.callback);
+		return *this;
+	}
+
+	synchronized_callback &operator=(std::function<void(Args...)> func) {
+		std::lock_guard lock(mutex);
+		set(std::move(func));
+		return *this;
+	}
+
+	bool operator()(Args... args) const {
+		std::lock_guard lock(mutex);
+		return call(std::move(args)...);
+	}
+
+	operator bool() const {
+		std::lock_guard lock(mutex);
+		return callback ? true : false;
+	}
+
+protected:
+	virtual void set(std::function<void(Args...)> func) { callback = std::move(func); }
+	virtual bool call(Args... args) const {
+		if (!callback)
+			return false;
+
+		callback(std::move(args)...);
+		return true;
+	}
+
+	std::function<void(Args...)> callback;
+	mutable std::recursive_mutex mutex;
+};
+
+// callback with built-in synchronization and replay of the last missed call
+template <typename... Args>
+class synchronized_stored_callback final : public synchronized_callback<Args...> {
+public:
+	template <typename... CArgs>
+	synchronized_stored_callback(CArgs &&...cargs)
+	    : synchronized_callback<Args...>(std::forward<CArgs>(cargs)...) {}
+	~synchronized_stored_callback() {}
+
+private:
+	void set(std::function<void(Args...)> func) {
+		synchronized_callback<Args...>::set(func);
+		if (func && stored) {
+			std::apply(func, std::move(*stored));
+			stored.reset();
+		}
+	}
+
+	bool call(Args... args) const {
+		if (!synchronized_callback<Args...>::call(args...))
+			stored.emplace(std::move(args)...);
+
+		return true;
+	}
+
+	mutable std::optional<std::tuple<Args...>> stored;
+};
+
+// pimpl base class
+template <typename T> using impl_ptr = std::shared_ptr<T>;
+template <typename T> class CheshireCat {
+public:
+	CheshireCat(impl_ptr<T> impl) : mImpl(std::move(impl)) {}
+	template <typename... Args>
+	CheshireCat(Args... args) : mImpl(std::make_shared<T>(std::forward<Args>(args)...)) {}
+	CheshireCat(CheshireCat<T> &&cc) { *this = std::move(cc); }
+	CheshireCat(const CheshireCat<T> &) = delete;
+
+	virtual ~CheshireCat() = default;
+
+	CheshireCat &operator=(CheshireCat<T> &&cc) {
+		mImpl = std::move(cc.mImpl);
+		return *this;
+	};
+	CheshireCat &operator=(const CheshireCat<T> &) = delete;
+
+protected:
+	impl_ptr<T> impl() { return mImpl; }
+	impl_ptr<const T> impl() const { return mImpl; }
+
+private:
+	impl_ptr<T> mImpl;
+};
+
+// case-insensitive string comparator using std::tolower
+class case_insensitive_less : public std::less<std::string> {
+public:
+	bool operator()(const std::string &lhs, const std::string &rhs) const {
+		return std::lexicographical_compare(lhs.begin(), lhs.end(), rhs.begin(), rhs.end(),
+		                                    [](auto a, auto b) {
+			                                    return std::tolower(static_cast<unsigned char>(a)) <
+			                                           std::tolower(static_cast<unsigned char>(b));
+		                                    });
+	}
+};
+
+} // namespace rtc
+
+#endif

libdatachannel/include/rtc/version.h

@@ -0,0 +1,9 @@
+#ifndef RTC_VERSION_H
+#define RTC_VERSION_H
+
+#define RTC_VERSION_MAJOR 0
+#define RTC_VERSION_MINOR 24
+#define RTC_VERSION_PATCH 4
+#define RTC_VERSION "0.24.4"
+
+#endif

