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webrtc / src / 62ec0f6130856599d9da4b4ad9e99f8e38c93bb9 / . / media / sctp / usrsctp_transport.cc
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/*
* 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
};
// Successful return value from usrsctp callbacks. Is not actually used by
// usrsctp, but all example programs for usrsctp use 1 as their return value.
constexpr int kSctpSuccessReturn = 1;
constexpr int kSctpErrorReturn = 0;
} // namespace
#include <stdarg.h>
#include <stdio.h>
#include <usrsctp.h>
#include <memory>
#include <unordered_map>
#include <utility>
#include "absl/algorithm/container.h"
#include "absl/base/attributes.h"
#include "absl/types/optional.h"
#include "api/sequence_checker.h"
#include "media/base/codec.h"
#include "media/base/media_channel.h"
#include "media/base/media_constants.h"
#include "media/base/stream_params.h"
#include "media/sctp/usrsctp_transport.h"
#include "p2p/base/dtls_transport_internal.h" // For PF_NORMAL
#include "rtc_base/arraysize.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/helpers.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/string_utils.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/task_utils/to_queued_task.h"
#include "rtc_base/thread_annotations.h"
#include "rtc_base/trace_event.h"
namespace {
// The biggest SCTP packet. Starting from a 'safe' wire MTU value of 1280,
// take off 85 bytes for DTLS/TURN/TCP/IP and ciphertext overhead.
//
// Additionally, it's possible that TURN adds an additional 4 bytes of overhead
// after a channel has been established, so we subtract an additional 4 bytes.
//
// 1280 IPV6 MTU
// -40 IPV6 header
// -8 UDP
// -24 GCM Cipher
// -13 DTLS record header
// -4 TURN ChannelData
// = 1191 bytes.
static constexpr size_t kSctpMtu = 1191;
// Set the initial value of the static SCTP Data Engines reference count.
ABSL_CONST_INIT int g_usrsctp_usage_count = 0;
ABSL_CONST_INIT bool g_usrsctp_initialized_ = false;
ABSL_CONST_INIT webrtc::GlobalMutex g_usrsctp_lock_(absl::kConstInit);
ABSL_CONST_INIT char kZero[] = {'\0'};
// 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:
// https://tools.ietf.org/html/rfc8831 Sec. 8
// 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 {
PPID_NONE = 0, // No protocol is specified.
PPID_CONTROL = 50,
PPID_TEXT_LAST = 51,
PPID_BINARY_PARTIAL = 52, // Deprecated
PPID_BINARY_LAST = 53,
PPID_TEXT_PARTIAL = 54, // Deprecated
PPID_TEXT_EMPTY = 56,
PPID_BINARY_EMPTY = 57,
};
// Should only be modified by UsrSctpWrapper.
ABSL_CONST_INIT cricket::UsrsctpTransportMap* g_transport_map_ = nullptr;
// Helper that will call C's free automatically.
// TODO(b/181900299): Figure out why unique_ptr with a custom deleter is causing
// issues in a certain build environment.
class AutoFreedPointer {
public:
explicit AutoFreedPointer(void* ptr) : ptr_(ptr) {}
AutoFreedPointer(AutoFreedPointer&& o) : ptr_(o.ptr_) { o.ptr_ = nullptr; }
~AutoFreedPointer() { free(ptr_); }
void* get() const { return ptr_; }
private:
void* ptr_;
};
// Helper for logging SCTP messages.
#if defined(__GNUC__)
__attribute__((__format__(__printf__, 1, 2)))
#endif
void DebugSctpPrintf(const char* format, ...) {
#if RTC_DCHECK_IS_ON
char s[255];
va_list ap;
va_start(ap, format);
vsnprintf(s, sizeof(s), format, ap);
RTC_LOG(LS_INFO) << "SCTP: " << s;
va_end(ap);
#endif
}
// Get the PPID to use for the terminating fragment of this type.
