| /* |
| * 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 <stdarg.h> |
| #include <stdio.h> |
| |
| #include <memory> |
| #include <sstream> |
| #include <vector> |
| |
| #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 constexpr size_t kSctpMtu = 1200; |
| |
| // The size of the SCTP association send buffer. 256kB, the usrsctp default. |
| static constexpr 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<SctpInboundPacket> InboundPacketMessage; |
| typedef rtc::ScopedMessageData<rtc::CopyOnWriteBuffer> 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<void*>(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<SctpDataMediaChannel*>(addr); |
| LOG(LS_VERBOSE) << "global OnSctpOutboundPacket():" |
| << "addr: " << addr << "; length: " << length |
| << "; tos: " << std::hex << static_cast<int>(tos) |
| << "; set_df: " << std::hex << static_cast<int>(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<uint8_t*>(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<SctpDataMediaChannel*>(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<SctpDataMediaChannel::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. |
| LOG(LS_ERROR) << "Received an unknown PPID " << ppid |
| << " on an SCTP packet. Dropping."; |
| } else { |
| SctpInboundPacket* packet = new SctpInboundPacket; |
| packet->buffer.SetData(reinterpret_cast<uint8_t*>(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. |
| usrsctp_sysctl_set_sctp_nr_outgoing_streams_default(kMaxSctpStreams); |
| } |
| |
| 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<struct sockaddr_conn*>(&addrs[0]); |
| SctpDataMediaChannel* channel = |
| reinterpret_cast<SctpDataMediaChannel*>(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; |
| } |
| |
| // 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<sockaddr *>(&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<sockaddr *>(&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; |
| } |
| // 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; |
| 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 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<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; |
| 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<char>(), 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 sid=" << ssrc << " because sid 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 sid=" << 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 sid=" << 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<const sctp_notification&>(*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<DataCodec>& 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<int>(value); |
| return true; |
| } |
| } |
| } |
| return false; |
| } |
| |
| bool SctpDataMediaChannel::SetSendCodecs(const std::vector<DataCodec>& codecs) { |
| return GetCodecIntParameter( |
| codecs, kGoogleSctpDataCodecId, kGoogleSctpDataCodecName, kCodecParamPort, |
| &remote_port_); |
| } |
| |
| bool SctpDataMediaChannel::SetRecvCodecs(const std::vector<DataCodec>& codecs) { |
| return GetCodecIntParameter( |
| codecs, kGoogleSctpDataCodecId, kGoogleSctpDataCodecName, kCodecParamPort, |
| &local_port_); |
| } |
| |
| void SctpDataMediaChannel::OnPacketFromSctpToNetwork( |
| rtc::CopyOnWriteBuffer* buffer) { |
| if (buffer->size() > (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; |
| } |
| 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<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_INCOMING | SCTP_STREAM_RESET_OUTGOING; |
| resetp->srs_number_streams = rtc::checked_cast<uint16_t>(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<socklen_t>(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<InboundPacketMessage> pdata( |
| static_cast<InboundPacketMessage*>(msg->pdata)); |
| OnInboundPacketFromSctpToChannel(pdata->data().get()); |
| break; |
| } |
| case MSG_SCTPOUTBOUNDPACKET: { |
| std::unique_ptr<OutboundPacketMessage> pdata( |
| static_cast<OutboundPacketMessage*>(msg->pdata)); |
| OnPacketFromSctpToNetwork(pdata->data().get()); |
| break; |
| } |
| } |
| } |
| } // namespace cricket |