| /* |
| * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include <errno.h> |
| namespace { |
| // Some ERRNO values get re-#defined to WSA* equivalents in some talk/ |
| // headers. We save the original ones in an enum. |
| enum PreservedErrno { |
| SCTP_EINPROGRESS = EINPROGRESS, |
| SCTP_EWOULDBLOCK = EWOULDBLOCK |
| }; |
| } // namespace |
| |
| #include "media/sctp/sctptransport.h" |
| |
| #include <stdarg.h> |
| #include <stdio.h> |
| |
| #include <algorithm> |
| #include <memory> |
| |
| #include "media/base/codec.h" |
| #include "media/base/mediaconstants.h" |
| #include "media/base/streamparams.h" |
| #include "p2p/base/dtlstransportinternal.h" // For PF_NORMAL |
| #include "rtc_base/arraysize.h" |
| #include "rtc_base/copyonwritebuffer.h" |
| #include "rtc_base/criticalsection.h" |
| #include "rtc_base/helpers.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/numerics/safe_conversions.h" |
| #include "rtc_base/stringutils.h" |
| #include "rtc_base/thread_checker.h" |
| #include "rtc_base/trace_event.h" |
| #include "usrsctplib/usrsctp.h" |
| |
| namespace { |
| |
| // The biggest SCTP packet. Starting from a 'safe' wire MTU value of 1280, |
| // take off 80 bytes for DTLS/TURN/TCP/IP overhead. |
| static constexpr size_t kSctpMtu = 1200; |
| |
| // The size of the SCTP association send buffer. 256kB, the usrsctp default. |
| static constexpr int kSendBufferSize = 256 * 1024; |
| |
| // Set the initial value of the static SCTP Data Engines reference count. |
| int g_usrsctp_usage_count = 0; |
| rtc::GlobalLockPod g_usrsctp_lock_; |
| |
| // DataMessageType is used for the SCTP "Payload Protocol Identifier", as |
| // defined in http://tools.ietf.org/html/rfc4960#section-14.4 |
| // |
| // For the list of IANA approved values see: |
| // http://www.iana.org/assignments/sctp-parameters/sctp-parameters.xml |
| // The value is not used by SCTP itself. It indicates the protocol running |
| // on top of SCTP. |
| enum PayloadProtocolIdentifier { |
| PPID_NONE = 0, // No protocol is specified. |
| // Matches the PPIDs in mozilla source and |
| // https://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-protocol Sec. 9 |
| // They're not yet assigned by IANA. |
| PPID_CONTROL = 50, |
| PPID_BINARY_PARTIAL = 52, |
| PPID_BINARY_LAST = 53, |
| PPID_TEXT_PARTIAL = 54, |
| PPID_TEXT_LAST = 51 |
| }; |
| |
| // Helper for logging SCTP messages. |
| #if defined(__GNUC__) |
| __attribute__((__format__(__printf__, 1, 2))) |
| #endif |
| void DebugSctpPrintf(const char* format, ...) { |
| #if RTC_DCHECK_IS_ON |
| char s[255]; |
| va_list ap; |
| va_start(ap, format); |
| vsnprintf(s, sizeof(s), format, ap); |
| RTC_LOG(LS_INFO) << "SCTP: " << s; |
| va_end(ap); |
| #endif |
| } |
| |
| // Get the PPID to use for the terminating fragment of this type. |
| PayloadProtocolIdentifier GetPpid(cricket::DataMessageType type) { |
| switch (type) { |
| default: |
| case cricket::DMT_NONE: |
| return PPID_NONE; |
| case cricket::DMT_CONTROL: |
| return PPID_CONTROL; |
| case cricket::DMT_BINARY: |
| return PPID_BINARY_LAST; |
| case cricket::DMT_TEXT: |
| return PPID_TEXT_LAST; |
| } |
| } |
| |
| bool GetDataMediaType(PayloadProtocolIdentifier ppid, |
| cricket::DataMessageType* dest) { |
| RTC_DCHECK(dest != NULL); |
| switch (ppid) { |
| case PPID_BINARY_PARTIAL: |
| case PPID_BINARY_LAST: |
| *dest = cricket::DMT_BINARY; |
| return true; |
| |
| case PPID_TEXT_PARTIAL: |
| case PPID_TEXT_LAST: |
| *dest = cricket::DMT_TEXT; |
| return true; |
| |
| case PPID_CONTROL: |
| *dest = cricket::DMT_CONTROL; |
| return true; |
| |
| case PPID_NONE: |
| *dest = cricket::DMT_NONE; |
| return true; |
| |
| default: |
| return false; |
| } |
| } |
| |
| // Log the packet in text2pcap format, if log level is at LS_VERBOSE. |
| // |
| // In order to turn these logs into a pcap file you can use, first filter the |
| // "SCTP_PACKET" log lines: |
| // |
| // cat chrome_debug.log | grep SCTP_PACKET > filtered.log |
| // |
| // Then run through text2pcap: |
| // |
| // text2pcap -t "%H:%M:%S." -D -u 1024,1024 filtered.log filtered.pcap |
| // |
| // The value "1024" isn't important, we just need a port for the dummy UDP |
| // headers generated. Lastly, you should be able to open filtered.pcap in |
| // Wireshark, then right click a packet and "Decode As..." SCTP. |
| // |
| // Why do all this? Because SCTP goes over DTLS, which is encrypted. So just |
| // getting a normal packet capture won't help you, unless you have the DTLS |
| // keying material. |
| void VerboseLogPacket(const void* data, size_t length, int direction) { |
| if (RTC_LOG_CHECK_LEVEL(LS_VERBOSE) && length > 0) { |
| char* dump_buf; |
| // Some downstream project uses an older version of usrsctp that expects |
| // a non-const "void*" as first parameter when dumping the packet, so we |
| // need to cast the const away here to avoid a compiler error. |
| if ((dump_buf = usrsctp_dumppacket(const_cast<void*>(data), length, |
| direction)) != NULL) { |
