/* * 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 "webrtc/media/sctp/sctpdataengine.h" #include #include #include #include #include #include "usrsctplib/usrsctp.h" #include "webrtc/base/arraysize.h" #include "webrtc/base/copyonwritebuffer.h" #include "webrtc/base/criticalsection.h" #include "webrtc/base/helpers.h" #include "webrtc/base/logging.h" #include "webrtc/base/safe_conversions.h" #include "webrtc/media/base/codec.h" #include "webrtc/media/base/mediaconstants.h" #include "webrtc/media/base/streamparams.h" namespace cricket { // The biggest SCTP packet. Starting from a 'safe' wire MTU value of 1280, // take off 80 bytes for DTLS/TURN/TCP/IP overhead. static const size_t kSctpMtu = 1200; // The size of the SCTP association send buffer. 256kB, the usrsctp default. static const int kSendBufferSize = 262144; struct SctpInboundPacket { rtc::CopyOnWriteBuffer buffer; ReceiveDataParams params; // The |flags| parameter is used by SCTP to distinguish notification packets // from other types of packets. int flags; }; namespace { // Set the initial value of the static SCTP Data Engines reference count. int g_usrsctp_usage_count = 0; rtc::GlobalLockPod g_usrsctp_lock_; typedef SctpDataMediaChannel::StreamSet StreamSet; // Returns a comma-separated, human-readable list of the stream IDs in 's' std::string ListStreams(const StreamSet& s) { std::stringstream result; bool first = true; for (StreamSet::const_iterator it = s.begin(); it != s.end(); ++it) { if (!first) { result << ", " << *it; } else { result << *it; first = false; } } return result.str(); } // Returns a pipe-separated, human-readable list of the SCTP_STREAM_RESET // flags in 'flags' std::string ListFlags(int flags) { std::stringstream result; bool first = true; // Skip past the first 12 chars (strlen("SCTP_STREAM_")) #define MAKEFLAG(X) { X, #X + 12} struct flaginfo_t { int value; const char* name; } flaginfo[] = { MAKEFLAG(SCTP_STREAM_RESET_INCOMING_SSN), MAKEFLAG(SCTP_STREAM_RESET_OUTGOING_SSN), MAKEFLAG(SCTP_STREAM_RESET_DENIED), MAKEFLAG(SCTP_STREAM_RESET_FAILED), MAKEFLAG(SCTP_STREAM_CHANGE_DENIED) }; #undef MAKEFLAG for (uint32_t i = 0; i < arraysize(flaginfo); ++i) { if (flags & flaginfo[i].value) { if (!first) result << " | "; result << flaginfo[i].name; first = false; } } return result.str(); } // Returns a comma-separated, human-readable list of the integers in 'array'. // All 'num_elems' of them. std::string ListArray(const uint16_t* array, int num_elems) { std::stringstream result; for (int i = 0; i < num_elems; ++i) { if (i) { result << ", " << array[i]; } else { result << array[i]; } } return result.str(); } typedef rtc::ScopedMessageData InboundPacketMessage; typedef rtc::ScopedMessageData OutboundPacketMessage; enum { MSG_SCTPINBOUNDPACKET = 1, // MessageData is SctpInboundPacket MSG_SCTPOUTBOUNDPACKET = 2, // MessageData is rtc:Buffer }; // Helper for logging SCTP messages. void DebugSctpPrintf(const char* format, ...) { #if (!defined(NDEBUG) || defined(DCHECK_ALWAYS_ON)) char s[255]; va_list ap; va_start(ap, format); vsnprintf(s, sizeof(s), format, ap); LOG(LS_INFO) << "SCTP: " << s; va_end(ap); #endif } // Get the PPID to use for the terminating fragment of this type. SctpDataMediaChannel::PayloadProtocolIdentifier GetPpid(DataMessageType type) { switch (type) { default: case DMT_NONE: return SctpDataMediaChannel::PPID_NONE; case DMT_CONTROL: return SctpDataMediaChannel::PPID_CONTROL; case DMT_BINARY: return SctpDataMediaChannel::PPID_BINARY_LAST; case DMT_TEXT: return SctpDataMediaChannel::PPID_TEXT_LAST; }; } bool GetDataMediaType(SctpDataMediaChannel::PayloadProtocolIdentifier ppid, DataMessageType* dest) { ASSERT(dest != NULL); switch (ppid) { case SctpDataMediaChannel::PPID_BINARY_PARTIAL: case SctpDataMediaChannel::PPID_BINARY_LAST: *dest = DMT_BINARY; return