libdatachannel/include/rtc/video_layers_allocation.hpp

@@ -0,0 +1,50 @@
+/**
+ * Copyright (c) 2026
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+#ifndef RTC_VIDEO_LAYERS_ALLOCATION_H
+#define RTC_VIDEO_LAYERS_ALLOCATION_H
+
+#if RTC_ENABLE_MEDIA
+
+#include "common.hpp"
+
+#include <cstdint>
+#include <memory>
+#include <vector>
+
+namespace rtc {
+
+// Google Video Layers Allocation for simulcast
+//
+// https://webrtc.googlesource.com/src/+/refs/heads/main/docs/native-code/rtp-hdrext/video-layers-allocation00
+
+struct RTC_CPP_EXPORT VideoLayersAllocation {
+	struct SpatialLayer {
+		uint16_t width = 0;
+		uint16_t height = 0;
+		uint8_t fps = 0;
+		std::vector<uint32_t> targetBitratesKbps;  // per temporal layer, cumulative, cannot be empty
+	};
+
+	struct RtpStream {
+		std::vector<SpatialLayer> spatialLayers;
+	};
+
+	std::vector<RtpStream> rtpStreams;  // up to 4 streams
+
+	/// Generate the wire format for Google Video Layers Allocation RTP header extension
+	/// @param streamIndex The RTP stream index (0-3) for this packet's stream
+	/// @return Binary payload for the RTP header extension, empty if allocation is invalid
+	binary generate(uint8_t streamIndex) const;
+};
+
+} // namespace rtc
+
+#endif /* RTC_ENABLE_MEDIA */
+
+#endif /* RTC_VIDEO_LAYERS_ALLOCATION_H */

libdatachannel/include/rtc/vp8rtpdepacketizer.hpp

@@ -0,0 +1,34 @@
+/**
+ * Copyright (c) 2026 Paul-Louis Ageneau
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+#ifndef RTC_VP8_RTP_DEPACKETIZER_H
+#define RTC_VP8_RTP_DEPACKETIZER_H
+
+#if RTC_ENABLE_MEDIA
+
+#include "common.hpp"
+#include "message.hpp"
+#include "rtpdepacketizer.hpp"
+
+namespace rtc {
+
+/// RTP depacketization for VP8
+class RTC_CPP_EXPORT VP8RtpDepacketizer final : public VideoRtpDepacketizer {
+public:
+	VP8RtpDepacketizer();
+	~VP8RtpDepacketizer();
+
+private:
+	message_ptr reassemble(message_buffer &buffer) override;
+};
+
+} // namespace rtc
+
+#endif // RTC_ENABLE_MEDIA
+
+#endif /* RTC_VP8_RTP_DEPACKETIZER_H */

libdatachannel/include/rtc/vp8rtppacketizer.hpp

@@ -0,0 +1,42 @@
+/**
+ * Copyright (c) 2026 Paul-Louis Ageneau
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+#ifndef RTC_VP8_RTP_PACKETIZER_H
+#define RTC_VP8_RTP_PACKETIZER_H
+
+#if RTC_ENABLE_MEDIA
+
+#include "rtppacketizer.hpp"
+
+namespace rtc {
+
+/// RTP packetization for VP8
+class RTC_CPP_EXPORT VP8RtpPacketizer final : public RtpPacketizer {
+public:
+	inline static const uint32_t ClockRate = VideoClockRate;
+	[[deprecated("Use ClockRate")]] inline static const uint32_t defaultClockRate = ClockRate;
+
+	/// Constructs VP8 payload packetizer with given RTP configuration.
+	/// @note RTP configuration is used in packetization process which may change some configuration
+	/// properties such as sequence number.
+	/// @param rtpConfig RTP configuration
+	/// @param maxFragmentSize maximum size of one packet payload
+	VP8RtpPacketizer(shared_ptr<RtpPacketizationConfig> rtpConfig,
+	                 size_t maxFragmentSize = DefaultMaxFragmentSize);
+
+private:
+	std::vector<binary> fragment(binary frame) override;
+
+	const size_t mMaxFragmentSize;
+};
+
+} // namespace rtc
+
+#endif /* RTC_ENABLE_MEDIA */
+
+#endif /* RTC_VP8_RTP_PACKETIZER_H */

libdatachannel/include/rtc/vp9rtpdepacketizer.hpp

@@ -0,0 +1,34 @@
+/**
+ * Copyright (c) 2026 Paul-Louis Ageneau
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+#ifndef RTC_VP9_RTP_DEPACKETIZER_H
+#define RTC_VP9_RTP_DEPACKETIZER_H
+
+#if RTC_ENABLE_MEDIA
+
+#include "common.hpp"
+#include "message.hpp"
+#include "rtpdepacketizer.hpp"
+
+namespace rtc {
+
+/// RTP depacketization for VP9
+class RTC_CPP_EXPORT VP9RtpDepacketizer final : public VideoRtpDepacketizer {
+public:
+	VP9RtpDepacketizer();
+	~VP9RtpDepacketizer();
+
+private:
+	message_ptr reassemble(message_buffer &buffer) override;
+};
+
+} // namespace rtc
+
+#endif // RTC_ENABLE_MEDIA
+
+#endif /* RTC_VP9_RTP_DEPACKETIZER_H */