| RTC_LOG(LS_VERBOSE) << dump_buf; |
| usrsctp_freedumpbuffer(dump_buf); |
| } |
| } |
| } |
| |
| } // namespace |
| |
| namespace cricket { |
| |
| // Handles global init/deinit, and mapping from usrsctp callbacks to |
| // SctpTransport calls. |
| class SctpTransport::UsrSctpWrapper { |
| public: |
| static void InitializeUsrSctp() { |
| RTC_LOG(LS_INFO) << __FUNCTION__; |
| // First argument is udp_encapsulation_port, which is not releveant for our |
| // AF_CONN use of sctp. |
| usrsctp_init(0, &UsrSctpWrapper::OnSctpOutboundPacket, &DebugSctpPrintf); |
| |
| // To turn on/off detailed SCTP debugging. You will also need to have the |
| // SCTP_DEBUG cpp defines flag. |
| // usrsctp_sysctl_set_sctp_debug_on(SCTP_DEBUG_ALL); |
| |
| // TODO(ldixon): Consider turning this on/off. |
| usrsctp_sysctl_set_sctp_ecn_enable(0); |
| |
| // This is harmless, but we should find out when the library default |
| // changes. |
| int send_size = usrsctp_sysctl_get_sctp_sendspace(); |
| if (send_size != kSendBufferSize) { |
| RTC_LOG(LS_ERROR) << "Got different send size than expected: " |
| << send_size; |
| } |
| |
| // TODO(ldixon): Consider turning this on/off. |
| // This is not needed right now (we don't do dynamic address changes): |
| // If SCTP Auto-ASCONF is enabled, the peer is informed automatically |
| // when a new address is added or removed. This feature is enabled by |
| // default. |
| // usrsctp_sysctl_set_sctp_auto_asconf(0); |
| |
| // TODO(ldixon): Consider turning this on/off. |
| // Add a blackhole sysctl. Setting it to 1 results in no ABORTs |
| // being sent in response to INITs, setting it to 2 results |
| // in no ABORTs being sent for received OOTB packets. |
| // This is similar to the TCP sysctl. |
| // |
| // See: http://lakerest.net/pipermail/sctp-coders/2012-January/009438.html |
| // See: http://svnweb.freebsd.org/base?view=revision&revision=229805 |
| // usrsctp_sysctl_set_sctp_blackhole(2); |
| |
| // Set the number of default outgoing streams. This is the number we'll |
| // send in the SCTP INIT message. |
| usrsctp_sysctl_set_sctp_nr_outgoing_streams_default(kMaxSctpStreams); |
| } |
| |
| static void UninitializeUsrSctp() { |
| RTC_LOG(LS_INFO) << __FUNCTION__; |
| // usrsctp_finish() may fail if it's called too soon after the transports |
| // are |
| // closed. Wait and try again until it succeeds for up to 3 seconds. |
| for (size_t i = 0; i < 300; ++i) { |
| if (usrsctp_finish() == 0) { |
| return; |
| } |
| |
| rtc::Thread::SleepMs(10); |
| } |
| RTC_LOG(LS_ERROR) << "Failed to shutdown usrsctp."; |
| } |
| |
| static void IncrementUsrSctpUsageCount() { |
| rtc::GlobalLockScope lock(&g_usrsctp_lock_); |
| if (!g_usrsctp_usage_count) { |
| InitializeUsrSctp(); |
| } |
| ++g_usrsctp_usage_count; |
| } |
| |
| static void DecrementUsrSctpUsageCount() { |
| rtc::GlobalLockScope lock(&g_usrsctp_lock_); |
| --g_usrsctp_usage_count; |
| if (!g_usrsctp_usage_count) { |
| UninitializeUsrSctp(); |
| } |
| } |
| |
| // This is the callback usrsctp uses when there's data to send on the network |
| // that has been wrapped appropriatly for the SCTP protocol. |
| static int OnSctpOutboundPacket(void* addr, |
| void* data, |
| size_t length, |
| uint8_t tos, |
| uint8_t set_df) { |
| SctpTransport* transport = static_cast<SctpTransport*>(addr); |
| RTC_LOG(LS_VERBOSE) << "global OnSctpOutboundPacket():" |
| << "addr: " << addr << "; length: " << length |
| << "; tos: " << rtc::ToHex(tos) |
| << "; set_df: " << rtc::ToHex(set_df); |
| |
| VerboseLogPacket(data, length, SCTP_DUMP_OUTBOUND); |
| // Note: We have to copy the data; the caller will delete it. |
| rtc::CopyOnWriteBuffer buf(reinterpret_cast<uint8_t*>(data), length); |
| // TODO(deadbeef): Why do we need an AsyncInvoke here? We're already on the |
| // right thread and don't need to unwind the stack. |
| transport->invoker_.AsyncInvoke<void>( |
| RTC_FROM_HERE, transport->network_thread_, |
| rtc::Bind(&SctpTransport::OnPacketFromSctpToNetwork, transport, buf)); |
| return 0; |
| } |
| |
| // This is the callback called from usrsctp when data has been received, after |
| // a packet has been interpreted and parsed by usrsctp and found to contain |
| // payload data. It is called by a usrsctp thread. It is assumed this function |
| // will free the memory used by 'data'. |
| static int OnSctpInboundPacket(struct socket* sock, |
| union sctp_sockstore addr, |
| void* data, |
| size_t length, |
| struct sctp_rcvinfo rcv, |
| int flags, |
| void* ulp_info) { |
| SctpTransport* transport = static_cast<SctpTransport*>(ulp_info); |
| // Post data to the transport's receiver thread (copying it). |
| // TODO(ldixon): Unclear if copy is needed as this method is responsible for |
| // memory cleanup. But this does simplify code. |
| const PayloadProtocolIdentifier ppid = |
| static_cast<PayloadProtocolIdentifier>( |
| rtc::HostToNetwork32(rcv.rcv_ppid)); |
| DataMessageType type = DMT_NONE; |
| if (!GetDataMediaType(ppid, &type) && !(flags & MSG_NOTIFICATION)) { |