true; case SctpDataMediaChannel::PPID_TEXT_PARTIAL: case SctpDataMediaChannel::PPID_TEXT_LAST: *dest = DMT_TEXT; return true; case SctpDataMediaChannel::PPID_CONTROL: *dest = DMT_CONTROL; return true; case SctpDataMediaChannel::PPID_NONE: *dest = DMT_NONE; return true; default: return false; } } // Log the packet in text2pcap format, if log level is at LS_VERBOSE. void VerboseLogPacket(const void* data, size_t length, int direction) { if (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(data), length, direction)) != NULL) { LOG(LS_VERBOSE) << dump_buf; usrsctp_freedumpbuffer(dump_buf); } } } // This is the callback usrsctp uses when there's data to send on the network // that has been wrapped appropriatly for the SCTP protocol. int OnSctpOutboundPacket(void* addr, void* data, size_t length, uint8_t tos, uint8_t set_df) { SctpDataMediaChannel* channel = static_cast(addr); LOG(LS_VERBOSE) << "global OnSctpOutboundPacket():" << "addr: " << addr << "; length: " << length << "; tos: " << std::hex << static_cast(tos) << "; set_df: " << std::hex << static_cast(set_df); VerboseLogPacket(data, length, SCTP_DUMP_OUTBOUND); // Note: We have to copy the data; the caller will delete it. auto* msg = new OutboundPacketMessage( new rtc::CopyOnWriteBuffer(reinterpret_cast(data), length)); channel->worker_thread()->Post(RTC_FROM_HERE, channel, MSG_SCTPOUTBOUNDPACKET, msg); 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'. int OnSctpInboundPacket(struct socket* sock, union sctp_sockstore addr, void* data, size_t length, struct sctp_rcvinfo rcv, int flags, void* ulp_info) { SctpDataMediaChannel* channel = static_cast(ulp_info); // Post data to the channel'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 SctpDataMediaChannel::PayloadProtocolIdentifier ppid = static_cast( 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. LOG(LS_ERROR) << "Received an unknown PPID " << ppid << " on an SCTP packet. Dropping."; } else { SctpInboundPacket* packet = new SctpInboundPacket; packet->buffer.SetData(reinterpret_cast(data), length); packet->params.ssrc = rcv.rcv_sid; packet->params.seq_num = rcv.rcv_ssn; packet->params.timestamp = rcv.rcv_tsn; packet->params.type = type; packet->flags = flags; // The ownership of |packet| transfers to |msg|. InboundPacketMessage* msg = new InboundPacketMessage(packet); channel->worker_thread()->Post(RTC_FROM_HERE, channel, MSG_SCTPINBOUNDPACKET, msg); } free(data); return 1; } void InitializeUsrSctp() { LOG(LS_INFO) << __FUNCTION__; // First argument is udp_encapsulation_port, which is not releveant for our // AF_CONN use of sctp. usrsctp_init(0, &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) { 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. The 'appropriate default' in the // second paragraph of // http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-05#section-6.2 // is kMaxSctpSid. usrsctp_sysctl_set_sctp_nr_outgoing_streams_default(kMaxSctpSid); } void UninitializeUsrSctp() { LOG(LS_INFO) << __FUNCTION__; // usrsctp_finish() may fail if it's called too soon after the channels 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); } LOG(LS_ERROR) << "Failed to shutdown usrsctp."; } void IncrementUsrSctpUsageCount() { rtc::GlobalLockScope lock(&g_usrsctp_lock_); if (!g_usrsctp_usage_count) { InitializeUsrSctp(); } ++g_usrsctp_usage_count; } void DecrementUsrSctpUsageCount() { rtc::GlobalLockScope lock(&g_usrsctp_lock_); --g_usrsctp_usage_count; if (!g_usrsctp_usage_count) { UninitializeUsrSctp(); } } DataCodec GetSctpDataCodec() { DataCodec codec(kGoogleSctpDataCodecId, kGoogleSctpDataCodecName); codec.SetParam(kCodecParamPort, kSctpDefaultPort); return codec; } } // namespace SctpDataEngine::SctpDataEngine() : codecs_(1, GetSctpDataCodec()) {} SctpDataEngine::~SctpDataEngine() {} // Called on the worker