Updated RFC references

README.md

@@ -16,21 +16,22 @@ The library implements the following communication protocols:
 The WebRTC stack has been tested to be compatible with Firefox and Chromium.
 
 Protocol stack:
-- SCTP-based Data Channels ([draft-ietf-rtcweb-data-channel-13](https://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-13))
-- SRTP-based Media Transport ([draft-ietf-rtcweb-rtp-usage-26](https://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-26))
+- SCTP-based Data Channels ([RFC8831](https://tools.ietf.org/html/rfc8831))
+- SRTP-based Media Transport ([RFC8834](https://tools.ietf.org/html/rfc8834))
 - DTLS/UDP ([RFC7350](https://tools.ietf.org/html/rfc7350) and [RFC8261](https://tools.ietf.org/html/rfc8261))
 - ICE ([RFC8445](https://tools.ietf.org/html/rfc8445)) with STUN ([RFC8489](https://tools.ietf.org/html/rfc8489)) and its extension TURN ([RFC8656](https://tools.ietf.org/html/rfc8656))
 
 Features:
-- Full IPv6 support
-- Trickle ICE ([draft-ietf-ice-trickle-21](https://tools.ietf.org/html/draft-ietf-ice-trickle-21))
-- JSEP compatible ([draft-ietf-rtcweb-jsep-26](https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-26))
-- Multicast DNS candidates ([draft-ietf-rtcweb-mdns-ice-candidates-04](https://tools.ietf.org/html/draft-ietf-rtcweb-mdns-ice-candidates-04))
+- Full IPv6 support (as mandated by [RFC8835](https://tools.ietf.org/html/rfc8835))
+- Trickle ICE ([RFC8838](https://tools.ietf.org/html/rfc8838))
+- JSEP-compatible session establishment with SDP ([RFC8829](https://tools.ietf.org/html/rfc8829))
+- SCTP over DTLS with SDP offer/answer ([RFC8841](https://tools.ietf.org/html/rfc8841))
 - DTLS with ECDSA or RSA keys ([RFC8824](https://tools.ietf.org/html/rfc8827))
 - SRTP and SRTCP key derivation from DTLS ([RFC5764](https://tools.ietf.org/html/rfc5764))
+- Multicast DNS candidates ([draft-ietf-rtcweb-mdns-ice-candidates-04](https://tools.ietf.org/html/draft-ietf-rtcweb-mdns-ice-candidates-04))
 - Differentiated Services QoS ([draft-ietf-tsvwg-rtcweb-qos-18](https://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-18))
 
-Note only SDP BUNDLE mode is supported for media multiplexing ([draft-ietf-mmusic-sdp-bundle-negotiation-54](https://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation-54)). The behavior is equivalent to the JSEP bundle-only policy: the library always negociates one unique network component, where SRTP media streams are multiplexed with SRTCP control packets ([RFC5761](https://tools.ietf.org/html/rfc5761)) and SCTP/DTLS data traffic ([RFC5764](https://tools.ietf.org/html/rfc5764)).
+Note only SDP BUNDLE mode is supported for media multiplexing ([RFC8843](https://tools.ietf.org/html/rfc8843)). The behavior is equivalent to the JSEP bundle-only policy: the library always negociates one unique network component, where SRTP media streams are multiplexed with SRTCP control packets ([RFC5761](https://tools.ietf.org/html/rfc5761)) and SCTP/DTLS data traffic ([RFC5764](https://tools.ietf.org/html/rfc5764)).
 
 ### WebSocket
 
@@ -99,12 +100,12 @@ MY_ON_RECV_CANDIDATE_FROM_REMOTE([&pc](string candidate, string mid) {
 ### Observe the PeerConnection state
 
 ```cpp
-pc.onStateChange([](PeerConnection::State state) {
-    cout << "State: " << state << endl;
+pc.onStateChange([](rtc::PeerConnection::State state) {
+    std::cout << "State: " << state << std::endl;
 });
 
-pc.onGatheringStateChange([](PeerConnection::GatheringState state) {
-    cout << "Gathering state: " << state << endl;
+pc.onGatheringStateChange([](rtc::PeerConnection::GatheringState state) {
+    std::cout << "Gathering state: " << state << std::endl;
 });
 ```
 
@@ -114,12 +115,12 @@ pc.onGatheringStateChange([](PeerConnection::GatheringState state) {
 auto dc = pc.createDataChannel("test");
 
 dc->onOpen([]() {
-    cout << "Open" << endl;
+    std::cout << "Open" << std::endl;
 });
 