| // It's neither a notification nor a recognized data packet. Drop it. |
| RTC_LOG(LS_ERROR) << "Received an unknown PPID " << ppid |
| << " on an SCTP packet. Dropping."; |
| free(data); |
| } else { |
| ReceiveDataParams params; |
| |
| params.sid = rcv.rcv_sid; |
| params.seq_num = rcv.rcv_ssn; |
| params.timestamp = rcv.rcv_tsn; |
| params.type = type; |
| |
| // Expect only continuation messages belonging to the same sid, the sctp |
| // stack should ensure this. |
| if ((transport->partial_message_.size() != 0) && |
| (rcv.rcv_sid != transport->partial_params_.sid)) { |
| // A message with a new sid, but haven't seen the EOR for the |
| // previous message. Deliver the previous partial message to avoid |
| // merging messages from different sid's. |
| transport->invoker_.AsyncInvoke<void>( |
| RTC_FROM_HERE, transport->network_thread_, |
| rtc::Bind(&SctpTransport::OnInboundPacketFromSctpToTransport, |
| transport, transport->partial_message_, |
| transport->partial_params_, transport->partial_flags_)); |
| |
| transport->partial_message_.Clear(); |
| } |
| |
| transport->partial_message_.AppendData(reinterpret_cast<uint8_t*>(data), |
| length); |
| transport->partial_params_ = params; |
| transport->partial_flags_ = flags; |
| |
| free(data); |
| |
| // Merge partial messages until they exceed the maximum send buffer size. |
| // This enables messages from a single send to be delivered in a single |
| // callback. Larger messages (originating from other implementations) will |
| // still be delivered in chunks. |
| if (!(flags & MSG_EOR) && |
| (transport->partial_message_.size() < kSendBufferSize)) { |
| return 1; |
| } |
| |
| // The ownership of the packet transfers to |invoker_|. Using |
| // CopyOnWriteBuffer is the most convenient way to do this. |
| transport->invoker_.AsyncInvoke<void>( |
| RTC_FROM_HERE, transport->network_thread_, |
| rtc::Bind(&SctpTransport::OnInboundPacketFromSctpToTransport, |
| transport, transport->partial_message_, params, flags)); |
| |
| transport->partial_message_.Clear(); |
| } |
| return 1; |
| } |
| |
| static SctpTransport* GetTransportFromSocket(struct socket* sock) { |
| struct sockaddr* addrs = nullptr; |
| int naddrs = usrsctp_getladdrs(sock, 0, &addrs); |
| if (naddrs <= 0 || addrs[0].sa_family != AF_CONN) { |
| return nullptr; |
| } |
| // usrsctp_getladdrs() returns the addresses bound to this socket, which |
| // contains the SctpTransport* as sconn_addr. Read the pointer, |
| // then free the list of addresses once we have the pointer. We only open |
| // AF_CONN sockets, and they should all have the sconn_addr set to the |
| // pointer that created them, so [0] is as good as any other. |
| struct sockaddr_conn* sconn = |
| reinterpret_cast<struct sockaddr_conn*>(&addrs[0]); |
| SctpTransport* transport = |
| reinterpret_cast<SctpTransport*>(sconn->sconn_addr); |
| usrsctp_freeladdrs(addrs); |
| |
| return transport; |
| } |
| |
| static int SendThresholdCallback(struct socket* sock, uint32_t sb_free) { |
| // Fired on our I/O thread. SctpTransport::OnPacketReceived() gets |
| // a packet containing acknowledgments, which goes into usrsctp_conninput, |
| // and then back here. |
| SctpTransport* transport = GetTransportFromSocket(sock); |
| if (!transport) { |
| RTC_LOG(LS_ERROR) |
| << "SendThresholdCallback: Failed to get transport for socket " |
| << sock; |
| return 0; |
| } |
| transport->OnSendThresholdCallback(); |
| return 0; |
| } |
| }; |
| |
| SctpTransport::SctpTransport(rtc::Thread* network_thread, |
| rtc::PacketTransportInternal* transport) |
| : network_thread_(network_thread), |
| transport_(transport), |
| was_ever_writable_(transport->writable()) { |
| RTC_DCHECK(network_thread_); |
| RTC_DCHECK(transport_); |
| RTC_DCHECK_RUN_ON(network_thread_); |
| ConnectTransportSignals(); |
| } |
| |
| SctpTransport::~SctpTransport() { |
| // Close abruptly; no reset procedure. |
| CloseSctpSocket(); |
| } |
| |
| void SctpTransport::SetDtlsTransport(rtc::PacketTransportInternal* transport) { |
| RTC_DCHECK_RUN_ON(network_thread_); |
| DisconnectTransportSignals(); |
| transport_ = transport; |
| ConnectTransportSignals(); |
| if (!was_ever_writable_ && transport && transport->writable()) { |
| was_ever_writable_ = true; |
| // New transport is writable, now we can start the SCTP connection if Start |
| // was called already. |
| if (started_) { |
| RTC_DCHECK(!sock_); |
| Connect(); |
| } |
| } |
| } |
| |
| bool SctpTransport::Start(int local_sctp_port, int remote_sctp_port) { |
| RTC_DCHECK_RUN_ON(network_thread_); |
| if (local_sctp_port == -1) { |
| local_sctp_port = kSctpDefaultPort; |
| } |
| if (remote_sctp_port == -1) { |
| remote_sctp_port = kSctpDefaultPort; |
| } |
| if (started_) { |
| if (local_sctp_port != local_port_ || remote_sctp_port != remote_port_) { |
| RTC_LOG(LS_ERROR) |
| << "Can't change SCTP port after SCTP association formed."; |
| return false; |
| } |
| return true; |
| } |