thread. DataMediaChannel* SctpDataEngine::CreateChannel( DataChannelType data_channel_type) { if (data_channel_type != DCT_SCTP) { return NULL; } return new SctpDataMediaChannel(rtc::Thread::Current()); } // static SctpDataMediaChannel* SctpDataMediaChannel::GetChannelFromSocket( 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 SctpDataMediaChannel* 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(&addrs[0]); SctpDataMediaChannel* channel = reinterpret_cast(sconn->sconn_addr); usrsctp_freeladdrs(addrs); return channel; } // static int SctpDataMediaChannel::SendThresholdCallback(struct socket* sock, uint32_t sb_free) { // Fired on our I/O thread. SctpDataMediaChannel::OnPacketReceived() gets // a packet containing acknowledgments, which goes into usrsctp_conninput, // and then back here. SctpDataMediaChannel* channel = GetChannelFromSocket(sock); if (!channel) { LOG(LS_ERROR) << "SendThresholdCallback: Failed to get channel for socket " << sock; return 0; } channel->OnSendThresholdCallback(); return 0; } SctpDataMediaChannel::SctpDataMediaChannel(rtc::Thread* thread) : worker_thread_(thread), local_port_(kSctpDefaultPort), remote_port_(kSctpDefaultPort), sock_(NULL), sending_(false), receiving_(false), debug_name_("SctpDataMediaChannel") { } SctpDataMediaChannel::~SctpDataMediaChannel() { CloseSctpSocket(); } void SctpDataMediaChannel::OnSendThresholdCallback() { RTC_DCHECK(rtc::Thread::Current() == worker_thread_); SignalReadyToSend(true); } sockaddr_conn SctpDataMediaChannel::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; } bool SctpDataMediaChannel::OpenSctpSocket() { if (sock_) { LOG(LS_VERBOSE) << debug_name_ << "->Ignoring attempt to re-create existing socket."; return false; } 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. const static int kSendThreshold = usrsctp_sysctl_get_sctp_sendspace() / 2; sock_ = usrsctp_socket( AF_CONN, SOCK_STREAM, IPPROTO_SCTP, OnSctpInboundPacket, &SctpDataMediaChannel::SendThresholdCallback, kSendThreshold, this); if (!sock_) { LOG_ERRNO(LS_ERROR) << debug_name_ << "Failed to create SCTP socket."; DecrementUsrSctpUsageCount(); return false; } // 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) { LOG_ERRNO(LS_ERROR) << debug_name_ << "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))) { LOG_ERRNO(LS_ERROR) << debug_name_ << "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))) { LOG_ERRNO(LS_ERROR) << debug_name_ << "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))) { LOG_ERRNO(LS_ERROR) << debug_name_ << "Failed to set SCTP_NODELAY."; return false; } // Disable MTU discovery sctp_paddrparams params = {{0}}; params.spp_assoc_id = 0; params.spp_flags = SPP_PMTUD_DISABLE; params.spp_pathmtu = kSctpMtu; if (usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_PEER_ADDR_PARAMS, ¶ms, sizeof(params))) { LOG_ERRNO(LS_ERROR) << debug_name_ << "Failed to set SCTP_PEER_ADDR_PARAMS."; 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) { LOG_ERRNO(LS_ERROR) << debug_name_ << "Failed to set SCTP_EVENT type: " << event.se_type; 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); sending_ = true; return true; } void SctpDataMediaChannel::CloseSctpSocket() { sending_ = false; 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_ = NULL; usrsctp_deregister_address(this); DecrementUsrSctpUsageCount(); } } bool SctpDataMediaChannel::Connect() { LOG(LS_VERBOSE) << debug_name_ << "->Connect()."; // If we already have a socket connection, just return. if (sock_) { LOG(LS_WARNING) << 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 (!sock_ && !