-dc->onMessage([](variant<binary, string> message) {
-    if (holds_alternative<string>(message)) {
-        cout << "Received: " << get<string>(message) << endl;
+dc->onMessage([](std::variant<binary, string> message) {
+    if (std::holds_alternative<string>(message)) {
+        std::cout << "Received: " << get<string>(message) << std::endl;
     }
 });
 ```
@@ -127,8 +128,8 @@ dc->onMessage([](variant<binary, string> message) {
 ### Receive a DataChannel
 
 ```cpp
-shared_ptr<rtc::DataChannel> dc;
-pc.onDataChannel([&dc](shared_ptr<rtc::DataChannel> incoming) {
+std::shared_ptr<rtc::DataChannel> dc;
+pc.onDataChannel([&dc](std::shared_ptr<rtc::DataChannel> incoming) {
     dc = incoming;
     dc->send("Hello world!");
 });
@@ -140,12 +141,12 @@ pc.onDataChannel([&dc](shared_ptr<rtc::DataChannel> incoming) {
 rtc::WebSocket ws;
 
 ws.onOpen([]() {
-	cout << "WebSocket open" << endl;
+    std::cout << "WebSocket open" << std::endl;
 });
 
-ws.onMessage([](variant<binary, string> message) {
-    if (holds_alternative<string>(message)) {
-        cout << "WebSocket received: " << get<string>(message) << endl;
+ws.onMessage([](std::variant<binary, string> message) {
+    if (std::holds_alternative<string>(message)) {
+        std::cout << "WebSocket received: " << std::get<string>(message) << endl;
     }
 });
 

src/description.cpp

@@ -243,9 +243,8 @@ string Description::generateSdp(string_view eol) const {
 	sdp << "s=-" << eol;
 	sdp << "t=0 0" << eol;
 
-	// Bundle
-	// see Negotiating Media Multiplexing Using the Session Description Protocol
-	// https://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation-54
+	// Bundle (RFC8843 Negotiating Media Multiplexing Using the Session Description Protocol)
+	// https://tools.ietf.org/html/rfc8843
 	sdp << "a=group:BUNDLE";
 	for (const auto &entry : mEntries)
 		sdp << ' ' << entry->mid();

src/impl/datachannel.cpp

@@ -36,8 +36,8 @@ using std::chrono::milliseconds;
 
 namespace rtc::impl {
 
-// Messages for the DataChannel establishment protocol
-// See https://tools.ietf.org/html/draft-ietf-rtcweb-data-protocol-09
+// Messages for the DataChannel establishment protocol (RFC 8832)
+// See https://tools.ietf.org/html/rfc8832
 
 enum MessageType : uint8_t {
 	MESSAGE_OPEN_REQUEST = 0x00,

src/impl/peerconnection.cpp

@@ -541,9 +541,11 @@ shared_ptr<DataChannel> PeerConnection::emplaceDataChannel(Description::Role rol
 		if (stream == 65535)
 			throw std::invalid_argument("Invalid DataChannel id");
 	} else {
-		// The active side must use streams with even identifiers, whereas the passive side must use
-		// streams with odd identifiers.
-		// See https://tools.ietf.org/html/draft-ietf-rtcweb-data-protocol-09#section-6
+		// RFC 8832: The peer that initiates opening a data channel selects a stream identifier for
+		// which the corresponding incoming and outgoing streams are unused.  If the side is acting
+		// as the DTLS client, it MUST choose an even stream identifier; if the side is acting as
+		// the DTLS server, it MUST choose an odd one.
+		// See https://tools.ietf.org/html/rfc8832#section-6
 		stream = (role == Description::Role::Active) ? 0 : 1;
 		while (mDataChannels.find(stream) != mDataChannels.end()) {
 			if (stream >= 65535 - 2)

src/impl/sctptransport.cpp

@@ -27,10 +27,9 @@
 #include <thread>
 #include <vector>
 
-// The IETF draft says:
-// SCTP MUST support performing Path MTU discovery without relying on ICMP or ICMPv6 as specified in
-// [RFC4821] using probing messages specified in [RFC4820].
-// See https://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-13#section-5
+// RFC 8831: SCTP MUST support performing Path MTU discovery without relying on ICMP or ICMPv6 as
+// specified in [RFC4821] by using probing messages specified in [RFC4820].
+// See https://tools.ietf.org/html/rfc8831#section-5
 //
 // However, usrsctp does not implement Path MTU discovery, so we need to disable it for now.
 // See https://github.com/sctplab/usrsctp/issues/205
@@ -101,9 +100,8 @@ void SctpTransport::Cleanup() {
 		std::this_thread::sleep_for(100ms);
 }
 