| local_port_ = local_sctp_port; |
| remote_port_ = remote_sctp_port; |
| started_ = true; |
| RTC_DCHECK(!sock_); |
| // Only try to connect if the DTLS transport has been writable before |
| // (indicating that the DTLS handshake is complete). |
| if (was_ever_writable_) { |
| return Connect(); |
| } |
| return true; |
| } |
| |
| bool SctpTransport::OpenStream(int sid) { |
| RTC_DCHECK_RUN_ON(network_thread_); |
| if (sid > kMaxSctpSid) { |
| RTC_LOG(LS_WARNING) << debug_name_ << "->OpenStream(...): " |
| << "Not adding data stream " |
| << "with sid=" << sid << " because sid is too high."; |
| return false; |
| } |
| auto it = stream_status_by_sid_.find(sid); |
| if (it == stream_status_by_sid_.end()) { |
| stream_status_by_sid_[sid] = StreamStatus(); |
| return true; |
| } |
| if (it->second.is_open()) { |
| RTC_LOG(LS_WARNING) << debug_name_ << "->OpenStream(...): " |
| << "Not adding data stream " |
| << "with sid=" << sid |
| << " because stream is already open."; |
| return false; |
| } else { |
| RTC_LOG(LS_WARNING) << debug_name_ << "->OpenStream(...): " |
| << "Not adding data stream " |
| << " with sid=" << sid |
| << " because stream is still closing."; |
| return false; |
| } |
| } |
| |
| bool SctpTransport::ResetStream(int sid) { |
| RTC_DCHECK_RUN_ON(network_thread_); |
| |
| auto it = stream_status_by_sid_.find(sid); |
| if (it == stream_status_by_sid_.end() || !it->second.is_open()) { |
| RTC_LOG(LS_WARNING) << debug_name_ << "->ResetStream(" << sid |
| << "): stream not open."; |
| return false; |
| } |
| |
| RTC_LOG(LS_VERBOSE) << debug_name_ << "->ResetStream(" << sid << "): " |
| << "Queuing RE-CONFIG chunk."; |
| it->second.closure_initiated = true; |
| |
| // Signal our stream-reset logic that it should try to send now, if it can. |
| SendQueuedStreamResets(); |
| |
| // The stream will actually get removed when we get the acknowledgment. |
| return true; |
| } |
| |
| bool SctpTransport::SendData(const SendDataParams& params, |
| const rtc::CopyOnWriteBuffer& payload, |
| SendDataResult* result) { |
| RTC_DCHECK_RUN_ON(network_thread_); |
| if (result) { |
| // Preset |result| to assume an error. If SendData succeeds, we'll |
| // overwrite |*result| once more at the end. |
| *result = SDR_ERROR; |
| } |
| |
| if (!sock_) { |
| RTC_LOG(LS_WARNING) << debug_name_ << "->SendData(...): " |
| << "Not sending packet with sid=" << params.sid |
| << " len=" << payload.size() << " before Start()."; |
| return false; |
| } |
| |
| if (params.type != DMT_CONTROL) { |
| auto it = stream_status_by_sid_.find(params.sid); |
| if (it == stream_status_by_sid_.end() || !it->second.is_open()) { |
| RTC_LOG(LS_WARNING) |
| << debug_name_ << "->SendData(...): " |
| << "Not sending data because sid is unknown or closing: " |
| << params.sid; |
| return false; |
| } |
| } |
| |
| // Send data using SCTP. |
| ssize_t send_res = 0; // result from usrsctp_sendv. |
| struct sctp_sendv_spa spa = {0}; |
| spa.sendv_flags |= SCTP_SEND_SNDINFO_VALID; |
| spa.sendv_sndinfo.snd_sid = params.sid; |
| spa.sendv_sndinfo.snd_ppid = rtc::HostToNetwork32(GetPpid(params.type)); |
| spa.sendv_sndinfo.snd_flags |= SCTP_EOR; |
| |
| // Ordered implies reliable. |
| if (!params.ordered) { |
| spa.sendv_sndinfo.snd_flags |= SCTP_UNORDERED; |
| if (params.max_rtx_count >= 0 || params.max_rtx_ms == 0) { |
| spa.sendv_flags |= SCTP_SEND_PRINFO_VALID; |
| spa.sendv_prinfo.pr_policy = SCTP_PR_SCTP_RTX; |
| spa.sendv_prinfo.pr_value = params.max_rtx_count; |
| } else { |
| spa.sendv_flags |= SCTP_SEND_PRINFO_VALID; |
| spa.sendv_prinfo.pr_policy = SCTP_PR_SCTP_TTL; |
| spa.sendv_prinfo.pr_value = params.max_rtx_ms; |
| } |
| } |
| |
| // We don't fragment. |
| send_res = usrsctp_sendv( |
| sock_, payload.data(), static_cast<size_t>(payload.size()), NULL, 0, &spa, |
| rtc::checked_cast<socklen_t>(sizeof(spa)), SCTP_SENDV_SPA, 0); |
| if (send_res < 0) { |
| if (errno == SCTP_EWOULDBLOCK) { |
| *result = SDR_BLOCK; |
| ready_to_send_data_ = false; |
| RTC_LOG(LS_INFO) << debug_name_ |
| << "->SendData(...): EWOULDBLOCK returned"; |
| } else { |
| RTC_LOG_ERRNO(LS_ERROR) << "ERROR:" << debug_name_ << "->SendData(...): " |
| << " usrsctp_sendv: "; |
| } |
| return false; |
| } |
| if (result) { |
| // Only way out now is success. |
| *result = SDR_SUCCESS; |
| } |
| return true; |
| } |
| |
| bool SctpTransport::ReadyToSendData() { |
| RTC_DCHECK_RUN_ON(network_thread_); |
| return ready_to_send_data_; |
| } |
| |
| void SctpTransport::ConnectTransportSignals() { |
| RTC_DCHECK_RUN_ON(network_thread_); |
| if (!transport_) { |
| return; |
| } |
| transport_->SignalWritableState.connect(this, |
| &SctpTransport::OnWritableState); |
| transport_->SignalReadPacket.connect(this, &SctpTransport::OnPacketRead); |
| } |
| |
| void SctpTransport::DisconnectTransportSignals() { |
| RTC_DCHECK_RUN_ON(network_thread_); |
| if (!transport_) { |
| return; |
| } |
| transport_->SignalWritableState.disconnect(this); |