uint32_t GetPpid(webrtc::DataMessageType type, size_t size) {
switch (type) {
case webrtc::DataMessageType::kControl:
return PPID_CONTROL;
case webrtc::DataMessageType::kBinary:
return size > 0 ? PPID_BINARY_LAST : PPID_BINARY_EMPTY;
case webrtc::DataMessageType::kText:
return size > 0 ? PPID_TEXT_LAST : PPID_TEXT_EMPTY;
}
}
bool GetDataMediaType(uint32_t ppid, webrtc::DataMessageType* dest) {
RTC_DCHECK(dest != NULL);
switch (ppid) {
case PPID_BINARY_PARTIAL:
case PPID_BINARY_LAST:
case PPID_BINARY_EMPTY:
*dest = webrtc::DataMessageType::kBinary;
return true;
case PPID_TEXT_PARTIAL:
case PPID_TEXT_LAST:
case PPID_TEXT_EMPTY:
*dest = webrtc::DataMessageType::kText;
return true;
case PPID_CONTROL:
*dest = webrtc::DataMessageType::kControl;
return true;
}
return false;
}
bool IsEmptyPPID(uint32_t ppid) {
return ppid == PPID_BINARY_EMPTY || ppid == PPID_TEXT_EMPTY;
}
// Log the packet in text2pcap format, if log level is at LS_VERBOSE.
//
// In order to turn these logs into a pcap file you can use, first filter the
// "SCTP_PACKET" log lines:
//
// cat chrome_debug.log | grep SCTP_PACKET > filtered.log
//
// Then run through text2pcap:
//
// text2pcap -n -l 248 -D -t '%H:%M:%S.' filtered.log filtered.pcapng
//
// Command flag information:
// -n: Outputs to a pcapng file, can specify inbound/outbound packets.
// -l: Specifies the link layer header type. 248 means SCTP. See:
// http://www.tcpdump.org/linktypes.html
// -D: Text before packet specifies if it is inbound or outbound.
// -t: Time format.
//
// Why do all this? Because SCTP goes over DTLS, which is encrypted. So just
// getting a normal packet capture won't help you, unless you have the DTLS
// keying material.
void VerboseLogPacket(const void* data, size_t length, int direction) {
if (RTC_LOG_CHECK_LEVEL(LS_VERBOSE) && length > 0) {
char* dump_buf;
// Some downstream project uses an older version of usrsctp that expects
// a non-const "void*" as first parameter when dumping the packet, so we
// need to cast the const away here to avoid a compiler error.
if ((dump_buf = usrsctp_dumppacket(const_cast<void*>(data), length,
direction)) != NULL) {
RTC_LOG(LS_VERBOSE) << dump_buf;
usrsctp_freedumpbuffer(dump_buf);
}
}
}
// Creates the sctp_sendv_spa struct used for setting flags in the
// sctp_sendv() call.
sctp_sendv_spa CreateSctpSendParams(int sid,
const webrtc::SendDataParams& params,
size_t size) {
struct sctp_sendv_spa spa = {0};
spa.sendv_flags |= SCTP_SEND_SNDINFO_VALID;
spa.sendv_sndinfo.snd_sid = sid;
spa.sendv_sndinfo.snd_ppid = rtc::HostToNetwork32(GetPpid(params.type, size));
// Explicitly marking the EOR flag turns the usrsctp_sendv call below into a
// non atomic operation. This means that the sctp lib might only accept the
// message partially. This is done in order to improve throughput, so that we
// don't have to wait for an empty buffer to send the max message length, for
// example.
spa.sendv_sndinfo.snd_flags |= SCTP_EOR;
if (!params.ordered) {
spa.sendv_sndinfo.snd_flags |= SCTP_UNORDERED;
}
if (params.max_rtx_count.has_value()) {
RTC_DCHECK(*params.max_rtx_count >= 0 &&
*params.max_rtx_count <= std::numeric_limits<uint16_t>::max());
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;
}
if (params.max_rtx_ms.has_value()) {
RTC_DCHECK(*params.max_rtx_ms >= 0 &&
*params.max_rtx_ms <= std::numeric_limits<uint16_t>::max());
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;
}
return spa;
}
} // namespace
namespace cricket {
// Maps SCTP transport ID to UsrsctpTransport object, necessary in send
// threshold callback and outgoing packet callback. It also provides a facility
// to safely post a task to an UsrsctpTransport's network thread from another
// thread.
class UsrsctpTransportMap {
public:
UsrsctpTransportMap() = default;
// Assigns a new unused ID to the following transport.