-SctpTransport::SctpTransport(shared_ptr<Transport> lower, uint16_t port,
-                             optional<size_t> mtu, message_callback recvCallback,
-                             amount_callback bufferedAmountCallback,
+SctpTransport::SctpTransport(shared_ptr<Transport> lower, uint16_t port, optional<size_t> mtu,
+                             message_callback recvCallback, amount_callback bufferedAmountCallback,
                              state_callback stateChangeCallback)
     : Transport(lower, std::move(stateChangeCallback)), mPort(port),
       mSendQueue(0, message_size_func), mBufferedAmountCallback(std::move(bufferedAmountCallback)) {
@@ -161,8 +159,9 @@ SctpTransport::SctpTransport(shared_ptr<Transport> lower, uint16_t port,
 		throw std::runtime_error("Could not subscribe to event SCTP_STREAM_RESET_EVENT, errno=" +
 		                         std::to_string(errno));
 
-	// The sender SHOULD disable the Nagle algorithm (see RFC1122) to minimize the latency.
-	// See https://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-13#section-6.6
+	// RFC 8831 6.6. Transferring User Data on a Data Channel
+	// The sender SHOULD disable the Nagle algorithm (see [RFC1122) to minimize the latency
+	// See https://tools.ietf.org/html/rfc8831#section-6.6
 	int nodelay = 1;
 	if (usrsctp_setsockopt(mSock, IPPROTO_SCTP, SCTP_NODELAY, &nodelay, sizeof(nodelay)))
 		throw std::runtime_error("Could not set socket option SCTP_NODELAY, errno=" +
@@ -203,9 +202,10 @@ SctpTransport::SctpTransport(shared_ptr<Transport> lower, uint16_t port,
 		throw std::runtime_error("Could not set socket option SCTP_PEER_ADDR_PARAMS, errno=" +
 		                         std::to_string(errno));
 
-	// The IETF draft recommends the number of streams negotiated during SCTP association to be
-	// 65535.
-	// See https://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-13#section-6.2
+	// RFC 8831 6.2. SCTP Association Management
+	// The number of streams negotiated during SCTP association setup SHOULD be 65535, which is the
+	// maximum number of streams that can be negotiated during the association setup.
+	// See https://tools.ietf.org/html/rfc8831#section-6.2
 	struct sctp_initmsg sinit = {};
 	sinit.sinit_num_ostreams = 65535;
 	sinit.sinit_max_instreams = 65535;
@@ -213,7 +213,7 @@ SctpTransport::SctpTransport(shared_ptr<Transport> lower, uint16_t port,
 		throw std::runtime_error("Could not set socket option SCTP_INITMSG, errno=" +
 		                         std::to_string(errno));
 
-	// Prevent fragmented interleave of messages (i.e. level 0), see RFC 6458 8.1.20.
+	// Prevent fragmented interleave of messages (i.e. level 0), see RFC 6458 section 8.1.20.
 	// Unless the user has set the fragmentation interleave level to 0, notifications
 	// may also be interleaved with partially delivered messages.
 	int level = 0;
@@ -293,9 +293,9 @@ void SctpTransport::connect() {
 	if (usrsctp_bind(mSock, reinterpret_cast<struct sockaddr *>(&sconn), sizeof(sconn)))
 		throw std::runtime_error("Could not bind usrsctp socket, errno=" + std::to_string(errno));
 
-	// According to the IETF draft, both endpoints must initiate the SCTP association, in a
+	// According to RFC 8841, both endpoints must initiate the SCTP association, in a
 	// simultaneous-open manner, irrelevent to the SDP setup role.
-	// See https://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-26#section-9.3
+	// See https://tools.ietf.org/html/rfc8841#section-9.3
 	int ret = usrsctp_connect(mSock, reinterpret_cast<struct sockaddr *>(&sconn), sizeof(sconn));
 	if (ret && errno != EINPROGRESS)
 		throw std::runtime_error("Connection attempt failed, errno=" + std::to_string(errno));
@@ -632,9 +632,11 @@ int SctpTransport::handleWrite(byte *data, size_t len, uint8_t /*tos*/, uint8_t
 void SctpTransport::processData(binary &&data, uint16_t sid, PayloadId ppid) {
 	PLOG_VERBOSE << "Process data, size=" << data.size();
 
-	// The usage of the PPIDs "WebRTC String Partial" and "WebRTC Binary Partial" is deprecated.
-	// See https://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-13#section-6.6
-	// We handle them at reception for compatibility reasons but should never send them.
+	// RFC 8831: The usage of the PPIDs "WebRTC String Partial" and "WebRTC Binary Partial" is
+	// deprecated. They were used for a PPID-based fragmentation and reassembly of user messages
+	// belonging to reliable and ordered data channels.
+	// See https://tools.ietf.org/html/rfc8831#section-6.6
+	// We handle those PPIDs at reception for compatibility reasons but shall never send them.
 	switch (ppid) {
 	case PPID_CONTROL:
 		recv(make_message(std::move(data), Message::Control, sid));