| transport_->SignalReadPacket.disconnect(this); |
| } |
| |
| bool SctpTransport::Connect() { |
| RTC_DCHECK_RUN_ON(network_thread_); |
| RTC_LOG(LS_VERBOSE) << debug_name_ << "->Connect()."; |
| |
| // If we already have a socket connection (which shouldn't ever happen), just |
| // return. |
| RTC_DCHECK(!sock_); |
| if (sock_) { |
| RTC_LOG(LS_ERROR) << debug_name_ |
| << "->Connect(): Ignored as socket " |
| "is already established."; |
| return true; |
| } |
| |
| // If no socket (it was closed) try to start it again. This can happen when |
| // the socket we are connecting to closes, does an sctp shutdown handshake, |
| // or behaves unexpectedly causing us to perform a CloseSctpSocket. |
| if (!OpenSctpSocket()) { |
| return false; |
| } |
| |
| // Note: conversion from int to uint16_t happens on assignment. |
| sockaddr_conn local_sconn = GetSctpSockAddr(local_port_); |
| if (usrsctp_bind(sock_, reinterpret_cast<sockaddr*>(&local_sconn), |
| sizeof(local_sconn)) < 0) { |
| RTC_LOG_ERRNO(LS_ERROR) |
| << debug_name_ << "->Connect(): " << ("Failed usrsctp_bind"); |
| CloseSctpSocket(); |
| return false; |
| } |
| |
| // Note: conversion from int to uint16_t happens on assignment. |
| sockaddr_conn remote_sconn = GetSctpSockAddr(remote_port_); |
| int connect_result = usrsctp_connect( |
| sock_, reinterpret_cast<sockaddr*>(&remote_sconn), sizeof(remote_sconn)); |
| if (connect_result < 0 && errno != SCTP_EINPROGRESS) { |
| RTC_LOG_ERRNO(LS_ERROR) << debug_name_ << "->Connect(): " |
| << "Failed usrsctp_connect. got errno=" << errno |
| << ", but wanted " << SCTP_EINPROGRESS; |
| CloseSctpSocket(); |
| return false; |
| } |
| // Set the MTU and disable MTU discovery. |
| // We can only do this after usrsctp_connect or it has no effect. |
| sctp_paddrparams params = {{0}}; |
| memcpy(¶ms.spp_address, &remote_sconn, sizeof(remote_sconn)); |
| params.spp_flags = SPP_PMTUD_DISABLE; |
| // The MTU value provided specifies the space available for chunks in the |
| // packet, so we subtract the SCTP header size. |
| params.spp_pathmtu = kSctpMtu - sizeof(struct sctp_common_header); |
| if (usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_PEER_ADDR_PARAMS, ¶ms, |
| sizeof(params))) { |
| RTC_LOG_ERRNO(LS_ERROR) << debug_name_ << "->Connect(): " |
| << "Failed to set SCTP_PEER_ADDR_PARAMS."; |
| } |
| // Since this is a fresh SCTP association, we'll always start out with empty |
| // queues, so "ReadyToSendData" should be true. |
| SetReadyToSendData(); |
| return true; |
| } |
| |
| bool SctpTransport::OpenSctpSocket() { |
| RTC_DCHECK_RUN_ON(network_thread_); |
| if (sock_) { |
| RTC_LOG(LS_WARNING) << debug_name_ << "->OpenSctpSocket(): " |
| << "Ignoring attempt to re-create existing socket."; |
| return false; |
| } |
| |
| UsrSctpWrapper::IncrementUsrSctpUsageCount(); |
| |
| // If kSendBufferSize isn't reflective of reality, we log an error, but we |
| // still have to do something reasonable here. Look up what the buffer's |
| // real size is and set our threshold to something reasonable. |
| static const int kSendThreshold = usrsctp_sysctl_get_sctp_sendspace() / 2; |
| |
| sock_ = usrsctp_socket( |
| AF_CONN, SOCK_STREAM, IPPROTO_SCTP, &UsrSctpWrapper::OnSctpInboundPacket, |
| &UsrSctpWrapper::SendThresholdCallback, kSendThreshold, this); |
| if (!sock_) { |
| RTC_LOG_ERRNO(LS_ERROR) << debug_name_ << "->OpenSctpSocket(): " |
| << "Failed to create SCTP socket."; |
| UsrSctpWrapper::DecrementUsrSctpUsageCount(); |
| return false; |
| } |
| |
| if (!ConfigureSctpSocket()) { |
| usrsctp_close(sock_); |
| sock_ = nullptr; |
| UsrSctpWrapper::DecrementUsrSctpUsageCount(); |
| return false; |
| } |
| // Register this class as an address for usrsctp. This is used by SCTP to |
| // direct the packets received (by the created socket) to this class. |
| usrsctp_register_address(this); |
| return true; |
| } |
| |
| bool SctpTransport::ConfigureSctpSocket() { |
| RTC_DCHECK_RUN_ON(network_thread_); |
| RTC_DCHECK(sock_); |
| // Make the socket non-blocking. Connect, close, shutdown etc will not block |
| // the thread waiting for the socket operation to complete. |
| if (usrsctp_set_non_blocking(sock_, 1) < 0) { |
| RTC_LOG_ERRNO(LS_ERROR) << debug_name_ << "->ConfigureSctpSocket(): " |
| << "Failed to set SCTP to non blocking."; |
| return false; |
| } |
| |
| // This ensures that the usrsctp close call deletes the association. This |
| // prevents usrsctp from calling OnSctpOutboundPacket with references to |
| // this class as the address. |
| linger linger_opt; |
| linger_opt.l_onoff = 1; |
| linger_opt.l_linger = 0; |
| if (usrsctp_setsockopt(sock_, SOL_SOCKET, SO_LINGER, &linger_opt, |
| sizeof(linger_opt))) { |
| RTC_LOG_ERRNO(LS_ERROR) << debug_name_ << "->ConfigureSctpSocket(): " |
| << "Failed to set SO_LINGER."; |
| return false; |
| } |
| |
| // Enable stream ID resets. |
| struct sctp_assoc_value stream_rst; |
| stream_rst.assoc_id = SCTP_ALL_ASSOC; |
| stream_rst.assoc_value = 1; |