uintptr_t Register(cricket::UsrsctpTransport* transport) {
webrtc::MutexLock lock(&lock_);
// usrsctp_connect fails with a value of 0...
if (next_id_ == 0) {
++next_id_;
}
// In case we've wrapped around and need to find an empty spot from a
// removed transport. Assumes we'll never be full.
while (map_.find(next_id_) != map_.end()) {
++next_id_;
if (next_id_ == 0) {
++next_id_;
}
}
map_[next_id_] = transport;
return next_id_++;
}
// Returns true if found.
bool Deregister(uintptr_t id) {
webrtc::MutexLock lock(&lock_);
return map_.erase(id) > 0;
}
// Posts |action| to the network thread of the transport identified by |id|
// and returns true if found, all while holding a lock to protect against the
// transport being simultaneously deleted/deregistered, or returns false if
// not found.
template <typename F>
bool PostToTransportThread(uintptr_t id, F action) const {
webrtc::MutexLock lock(&lock_);
UsrsctpTransport* transport = RetrieveWhileHoldingLock(id);
if (!transport) {
return false;
}
transport->network_thread_->PostTask(ToQueuedTask(
transport->task_safety_,
[transport, action{std::move(action)}]() { action(transport); }));
return true;
}
private:
UsrsctpTransport* RetrieveWhileHoldingLock(uintptr_t id) const
RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_) {
auto it = map_.find(id);
if (it == map_.end()) {
return nullptr;
}
return it->second;
}
mutable webrtc::Mutex lock_;
uintptr_t next_id_ RTC_GUARDED_BY(lock_) = 0;
std::unordered_map<uintptr_t, UsrsctpTransport*> map_ RTC_GUARDED_BY(lock_);
};
// Handles global init/deinit, and mapping from usrsctp callbacks to
// UsrsctpTransport calls.
class UsrsctpTransport::UsrSctpWrapper {
public:
static void InitializeUsrSctp() {
RTC_LOG(LS_INFO) << __FUNCTION__;
// UninitializeUsrSctp tries to call usrsctp_finish in a loop for three
// seconds; if that failed and we were left in a still-initialized state, we
// don't want to call usrsctp_init again as that will result in undefined
// behavior.
if (g_usrsctp_initialized_) {
RTC_LOG(LS_WARNING) << "Not reinitializing usrsctp since last attempt at "
"usrsctp_finish failed.";
} else {
// First argument is udp_encapsulation_port, which is not releveant for
// our AF_CONN use of sctp.
usrsctp_init(0, &UsrSctpWrapper::OnSctpOutboundPacket, &DebugSctpPrintf);
g_usrsctp_initialized_ = true;
}
// To turn on/off detailed SCTP debugging. You will also need to have the
// SCTP_DEBUG cpp defines flag, which can be turned on in media/BUILD.gn.
// usrsctp_sysctl_set_sctp_debug_on(SCTP_DEBUG_ALL);
// TODO(ldixon): Consider turning this on/off.
usrsctp_sysctl_set_sctp_ecn_enable(0);
// WebRTC doesn't use these features, so disable them to reduce the
// potential attack surface.
usrsctp_sysctl_set_sctp_asconf_enable(0);
usrsctp_sysctl_set_sctp_auth_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 != kSctpSendBufferSize) {
RTC_LOG(LS_ERROR) << "Got different send size than expected: "
<< send_size;
}
// TODO(ldixon): Consider turning this on/off.
// This is not needed right now (we don't do dynamic address changes):
// If SCTP Auto-ASCONF is enabled, the peer is informed automatically
// when a new address is added or removed. This feature is enabled by
// default.
// usrsctp_sysctl_set_sctp_auto_asconf(0);
// TODO(ldixon): Consider turning this on/off.
// Add a blackhole sysctl. Setting it to 1 results in no ABORTs
// being sent in response to INITs, setting it to 2 results
// in no ABORTs being sent for received OOTB packets.
// This is similar to the TCP sysctl.
