|
|
@@ -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));
|
Paul-Louis Ageneau