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(&local_sconn), sizeof(local_sconn)) < 0) { 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(&remote_sconn), sizeof(remote_sconn)); if (connect_result < 0 && errno != SCTP_EINPROGRESS) { LOG_ERRNO(LS_ERROR) << debug_name_ << "Failed usrsctp_connect. got errno=" << errno << ", but wanted " << SCTP_EINPROGRESS; CloseSctpSocket(); return false; } return true; } void SctpDataMediaChannel::Disconnect() { // TODO(ldixon): Consider calling |usrsctp_shutdown(sock_, ...)| to do a // shutdown handshake and remove the association. CloseSctpSocket(); } bool SctpDataMediaChannel::SetSend(bool send) { if (!sending_ && send) { return Connect(); } if (sending_ && !send) { Disconnect(); } return true; } bool SctpDataMediaChannel::SetReceive(bool receive) { receiving_ = receive; return true; } bool SctpDataMediaChannel::SetSendParameters(const DataSendParameters& params) { return SetSendCodecs(params.codecs); } bool SctpDataMediaChannel::SetRecvParameters(const DataRecvParameters& params) { return SetRecvCodecs(params.codecs); } bool SctpDataMediaChannel::AddSendStream(const StreamParams& stream) { return AddStream(stream); } bool SctpDataMediaChannel::RemoveSendStream(uint32_t ssrc) { return ResetStream(ssrc); } bool SctpDataMediaChannel::AddRecvStream(const StreamParams& stream) { // SCTP DataChannels are always bi-directional and calling AddSendStream will // enable both sending and receiving on the stream. So AddRecvStream is a // no-op. return true; } bool SctpDataMediaChannel::RemoveRecvStream(uint32_t ssrc) { // SCTP DataChannels are always bi-directional and calling RemoveSendStream // will disable both sending and receiving on the stream. So RemoveRecvStream // is a no-op. return true; } bool SctpDataMediaChannel::SendData( const SendDataParams& params, const rtc::CopyOnWriteBuffer& payload, SendDataResult* result) { if (result) { // Preset |result| to assume an error. If SendData succeeds, we'll // overwrite |*result| once more at the end. *result = SDR_ERROR; } if (!sending_) { LOG(LS_WARNING) << debug_name_ << "->SendData(...): " << "Not sending packet with ssrc=" << params.ssrc << " len=" << payload.size() << " before SetSend(true)."; return false; } if (params.type != DMT_CONTROL && open_streams_.find(params.ssrc) == open_streams_.end()) { LOG(LS_WARNING) << debug_name_ << "->SendData(...): " << "Not sending data because ssrc is unknown: " << params.ssrc; 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.ssrc; spa.sendv_sndinfo.snd_ppid = rtc::HostToNetwork32( GetPpid(params.type)); // 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(payload.size()), NULL, 0, &spa, rtc::checked_cast(sizeof(spa)), SCTP_SENDV_SPA, 0); if (send_res < 0) { if (errno == SCTP_EWOULDBLOCK) { *result = SDR_BLOCK; LOG(LS_INFO) << debug_name_ << "->SendData(...): EWOULDBLOCK returned"; } else { 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; } // Called by network interface when a packet has been received. void SctpDataMediaChannel::OnPacketReceived( rtc::CopyOnWriteBuffer* packet, const rtc::PacketTime& packet_time) { RTC_DCHECK(rtc::Thread::Current() == worker_thread_); LOG(LS_VERBOSE) << debug_name_ << "->OnPacketReceived(...): " << " length=" << packet->size() << ", sending: " << sending_; // 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 (sending_) { // 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 a Post and handled with OnMessage. VerboseLogPacket(packet->cdata(), packet->size(), SCTP_DUMP_INBOUND); usrsctp_conninput(this, packet->cdata(), packet->size(), 0); } else { // TODO(ldixon): Consider caching the packet for very slightly better // reliability. } } void SctpDataMediaChannel::OnInboundPacketFromSctpToChannel( SctpInboundPacket* packet) { LOG(LS_VERBOSE) << debug_name_ << "->OnInboundPacketFromSctpToChannel(...): " << "Received SCTP data:" << " ssrc=" << packet->params.ssrc << " notification: " << (packet->flags & MSG_NOTIFICATION) << " length=" << packet->buffer.size(); // Sending a packet with data == NULL (no