//
// See: http://lakerest.net/pipermail/sctp-coders/2012-January/009438.html
// See: http://svnweb.freebsd.org/base?view=revision&revision=229805
// usrsctp_sysctl_set_sctp_blackhole(2);
// Set the number of default outgoing streams. This is the number we'll
// send in the SCTP INIT message.
usrsctp_sysctl_set_sctp_nr_outgoing_streams_default(kMaxSctpStreams);
g_transport_map_ = new UsrsctpTransportMap();
}
static void UninitializeUsrSctp() {
RTC_LOG(LS_INFO) << __FUNCTION__;
// usrsctp_finish() may fail if it's called too soon after the transports
// are
// closed. Wait and try again until it succeeds for up to 3 seconds.
for (size_t i = 0; i < 300; ++i) {
if (usrsctp_finish() == 0) {
g_usrsctp_initialized_ = false;
delete g_transport_map_;
g_transport_map_ = nullptr;
return;
}
rtc::Thread::SleepMs(10);
}
delete g_transport_map_;
g_transport_map_ = nullptr;
RTC_LOG(LS_ERROR) << "Failed to shutdown usrsctp.";
}
static void IncrementUsrSctpUsageCount() {
webrtc::GlobalMutexLock lock(&g_usrsctp_lock_);
if (!g_usrsctp_usage_count) {
InitializeUsrSctp();
}
++g_usrsctp_usage_count;
}
static void DecrementUsrSctpUsageCount() {
webrtc::GlobalMutexLock 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) {
if (!g_transport_map_) {
RTC_LOG(LS_ERROR)
<< "OnSctpOutboundPacket called after usrsctp uninitialized?";
return EINVAL;
}
RTC_LOG(LS_VERBOSE) << "global OnSctpOutboundPacket():"
"addr: "
<< addr << "; length: " << length
<< "; tos: " << rtc::ToHex(tos)
<< "; set_df: " << rtc::ToHex(set_df);
VerboseLogPacket(data, length, SCTP_DUMP_OUTBOUND);
// Note: We have to copy the data; the caller will delete it.
rtc::CopyOnWriteBuffer buf(reinterpret_cast<uint8_t*>(data), length);
// PostsToTransportThread protects against the transport being
// simultaneously deregistered/deleted, since this callback may come from
// the SCTP timer thread and thus race with the network thread.
bool found = g_transport_map_->PostToTransportThread(
reinterpret_cast<uintptr_t>(addr), [buf](UsrsctpTransport* transport) {
transport->OnPacketFromSctpToNetwork(buf);
});
if (!found) {
RTC_LOG(LS_ERROR)
<< "OnSctpOutboundPacket: Failed to get transport for socket ID "
<< addr << "; possibly was already destroyed.";
return EINVAL;
}
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) {
AutoFreedPointer owned_data(data);
absl::optional<uintptr_t> id = GetTransportIdFromSocket(sock);
if (!id) {
RTC_LOG(LS_ERROR)
<< "OnSctpInboundPacket: Failed to get transport ID from socket "
<< sock;
return kSctpErrorReturn;
}
if (!g_transport_map_) {
RTC_LOG(LS_ERROR)
<< "OnSctpInboundPacket called after usrsctp uninitialized?";
return kSctpErrorReturn;
}
// PostsToTransportThread protects against the transport being
// simultaneously deregistered/deleted, since this callback may come from
// the SCTP timer thread and thus race with the network thread.
bool found = g_transport_map_->PostToTransportThread(
*id, [owned_data{std::move(owned_data)}, length, rcv,
flags](UsrsctpTransport* transport) {
transport->OnDataOrNotificationFromSctp(owned_data.get(), length, rcv,
flags);
});
if (!found) {
RTC_LOG(LS_ERROR)
<< "OnSctpInboundPacket: Failed to get transport for socket ID "
<< *id << "; possibly was already destroyed.";
return kSctpErrorReturn;
}
return kSctpSuccessReturn;
}
static absl::optional<uintptr_t> GetTransportIdFromSocket(
struct socket* sock) {
absl::optional<uintptr_t> ret;
struct sockaddr* addrs = nullptr;
int naddrs = usrsctp_getladdrs(sock, 0, &addrs);
if (naddrs <= 0 || addrs[0].sa_family != AF_CONN) {
return ret;
}
// usrsctp_getladdrs() returns the addresses bound to this socket, which
// contains the UsrsctpTransport id as sconn_addr. Read the id,
// 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
// id of the transport that created them, so [0] is as good as any other.
