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Unified Diff: webrtc/media/sctp/sctptransport.cc

Issue 2614813003: Revert of Separating SCTP code from BaseChannel/MediaChannel. (Closed)
Patch Set: Also reverting https://codereview.webrtc.org/2612963002 Created 3 years, 11 months ago
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Index: webrtc/media/sctp/sctptransport.cc
diff --git a/webrtc/media/sctp/sctptransport.cc b/webrtc/media/sctp/sctptransport.cc
deleted file mode 100644
index b95cf8a4baf444bc3f14aee79e47e9a9e2983d0a..0000000000000000000000000000000000000000
--- a/webrtc/media/sctp/sctptransport.cc
+++ /dev/null
@@ -1,1090 +0,0 @@
-/*
- * 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
-};
-}
-
-#include "webrtc/media/sctp/sctptransport.h"
-
-#include <stdarg.h>
-#include <stdio.h>
-
-#include <memory>
-#include <sstream>
-
-#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/base/thread_checker.h"
-#include "webrtc/base/trace_event.h"
-#include "webrtc/media/base/codec.h"
-#include "webrtc/media/base/mediaconstants.h"
-#include "webrtc/media/base/rtputils.h" // For IsRtpPacket
-#include "webrtc/media/base/streamparams.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 = 262144;
-
-// 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
-};
-
-typedef std::set<uint32_t> 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();
-}
-
-// Helper for logging SCTP messages.
-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);
- 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.
-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);
- }
- }
-}
-
-} // namespace
-
-namespace cricket {
-
-// Handles global init/deinit, and mapping from usrsctp callbacks to
-// SctpTransport calls.
-class SctpTransport::UsrSctpWrapper {
- public:
- static 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, &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) {
- 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() {
- 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);
- }
- 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);
- 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.
- 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.
- LOG(LS_ERROR) << "Received an unknown PPID " << ppid
- << " on an SCTP packet. Dropping.";
- } else {
- rtc::CopyOnWriteBuffer buffer;
- ReceiveDataParams params;
- buffer.SetData(reinterpret_cast<uint8_t*>(data), length);
- params.sid = rcv.rcv_sid;
- params.seq_num = rcv.rcv_ssn;
- params.timestamp = rcv.rcv_tsn;
- params.type = type;
- // 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::OnInboundPacketFromSctpToChannel, transport,
- buffer, params, flags));
- }
- free(data);
- 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) {
- LOG(LS_ERROR)
- << "SendThresholdCallback: Failed to get transport for socket "
- << sock;
- return 0;
- }
- transport->OnSendThresholdCallback();
- return 0;
- }
-};
-
-SctpTransport::SctpTransport(rtc::Thread* network_thread,
- TransportChannel* channel)
- : network_thread_(network_thread),
- transport_channel_(channel),
- was_ever_writable_(channel->writable()) {
- RTC_DCHECK(network_thread_);
- RTC_DCHECK(transport_channel_);
- RTC_DCHECK_RUN_ON(network_thread_);
- ConnectTransportChannelSignals();
-}
-
-SctpTransport::~SctpTransport() {
- // Close abruptly; no reset procedure.
- CloseSctpSocket();
-}
-
-void SctpTransport::SetTransportChannel(cricket::TransportChannel* channel) {
- RTC_DCHECK_RUN_ON(network_thread_);
- RTC_DCHECK(channel);
- DisconnectTransportChannelSignals();
- transport_channel_ = channel;
- ConnectTransportChannelSignals();
- if (!was_ever_writable_ && channel->writable()) {
- was_ever_writable_ = true;
- // New channel 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_) {
- 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 channel 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) {
- LOG(LS_WARNING) << debug_name_ << "->OpenStream(...): "
- << "Not adding data stream "
- << "with sid=" << sid << " because sid is too high.";
- return false;
- } else if (open_streams_.find(sid) != open_streams_.end()) {
- LOG(LS_WARNING) << debug_name_ << "->OpenStream(...): "
- << "Not adding data stream "
- << "with sid=" << sid << " because stream is already open.";
- return false;
- } else if (queued_reset_streams_.find(sid) != queued_reset_streams_.end() ||
- sent_reset_streams_.find(sid) != sent_reset_streams_.end()) {
- LOG(LS_WARNING) << debug_name_ << "->OpenStream(...): "
- << "Not adding data stream "
- << " with sid=" << sid
- << " because stream is still closing.";
- return false;
- }
-
- open_streams_.insert(sid);
- return true;
-}
-
-bool SctpTransport::ResetStream(int sid) {
- RTC_DCHECK_RUN_ON(network_thread_);
- StreamSet::iterator found = open_streams_.find(sid);
- if (found == open_streams_.end()) {
- LOG(LS_WARNING) << debug_name_ << "->ResetStream(" << sid << "): "
- << "stream not found.";
- return false;
- } else {
- LOG(LS_VERBOSE) << debug_name_ << "->ResetStream(" << sid << "): "