data) is SCTPs "close the // connection" message. This sets sock_ = NULL; if (!packet->buffer.size() || !packet->buffer.data()) { LOG(LS_INFO) << debug_name_ << "->OnInboundPacketFromSctpToChannel(...): " "No data, closing."; return; } if (packet->flags & MSG_NOTIFICATION) { OnNotificationFromSctp(packet->buffer); } else { OnDataFromSctpToChannel(packet->params, packet->buffer); } } void SctpDataMediaChannel::OnDataFromSctpToChannel( const ReceiveDataParams& params, const rtc::CopyOnWriteBuffer& buffer) { if (receiving_) { LOG(LS_VERBOSE) << debug_name_ << "->OnDataFromSctpToChannel(...): " << "Posting with length: " << buffer.size() << " on stream " << params.ssrc; // Reports all received messages to upper layers, no matter whether the sid // is known. SignalDataReceived(params, buffer.data(), buffer.size()); } else { LOG(LS_WARNING) << debug_name_ << "->OnDataFromSctpToChannel(...): " << "Not receiving packet with sid=" << params.ssrc << " len=" << buffer.size() << " before SetReceive(true)."; } } bool SctpDataMediaChannel::AddStream(const StreamParams& stream) { if (!stream.has_ssrcs()) { return false; } const uint32_t ssrc = stream.first_ssrc(); if (ssrc >= kMaxSctpSid) { LOG(LS_WARNING) << debug_name_ << "->Add(Send|Recv)Stream(...): " << "Not adding data stream '" << stream.id << "' with ssrc=" << ssrc << " because stream ssrc is too high."; return false; } else if (open_streams_.find(ssrc) != open_streams_.end()) { LOG(LS_WARNING) << debug_name_ << "->Add(Send|Recv)Stream(...): " << "Not adding data stream '" << stream.id << "' with ssrc=" << ssrc << " because stream is already open."; return false; } else if (queued_reset_streams_.find(ssrc) != queued_reset_streams_.end() || sent_reset_streams_.find(ssrc) != sent_reset_streams_.end()) { LOG(LS_WARNING) << debug_name_ << "->Add(Send|Recv)Stream(...): " << "Not adding data stream '" << stream.id << "' with ssrc=" << ssrc << " because stream is still closing."; return false; } open_streams_.insert(ssrc); return true; } bool SctpDataMediaChannel::ResetStream(uint32_t ssrc) { // We typically get this called twice for the same stream, once each for // Send and Recv. StreamSet::iterator found = open_streams_.find(ssrc); if (found == open_streams_.end()) { LOG(LS_VERBOSE) << debug_name_ << "->ResetStream(" << ssrc << "): " << "stream not found."; return false; } else { LOG(LS_VERBOSE) << debug_name_ << "->ResetStream(" << ssrc << "): " << "Removing and queuing RE-CONFIG chunk."; open_streams_.erase(found); } // SCTP won't let you have more than one stream reset pending at a time, but // you can close multiple streams in a single reset. So, we keep an internal // queue of streams-to-reset, and send them as one reset message in // SendQueuedStreamResets(). queued_reset_streams_.insert(ssrc); // 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; } void SctpDataMediaChannel::OnNotificationFromSctp( const rtc::CopyOnWriteBuffer& buffer) { const sctp_notification& notification = reinterpret_cast(*buffer.data()); ASSERT(notification.sn_header.sn_length == buffer.size()); // TODO(ldixon): handle notifications appropriately. switch (notification.sn_header.sn_type) { case SCTP_ASSOC_CHANGE: LOG(LS_VERBOSE) << "SCTP_ASSOC_CHANGE"; OnNotificationAssocChange(notification.sn_assoc_change); break; case SCTP_REMOTE_ERROR: LOG(LS_INFO) << "SCTP_REMOTE_ERROR"; break; case SCTP_SHUTDOWN_EVENT: LOG(LS_INFO) << "SCTP_SHUTDOWN_EVENT"; break; case SCTP_ADAPTATION_INDICATION: LOG(LS_INFO) << "SCTP_ADAPTATION_INDICATION"; break; case SCTP_PARTIAL_DELIVERY_EVENT: LOG(LS_INFO) << "SCTP_PARTIAL_DELIVERY_EVENT"; break; case SCTP_AUTHENTICATION_EVENT: LOG(LS_INFO) << "SCTP_AUTHENTICATION_EVENT"; break; case SCTP_SENDER_DRY_EVENT: LOG(LS_VERBOSE) << "SCTP_SENDER_DRY_EVENT"; SignalReadyToSend(true); break; // TODO(ldixon): Unblock