libdatachannel/include/rtc/vp9rtppacketizer.hpp

@@ -0,0 +1,41 @@
+/**
+ * Copyright (c) 2026 Paul-Louis Ageneau
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+#ifndef RTC_VP9_RTP_PACKETIZER_H
+#define RTC_VP9_RTP_PACKETIZER_H
+
+#if RTC_ENABLE_MEDIA
+
+#include "rtppacketizer.hpp"
+
+namespace rtc {
+
+/// RTP packetization for VP9
+class RTC_CPP_EXPORT VP9RtpPacketizer final : public RtpPacketizer {
+public:
+	inline static const uint32_t ClockRate = VideoClockRate;
+
+	/// Constructs VP9 payload packetizer with given RTP configuration.
+	/// @note RTP configuration is used in packetization process which may change some configuration
+	/// properties such as sequence number.
+	/// @param rtpConfig RTP configuration
+	/// @param maxFragmentSize maximum size of one packet payload
+	VP9RtpPacketizer(shared_ptr<RtpPacketizationConfig> rtpConfig,
+	                 size_t maxFragmentSize = DefaultMaxFragmentSize);
+
+private:
+	std::vector<binary> fragment(binary frame) override;
+
+	const size_t mMaxFragmentSize;
+};
+
+} // namespace rtc
+
+#endif /* RTC_ENABLE_MEDIA */
+
+#endif /* RTC_VP9_RTP_PACKETIZER_H */

libdatachannel/include/rtc/websocket.hpp

@@ -0,0 +1,72 @@
+/**
+ * Copyright (c) 2020-2021 Paul-Louis Ageneau
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+#ifndef RTC_WEBSOCKET_H
+#define RTC_WEBSOCKET_H
+
+#if RTC_ENABLE_WEBSOCKET
+
+#include "channel.hpp"
+#include "common.hpp"
+#include "configuration.hpp"
+
+#include <map>
+
+namespace rtc {
+
+namespace impl {
+
+struct WebSocket;
+
+} // namespace impl
+
+class RTC_CPP_EXPORT WebSocket final : private CheshireCat<impl::WebSocket>, public Channel {
+public:
+	enum class State : int {
+		Connecting = 0,
+		Open = 1,
+		Closing = 2,
+		Closed = 3,
+	};
+
+	using Configuration = WebSocketConfiguration;
+
+	using Headers = std::map<string, string, case_insensitive_less>;
+
+	WebSocket();
+	WebSocket(Configuration config);
+	WebSocket(impl_ptr<impl::WebSocket> impl);
+	~WebSocket() override;
+
+	State readyState() const;
+
+	bool isOpen() const override;
+	bool isClosed() const override;
+	size_t maxMessageSize() const override;
+
+	void open(const string &url, const Headers &headers = {});
+	void close() override;
+	void forceClose();
+	bool send(const message_variant data) override;
+	bool send(const byte *data, size_t size) override;
+
+	optional<string> remoteAddress() const;
+	optional<string> path() const;
+	std::multimap<string, string, case_insensitive_less> requestHeaders() const;
+
+private:
+	using CheshireCat<impl::WebSocket>::impl;
+};
+
+std::ostream &operator<<(std::ostream &out, WebSocket::State state);
+
+} // namespace rtc
+
+#endif
+
+#endif // RTC_WEBSOCKET_H

libdatachannel/include/rtc/websocketserver.hpp

@@ -0,0 +1,48 @@
+/**
+ * Copyright (c) 2021 Paul-Louis Ageneau
+ *
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/.
+ */
+
+#ifndef RTC_WEBSOCKETSERVER_H
+#define RTC_WEBSOCKETSERVER_H
+
+#if RTC_ENABLE_WEBSOCKET
+
+#include "common.hpp"
+#include "configuration.hpp"
+#include "websocket.hpp"
+
+namespace rtc {
+
+namespace impl {
+
+struct WebSocketServer;
+
+}
+
+class RTC_CPP_EXPORT WebSocketServer final : private CheshireCat<impl::WebSocketServer> {
+public:
+	using Configuration = WebSocketServerConfiguration;
+
+	WebSocketServer();
+	WebSocketServer(Configuration config);
+	~WebSocketServer();
+
+	void stop();
+
+	uint16_t port() const;
+
+	void onClient(std::function<void(shared_ptr<WebSocket>)> callback);
+
+private:
+	using CheshireCat<impl::WebSocketServer>::impl;
+};
+
+} // namespace rtc
+
+#endif
+
+#endif // RTC_WEBSOCKET_H