| if (usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_ENABLE_STREAM_RESET, |
| &stream_rst, sizeof(stream_rst))) { |
| RTC_LOG_ERRNO(LS_ERROR) << debug_name_ << "->ConfigureSctpSocket(): " |
| |
| << "Failed to set SCTP_ENABLE_STREAM_RESET."; |
| return false; |
| } |
| |
| // Nagle. |
| uint32_t nodelay = 1; |
| if (usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_NODELAY, &nodelay, |
| sizeof(nodelay))) { |
| RTC_LOG_ERRNO(LS_ERROR) << debug_name_ << "->ConfigureSctpSocket(): " |
| << "Failed to set SCTP_NODELAY."; |
| return false; |
| } |
| |
| // Explicit EOR. |
| uint32_t eor = 1; |
| if (usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_EXPLICIT_EOR, &eor, |
| sizeof(eor))) { |
| RTC_LOG_ERRNO(LS_ERROR) << debug_name_ << "->ConfigureSctpSocket(): " |
| << "Failed to set SCTP_EXPLICIT_EOR."; |
| return false; |
| } |
| |
| // Subscribe to SCTP event notifications. |
| int event_types[] = {SCTP_ASSOC_CHANGE, SCTP_PEER_ADDR_CHANGE, |
| SCTP_SEND_FAILED_EVENT, SCTP_SENDER_DRY_EVENT, |
| SCTP_STREAM_RESET_EVENT}; |
| struct sctp_event event = {0}; |
| event.se_assoc_id = SCTP_ALL_ASSOC; |
| event.se_on = 1; |
| for (size_t i = 0; i < arraysize(event_types); i++) { |
| event.se_type = event_types[i]; |
| if (usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_EVENT, &event, |
| sizeof(event)) < 0) { |
| RTC_LOG_ERRNO(LS_ERROR) |
| << debug_name_ << "->ConfigureSctpSocket(): " |
| |
| << "Failed to set SCTP_EVENT type: " << event.se_type; |
| return false; |
| } |
| } |
| return true; |
| } |
| |
| void SctpTransport::CloseSctpSocket() { |
| RTC_DCHECK_RUN_ON(network_thread_); |
| if (sock_) { |
| // We assume that SO_LINGER option is set to close the association when |
| // close is called. This means that any pending packets in usrsctp will be |
| // discarded instead of being sent. |
| usrsctp_close(sock_); |
| sock_ = nullptr; |
| usrsctp_deregister_address(this); |
| UsrSctpWrapper::DecrementUsrSctpUsageCount(); |
| ready_to_send_data_ = false; |
| } |
| } |
| |
| bool SctpTransport::SendQueuedStreamResets() { |
| RTC_DCHECK_RUN_ON(network_thread_); |
| |
| // Figure out how many streams need to be reset. We need to do this so we can |
| // allocate the right amount of memory for the sctp_reset_streams structure. |
| size_t num_streams = std::count_if( |
| stream_status_by_sid_.begin(), stream_status_by_sid_.end(), |
| [](const std::map<uint32_t, StreamStatus>::value_type& stream) { |
| return stream.second.need_outgoing_reset(); |
| }); |
| if (num_streams == 0) { |
| // Nothing to reset. |
| return true; |
| } |
| |
| RTC_LOG(LS_VERBOSE) << "SendQueuedStreamResets[" << debug_name_ |
| << "]: Resetting " << num_streams << " outgoing streams."; |
| |
| const size_t num_bytes = |
| sizeof(struct sctp_reset_streams) + (num_streams * sizeof(uint16_t)); |
| std::vector<uint8_t> reset_stream_buf(num_bytes, 0); |
| struct sctp_reset_streams* resetp = |
| reinterpret_cast<sctp_reset_streams*>(&reset_stream_buf[0]); |
| resetp->srs_assoc_id = SCTP_ALL_ASSOC; |
| resetp->srs_flags = SCTP_STREAM_RESET_OUTGOING; |
| resetp->srs_number_streams = rtc::checked_cast<uint16_t>(num_streams); |
| int result_idx = 0; |
| |
| for (const std::map<uint32_t, StreamStatus>::value_type& stream : |
| stream_status_by_sid_) { |
| if (!stream.second.need_outgoing_reset()) { |
| continue; |
| } |
| resetp->srs_stream_list[result_idx++] = stream.first; |
| } |
| |
| int ret = |
| usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_RESET_STREAMS, resetp, |
| rtc::checked_cast<socklen_t>(reset_stream_buf.size())); |
| if (ret < 0) { |
| // Note that usrsctp only lets us have one reset in progress at a time |
| // (even though multiple streams can be reset at once). If this happens, |
| // SendQueuedStreamResets will end up called after the current in-progress |
| // reset finishes, in OnStreamResetEvent. |
| RTC_LOG_ERRNO(LS_WARNING) << debug_name_ |
| << "->SendQueuedStreamResets(): " |
| "Failed to send a stream reset for " |
| << num_streams << " streams"; |
| return false; |
| } |
| |
| // Since the usrsctp call completed successfully, update our stream status |
| // map to note that we started the outgoing reset. |
| for (auto it = stream_status_by_sid_.begin(); |
| it != stream_status_by_sid_.end(); ++it) { |
| if (it->second.need_outgoing_reset()) { |
| it->second.outgoing_reset_initiated = true; |
| } |
| } |
| return true; |
| } |
| |
| void SctpTransport::SetReadyToSendData() { |
| RTC_DCHECK_RUN_ON(network_thread_); |
| if (!ready_to_send_data_) { |
| ready_to_send_data_ = true; |
| SignalReadyToSendData(); |
| } |
| } |
| |
| void SctpTransport::OnWritableState(rtc::PacketTransportInternal* transport) { |
| RTC_DCHECK_RUN_ON(network_thread_); |
| RTC_DCHECK_EQ(transport_, transport); |
| if (!was_ever_writable_ && transport->writable()) { |
| was_ever_writable_ = true; |
| if (started_) { |
| Connect(); |
| } |
| } |
| } |
| |
| // Called by network interface when a packet has been received. |