after congestion. case SCTP_NOTIFICATIONS_STOPPED_EVENT: LOG(LS_INFO) << "SCTP_NOTIFICATIONS_STOPPED_EVENT"; break; case SCTP_SEND_FAILED_EVENT: LOG(LS_INFO) << "SCTP_SEND_FAILED_EVENT"; break; case SCTP_STREAM_RESET_EVENT: OnStreamResetEvent(¬ification.sn_strreset_event); break; case SCTP_ASSOC_RESET_EVENT: LOG(LS_INFO) << "SCTP_ASSOC_RESET_EVENT"; break; case SCTP_STREAM_CHANGE_EVENT: 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: LOG(LS_WARNING) << "Unknown SCTP event: " << notification.sn_header.sn_type; break; } } void SctpDataMediaChannel::OnNotificationAssocChange( const sctp_assoc_change& change) { switch (change.sac_state) { case SCTP_COMM_UP: LOG(LS_VERBOSE) << "Association change SCTP_COMM_UP"; break; case SCTP_COMM_LOST: LOG(LS_INFO) << "Association change SCTP_COMM_LOST"; break; case SCTP_RESTART: LOG(LS_INFO) << "Association change SCTP_RESTART"; break; case SCTP_SHUTDOWN_COMP: LOG(LS_INFO) << "Association change SCTP_SHUTDOWN_COMP"; break; case SCTP_CANT_STR_ASSOC: LOG(LS_INFO) << "Association change SCTP_CANT_STR_ASSOC"; break; default: LOG(LS_INFO) << "Association change UNKNOWN"; break; } } void SctpDataMediaChannel::OnStreamResetEvent( const struct sctp_stream_reset_event* evt) { // A stream reset always involves two RE-CONFIG chunks for us -- we always // simultaneously reset a sid's sequence number in both directions. The // requesting side transmits a RE-CONFIG chunk and waits for the peer to send // one back. Both sides get this SCTP_STREAM_RESET_EVENT when they receive // RE-CONFIGs. const int num_ssrcs = (evt->strreset_length - sizeof(*evt)) / sizeof(evt->strreset_stream_list[0]); LOG(LS_VERBOSE) << "SCTP_STREAM_RESET_EVENT(" << debug_name_ << "): Flags = 0x" << std::hex << evt->strreset_flags << " (" << ListFlags(evt->strreset_flags) << ")"; LOG(LS_VERBOSE) << "Assoc = " << evt->strreset_assoc_id << ", Streams = [" << ListArray(evt->strreset_stream_list, num_ssrcs) << "], Open: [" << ListStreams(open_streams_) << "], Q'd: [" << ListStreams(queued_reset_streams_) << "], Sent: [" << ListStreams(sent_reset_streams_) << "]"; // If both sides try to reset some streams at the same time (even if they're // disjoint sets), we can get reset failures. if (evt->strreset_flags & SCTP_STREAM_RESET_FAILED) { // OK, just try again. The stream IDs sent over when the RESET_FAILED flag // is set seem to be garbage values. Ignore them. queued_reset_streams_.insert( sent_reset_streams_.begin(), sent_reset_streams_.end()); sent_reset_streams_.clear(); } else if (evt->strreset_flags & SCTP_STREAM_RESET_INCOMING_SSN) { // Each side gets an event for each direction of a stream. That is, // closing sid k will make each side receive INCOMING and OUTGOING reset // events for k. As per RFC6525, Section 5, paragraph 2, each side will // get an INCOMING event first. for (int i = 0; i < num_ssrcs; i++) { const int stream_id = evt->strreset_stream_list[i]; // See if this stream ID was closed by our peer or ourselves. StreamSet::iterator it = sent_reset_streams_.find(stream_id); // The reset was requested locally. if (it != sent_reset_streams_.end()) { LOG(LS_VERBOSE) << "SCTP_STREAM_RESET_EVENT(" << debug_name_ << "): local sid " << stream_id << " acknowledged."; sent_reset_streams_.erase(it); } else if ((it = open_streams_.find(stream_id)) != open_streams_.end()) { // The peer requested the reset. LOG(LS_VERBOSE) << "SCTP_STREAM_RESET_EVENT(" << debug_name_ << "): closing sid " << stream_id; open_streams_.erase(it); SignalStreamClosedRemotely(stream_id); } else if ((it = queued_reset_streams_.find(stream_id)) != queued_reset_streams_.end()) { // The peer requested the reset, but there was a local reset // queued. LOG(LS_VERBOSE) << "SCTP_STREAM_RESET_EVENT(" << debug_name_ << "): double-sided close for sid " << stream_id; // Both sides want the