- << "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(sid);
-
- // 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_) {
- 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 &&
- open_streams_.find(params.sid) == open_streams_.end()) {
- LOG(LS_WARNING) << debug_name_ << "->SendData(...): "
- << "Not sending data because sid is unknown: "
- << 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));
-
- // 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;
- 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;
-}
-
-bool SctpTransport::ReadyToSendData() {
- RTC_DCHECK_RUN_ON(network_thread_);
- return ready_to_send_data_;
-}
-
-void SctpTransport::ConnectTransportChannelSignals() {
- RTC_DCHECK_RUN_ON(network_thread_);
- transport_channel_->SignalWritableState.connect(
- this, &SctpTransport::OnWritableState);
- transport_channel_->SignalReadPacket.connect(this,
- &SctpTransport::OnPacketRead);
-}
-
-void SctpTransport::DisconnectTransportChannelSignals() {
- RTC_DCHECK_RUN_ON(network_thread_);
- transport_channel_->SignalWritableState.disconnect(this);
- transport_channel_->SignalReadPacket.disconnect(this);
-}
-
-bool SctpTransport::Connect() {
- RTC_DCHECK_RUN_ON(network_thread_);
- 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_) {
- 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) {
- 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_ << "->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(&params.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, &params,
- sizeof(params))) {
- 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_) {
- 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_) {
- 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) {
- 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))) {
- 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))) {
- 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))) {
- LOG_ERRNO(LS_ERROR) << debug_name_ << "->ConfigureSctpSocket(): "
- << "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_ << "->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_);
- 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_ << "->SendQueuedStreamResets(): "
- "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 SctpTransport::SetReadyToSendData() {
- RTC_DCHECK_RUN_ON(network_thread_);
- if (!ready_to_send_data_) {
- ready_to_send_data_ = true;
- SignalReadyToSendData();
- }
-}
-
-void SctpTransport::OnWritableState(rtc::PacketTransportInterface* transport) {
- RTC_DCHECK_RUN_ON(network_thread_);
- RTC_DCHECK_EQ(transport_channel_, 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::PacketTransportInterface* transport,
- const char* data,
- size_t len,
- const rtc::PacketTime& packet_time,
- int flags) {
- RTC_DCHECK_RUN_ON(network_thread_);
- RTC_DCHECK_EQ(transport_channel_, transport);
- TRACE_EVENT0("webrtc", "SctpTransport::OnPacketRead");
-
- // TODO(pthatcher): Do this in a more robust way by checking for
- // SCTP or DTLS.
- if (IsRtpPacket(data, len)) {
- return;
- }
-
- 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)) {
- 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 channel isn't
- // even writable.
- if (!transport_channel_->writable()) {
- return;
- }
-
- // Bon voyage.
- transport_channel_->SendPacket(buffer.data<char>(), buffer.size(),
- rtc::PacketOptions(), PF_NORMAL);
-}
-
-void SctpTransport::OnInboundPacketFromSctpToChannel(
- const rtc::CopyOnWriteBuffer& buffer,
- ReceiveDataParams params,
- int flags) {
- RTC_DCHECK_RUN_ON(network_thread_);
- LOG(LS_VERBOSE) << debug_name_ << "->OnInboundPacketFromSctpToChannel(...): "
- << "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()) {
- LOG(LS_INFO) << debug_name_ << "->OnInboundPacketFromSctpToChannel(...): "
- "No data, closing.";
- return;
- }
- if (flags & MSG_NOTIFICATION) {
- OnNotificationFromSctp(buffer);
- } else {
- OnDataFromSctpToChannel(params, buffer);
- }
-}
-
-void SctpTransport::OnDataFromSctpToChannel(
- const ReceiveDataParams& params,
- const rtc::CopyOnWriteBuffer& buffer) {
- RTC_DCHECK_RUN_ON(network_thread_);
- LOG(LS_VERBOSE) << debug_name_ << "->OnDataFromSctpToChannel(...): "
- << "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:
- 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";
- SetReadyToSendData();
- 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(&notification.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 SctpTransport::OnNotificationAssocChange(const sctp_assoc_change& change) {
- RTC_DCHECK_RUN_ON(network_thread_);
- 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 SctpTransport::OnStreamResetEvent(
- const struct sctp_stream_reset_event* evt) {
- RTC_DCHECK_RUN_ON(network_thread_);
- // 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_sids = (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_sids)
- << "], 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_sids; 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();
-}
-
-} // namespace cricket
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