| void SctpTransport::OnPacketRead(rtc::PacketTransportInternal* transport, |
| const char* data, |
| size_t len, |
| const int64_t& /* packet_time_us */, |
| int flags) { |
| RTC_DCHECK_RUN_ON(network_thread_); |
| RTC_DCHECK_EQ(transport_, transport); |
| TRACE_EVENT0("webrtc", "SctpTransport::OnPacketRead"); |
| |
| if (flags & PF_SRTP_BYPASS) { |
| // We are only interested in SCTP packets. |
| return; |
| } |
| |
| RTC_LOG(LS_VERBOSE) << debug_name_ << "->OnPacketRead(...): " |
| << " length=" << len << ", started: " << started_; |
| // Only give receiving packets to usrsctp after if connected. This enables two |
| // peers to each make a connect call, but for them not to receive an INIT |
| // packet before they have called connect; least the last receiver of the INIT |
| // packet will have called connect, and a connection will be established. |
| if (sock_) { |
| // Pass received packet to SCTP stack. Once processed by usrsctp, the data |
| // will be will be given to the global OnSctpInboundData, and then, |
| // marshalled by the AsyncInvoker. |
| VerboseLogPacket(data, len, SCTP_DUMP_INBOUND); |
| usrsctp_conninput(this, data, len, 0); |
| } else { |
| // TODO(ldixon): Consider caching the packet for very slightly better |
| // reliability. |
| } |
| } |
| |
| void SctpTransport::OnSendThresholdCallback() { |
| RTC_DCHECK_RUN_ON(network_thread_); |
| SetReadyToSendData(); |
| } |
| |
| sockaddr_conn SctpTransport::GetSctpSockAddr(int port) { |
| sockaddr_conn sconn = {0}; |
| sconn.sconn_family = AF_CONN; |
| #ifdef HAVE_SCONN_LEN |
| sconn.sconn_len = sizeof(sockaddr_conn); |
| #endif |
| // Note: conversion from int to uint16_t happens here. |
| sconn.sconn_port = rtc::HostToNetwork16(port); |
| sconn.sconn_addr = this; |
| return sconn; |
| } |
| |
| void SctpTransport::OnPacketFromSctpToNetwork( |
| const rtc::CopyOnWriteBuffer& buffer) { |
| RTC_DCHECK_RUN_ON(network_thread_); |
| if (buffer.size() > (kSctpMtu)) { |
| RTC_LOG(LS_ERROR) << debug_name_ << "->OnPacketFromSctpToNetwork(...): " |
| << "SCTP seems to have made a packet that is bigger " |
| << "than its official MTU: " << buffer.size() |
| << " vs max of " << kSctpMtu; |
| } |
| TRACE_EVENT0("webrtc", "SctpTransport::OnPacketFromSctpToNetwork"); |
| |
| // Don't create noise by trying to send a packet when the DTLS transport isn't |
| // even writable. |
| if (!transport_ || !transport_->writable()) { |
| return; |
| } |
| |
| // Bon voyage. |
| transport_->SendPacket(buffer.data<char>(), buffer.size(), |
| rtc::PacketOptions(), PF_NORMAL); |
| } |
| |
| void SctpTransport::OnInboundPacketFromSctpToTransport( |
| const rtc::CopyOnWriteBuffer& buffer, |
| ReceiveDataParams params, |
| int flags) { |
| RTC_DCHECK_RUN_ON(network_thread_); |
| RTC_LOG(LS_VERBOSE) << debug_name_ |
| << "->OnInboundPacketFromSctpToTransport(...): " |
| << "Received SCTP data:" |
| << " sid=" << params.sid |
| << " notification: " << (flags & MSG_NOTIFICATION) |
| << " length=" << buffer.size(); |
| // Sending a packet with data == NULL (no data) is SCTPs "close the |
| // connection" message. This sets sock_ = NULL; |
| if (!buffer.size() || !buffer.data()) { |
| RTC_LOG(LS_INFO) << debug_name_ |
| << "->OnInboundPacketFromSctpToTransport(...): " |
| "No data, closing."; |
| return; |
| } |
| if (flags & MSG_NOTIFICATION) { |
| OnNotificationFromSctp(buffer); |
| } else { |
| OnDataFromSctpToTransport(params, buffer); |
| } |
| } |
| |
| void SctpTransport::OnDataFromSctpToTransport( |
| const ReceiveDataParams& params, |
| const rtc::CopyOnWriteBuffer& buffer) { |
| RTC_DCHECK_RUN_ON(network_thread_); |
| RTC_LOG(LS_VERBOSE) << debug_name_ << "->OnDataFromSctpToTransport(...): " |
| << "Posting with length: " << buffer.size() |
| << " on stream " << params.sid; |
| // Reports all received messages to upper layers, no matter whether the sid |
| // is known. |
| SignalDataReceived(params, buffer); |
| } |
| |
| void SctpTransport::OnNotificationFromSctp( |
| const rtc::CopyOnWriteBuffer& buffer) { |
| RTC_DCHECK_RUN_ON(network_thread_); |
| const sctp_notification& notification = |
| reinterpret_cast<const sctp_notification&>(*buffer.data()); |
| RTC_DCHECK(notification.sn_header.sn_length == buffer.size()); |
| |
| // TODO(ldixon): handle notifications appropriately. |
| switch (notification.sn_header.sn_type) { |
| case SCTP_ASSOC_CHANGE: |
| RTC_LOG(LS_VERBOSE) << "SCTP_ASSOC_CHANGE"; |
| OnNotificationAssocChange(notification.sn_assoc_change); |
| break; |
| case SCTP_REMOTE_ERROR: |
| RTC_LOG(LS_INFO) << "SCTP_REMOTE_ERROR"; |
| break; |
| case SCTP_SHUTDOWN_EVENT: |
| RTC_LOG(LS_INFO) << "SCTP_SHUTDOWN_EVENT"; |
| break; |
| case SCTP_ADAPTATION_INDICATION: |
| RTC_LOG(LS_INFO) << "SCTP_ADAPTATION_INDICATION"; |
| break; |
| case SCTP_PARTIAL_DELIVERY_EVENT: |
| RTC_LOG(LS_INFO) << "SCTP_PARTIAL_DELIVERY_EVENT"; |
| break; |
| case SCTP_AUTHENTICATION_EVENT: |
| RTC_LOG(LS_INFO) << "SCTP_AUTHENTICATION_EVENT"; |
| break; |
| case SCTP_SENDER_DRY_EVENT: |