stream closed, and the peer got to send the // RE-CONFIG first. Treat it like the local Remove(Send|Recv)Stream // finished quickly. queued_reset_streams_.erase(it); } else { // This stream is unknown. Sometimes this can be from an // RESET_FAILED-related retransmit. LOG(LS_VERBOSE) << "SCTP_STREAM_RESET_EVENT(" << debug_name_ << "): Unknown sid " << stream_id; } } } // Always try to send the queued RESET because this call indicates that the // last local RESET or remote RESET has made some progress. SendQueuedStreamResets(); } // Puts the specified |param| from the codec identified by |id| into |dest| // and returns true. Or returns false if it wasn't there, leaving |dest| // untouched. static bool GetCodecIntParameter(const std::vector& codecs, int id, const std::string& name, const std::string& param, int* dest) { std::string value; Codec match_pattern; match_pattern.id = id; match_pattern.name = name; for (size_t i = 0; i < codecs.size(); ++i) { if (codecs[i].Matches(match_pattern)) { if (codecs[i].GetParam(param, &value)) { *dest = rtc::FromString(value); return true; } } } return false; } bool SctpDataMediaChannel::SetSendCodecs(const std::vector& codecs) { return GetCodecIntParameter( codecs, kGoogleSctpDataCodecId, kGoogleSctpDataCodecName, kCodecParamPort, &remote_port_); } bool SctpDataMediaChannel::SetRecvCodecs(const std::vector& codecs) { return GetCodecIntParameter( codecs, kGoogleSctpDataCodecId, kGoogleSctpDataCodecName, kCodecParamPort, &local_port_); } void SctpDataMediaChannel::OnPacketFromSctpToNetwork( rtc::CopyOnWriteBuffer* buffer) { // usrsctp seems to interpret the MTU we give it strangely -- it seems to // give us back packets bigger than that MTU, if only by a fixed amount. // This is that amount that we've observed. const int kSctpOverhead = 76; if (buffer->size() > (kSctpOverhead + kSctpMtu)) { 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 << " even after adding " << kSctpOverhead << " extra SCTP overhead"; } MediaChannel::SendPacket(buffer, rtc::PacketOptions()); } bool SctpDataMediaChannel::SendQueuedStreamResets() { if (!sent_reset_streams_.empty() || queued_reset_streams_.empty()) { return true; } LOG(LS_VERBOSE) << "SendQueuedStreamResets[" << debug_name_ << "]: Sending [" << ListStreams(queued_reset_streams_) << "], Open: [" << ListStreams(open_streams_) << "], Sent: [" << ListStreams(sent_reset_streams_) << "]"; const size_t num_streams = queued_reset_streams_.size(); const size_t num_bytes = sizeof(struct sctp_reset_streams) + (num_streams * sizeof(uint16_t)); std::vector reset_stream_buf(num_bytes, 0); struct sctp_reset_streams* resetp = reinterpret_cast( &reset_stream_buf[0]); resetp->srs_assoc_id = SCTP_ALL_ASSOC; resetp->srs_flags = SCTP_STREAM_RESET_INCOMING | SCTP_STREAM_RESET_OUTGOING; resetp->srs_number_streams = rtc::checked_cast(num_streams); int result_idx = 0; for (StreamSet::iterator it = queued_reset_streams_.begin(); it != queued_reset_streams_.end(); ++it) { resetp->srs_stream_list[result_idx++] = *it; } int ret = usrsctp_setsockopt( sock_, IPPROTO_SCTP, SCTP_RESET_STREAMS, resetp, rtc::checked_cast(reset_stream_buf.size())); if (ret < 0) { LOG_ERRNO(LS_ERROR) << debug_name_ << "Failed to send a stream reset for " << num_streams << " streams"; return false; } // sent_reset_streams_ is empty, and all the queued_reset_streams_ go into // it now. queued_reset_streams_.swap(sent_reset_streams_); return true; } void SctpDataMediaChannel::OnMessage(rtc::Message* msg) { switch (msg->message_id) { case MSG_SCTPINBOUNDPACKET: { std::unique_ptr pdata( static_cast(msg->pdata)); OnInboundPacketFromSctpToChannel(pdata->data().get()); break; } case MSG_SCTPOUTBOUNDPACKET: { std::unique_ptr pdata( static_cast(msg->pdata)); OnPacketFromSctpToNetwork(pdata->data().get()); break; } } } } // namespace cricket