| RTC_LOG(LS_VERBOSE) << "SCTP_SENDER_DRY_EVENT"; |
| SetReadyToSendData(); |
| break; |
| // TODO(ldixon): Unblock after congestion. |
| case SCTP_NOTIFICATIONS_STOPPED_EVENT: |
| RTC_LOG(LS_INFO) << "SCTP_NOTIFICATIONS_STOPPED_EVENT"; |
| break; |
| case SCTP_SEND_FAILED_EVENT: |
| RTC_LOG(LS_INFO) << "SCTP_SEND_FAILED_EVENT"; |
| break; |
| case SCTP_STREAM_RESET_EVENT: |
| OnStreamResetEvent(¬ification.sn_strreset_event); |
| break; |
| case SCTP_ASSOC_RESET_EVENT: |
| RTC_LOG(LS_INFO) << "SCTP_ASSOC_RESET_EVENT"; |
| break; |
| case SCTP_STREAM_CHANGE_EVENT: |
| RTC_LOG(LS_INFO) << "SCTP_STREAM_CHANGE_EVENT"; |
| // An acknowledgment we get after our stream resets have gone through, |
| // if they've failed. We log the message, but don't react -- we don't |
| // keep around the last-transmitted set of SSIDs we wanted to close for |
| // error recovery. It doesn't seem likely to occur, and if so, likely |
| // harmless within the lifetime of a single SCTP association. |
| break; |
| default: |
| RTC_LOG(LS_WARNING) << "Unknown SCTP event: " |
| << notification.sn_header.sn_type; |
| break; |
| } |
| } |
| |
| void SctpTransport::OnNotificationAssocChange(const sctp_assoc_change& change) { |
| RTC_DCHECK_RUN_ON(network_thread_); |
| switch (change.sac_state) { |
| case SCTP_COMM_UP: |
| RTC_LOG(LS_VERBOSE) << "Association change SCTP_COMM_UP"; |
| break; |
| case SCTP_COMM_LOST: |
| RTC_LOG(LS_INFO) << "Association change SCTP_COMM_LOST"; |
| break; |
| case SCTP_RESTART: |
| RTC_LOG(LS_INFO) << "Association change SCTP_RESTART"; |
| break; |
| case SCTP_SHUTDOWN_COMP: |
| RTC_LOG(LS_INFO) << "Association change SCTP_SHUTDOWN_COMP"; |
| break; |
| case SCTP_CANT_STR_ASSOC: |
| RTC_LOG(LS_INFO) << "Association change SCTP_CANT_STR_ASSOC"; |
| break; |
| default: |
| RTC_LOG(LS_INFO) << "Association change UNKNOWN"; |
| break; |
| } |
| } |
| |
| void SctpTransport::OnStreamResetEvent( |
| const struct sctp_stream_reset_event* evt) { |
| RTC_DCHECK_RUN_ON(network_thread_); |
| |
| // This callback indicates that a reset is complete for incoming and/or |
| // outgoing streams. The reset may have been initiated by us or the remote |
| // side. |
| const int num_sids = (evt->strreset_length - sizeof(*evt)) / |
| sizeof(evt->strreset_stream_list[0]); |
| |
| if (evt->strreset_flags & SCTP_STREAM_RESET_FAILED) { |
| // OK, just try sending any previously sent stream resets again. The stream |
| // IDs sent over when the RESET_FIALED flag is set seem to be garbage |
| // values. Ignore them. |
| for (std::map<uint32_t, StreamStatus>::value_type& stream : |
| stream_status_by_sid_) { |
| stream.second.outgoing_reset_initiated = false; |
| } |
| SendQueuedStreamResets(); |
| // TODO(deadbeef): If this happens, the entire SCTP association is in quite |
| // crippled state. The SCTP session should be dismantled, and the WebRTC |
| // connectivity errored because is clear that the distant party is not |
| // playing ball: malforms the transported data. |
| return; |
| } |
| |
| // Loop over the received events and properly update the StreamStatus map. |
| for (int i = 0; i < num_sids; i++) { |
| const uint32_t sid = evt->strreset_stream_list[i]; |
| auto it = stream_status_by_sid_.find(sid); |
| if (it == stream_status_by_sid_.end()) { |
| // This stream is unknown. Sometimes this can be from a |
| // RESET_FAILED-related retransmit. |
| RTC_LOG(LS_VERBOSE) << "SCTP_STREAM_RESET_EVENT(" << debug_name_ |
| << "): Unknown sid " << sid; |
| continue; |
| } |
| StreamStatus& status = it->second; |
| |
| if (evt->strreset_flags & SCTP_STREAM_RESET_INCOMING_SSN) { |
| RTC_LOG(LS_VERBOSE) << "SCTP_STREAM_RESET_INCOMING_SSN(" << debug_name_ |
| << "): sid " << sid; |
| status.incoming_reset_complete = true; |
| // If we receive an incoming stream reset and we haven't started the |
| // closing procedure ourselves, this means the remote side started the |
| // closing procedure; fire a signal so that the relevant data channel |
| // can change to "closing" (we still need to reset the outgoing stream |
| // before it changes to "closed"). |
| if (!status.closure_initiated) { |
| SignalClosingProcedureStartedRemotely(sid); |
| } |
| } |
| if (evt->strreset_flags & SCTP_STREAM_RESET_OUTGOING_SSN) { |
| RTC_LOG(LS_VERBOSE) << "SCTP_STREAM_RESET_OUTGOING_SSN(" << debug_name_ |
| << "): sid " << sid; |
| status.outgoing_reset_complete = true; |
| } |
| |
| // If this reset completes the closing procedure, remove the stream from |
| // our map so we can consider it closed, and fire a signal such that the |
| // relevant DataChannel will change its state to "closed" and its ID can be |
| // re-used. |
| if (status.reset_complete()) { |
| stream_status_by_sid_.erase(it); |
| SignalClosingProcedureComplete(sid); |
| } |
| } |
| |
| // Always try to send any queued resets because this call indicates that the |
| // last outgoing or incoming reset has made some progress. |
| SendQueuedStreamResets(); |
| } |
| |
| } // namespace cricket |