struct sockaddr_conn* sconn =
reinterpret_cast<struct sockaddr_conn*>(&addrs[0]);
ret = reinterpret_cast<uintptr_t>(sconn->sconn_addr);
usrsctp_freeladdrs(addrs);
return ret;
}
// TODO(crbug.com/webrtc/11899): This is a legacy callback signature, remove
// when usrsctp is updated.
static int SendThresholdCallback(struct socket* sock, uint32_t sb_free) {
// Fired on our I/O thread. UsrsctpTransport::OnPacketReceived() gets
// a packet containing acknowledgments, which goes into usrsctp_conninput,
// and then back here.
absl::optional<uintptr_t> id = GetTransportIdFromSocket(sock);
if (!id) {
RTC_LOG(LS_ERROR)
<< "SendThresholdCallback: Failed to get transport ID from socket "
<< sock;
return 0;
}
if (!g_transport_map_) {
RTC_LOG(LS_ERROR)
<< "SendThresholdCallback called after usrsctp uninitialized?";
return 0;
}
bool found = g_transport_map_->PostToTransportThread(
*id, [](UsrsctpTransport* transport) {
transport->OnSendThresholdCallback();
});
if (!found) {
RTC_LOG(LS_ERROR)
<< "SendThresholdCallback: Failed to get transport for socket ID "
<< *id << "; possibly was already destroyed.";
}
return 0;
}
static int SendThresholdCallback(struct socket* sock,
uint32_t sb_free,
void* ulp_info) {
// Fired on our I/O thread. UsrsctpTransport::OnPacketReceived() gets
// a packet containing acknowledgments, which goes into usrsctp_conninput,
// and then back here.
absl::optional<uintptr_t> id = GetTransportIdFromSocket(sock);
if (!id) {
RTC_LOG(LS_ERROR)
<< "SendThresholdCallback: Failed to get transport ID from socket "
<< sock;
return 0;
}
if (!g_transport_map_) {
RTC_LOG(LS_ERROR)
<< "SendThresholdCallback called after usrsctp uninitialized?";
return 0;
}
bool found = g_transport_map_->PostToTransportThread(
*id, [](UsrsctpTransport* transport) {
transport->OnSendThresholdCallback();
});
if (!found) {
RTC_LOG(LS_ERROR)
<< "SendThresholdCallback: Failed to get transport for socket ID "
<< *id << "; possibly was already destroyed.";
}
return 0;
}
};
UsrsctpTransport::UsrsctpTransport(rtc::Thread* network_thread,
rtc::PacketTransportInternal* transport)
: network_thread_(network_thread),
transport_(transport),
was_ever_writable_(transport ? transport->writable() : false) {
RTC_DCHECK(network_thread_);
RTC_DCHECK_RUN_ON(network_thread_);
ConnectTransportSignals();
}
UsrsctpTransport::~UsrsctpTransport() {
RTC_DCHECK_RUN_ON(network_thread_);
// Close abruptly; no reset procedure.
CloseSctpSocket();
// It's not strictly necessary to reset these fields to nullptr,
// but having these fields set to nullptr is a clear indication that
// object was destructed. There was a bug in usrsctp when it
// invoked OnSctpOutboundPacket callback for destructed UsrsctpTransport,
// which caused obscure SIGSEGV on access to these fields,
// having this fields set to nullptr will make it easier to understand
// that UsrsctpTransport was destructed and "use-after-free" bug happen.
// SIGSEGV error triggered on dereference these pointers will also
// be easier to understand due to 0x0 address. All of this assumes
// that ASAN is not enabled to detect "use-after-free", which is
// currently default configuration.
network_thread_ = nullptr;
transport_ = nullptr;
}
void UsrsctpTransport::SetDtlsTransport(
rtc::PacketTransportInternal* transport) {
RTC_DCHECK_RUN_ON(network_thread_);
DisconnectTransportSignals();
transport_ = transport;
ConnectTransportSignals();
if (!was_ever_writable_ && transport && transport->writable()) {
was_ever_writable_ = true;
// New transport is writable, now we can start the SCTP connection if Start
// was called already.
if (started_) {
RTC_DCHECK(!sock_);
Connect();
}
}
}
bool UsrsctpTransport::Start(int local_sctp_port,
int remote_sctp_port,
int max_message_size) {
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 (max_message_size > kSctpSendBufferSize) {
RTC_LOG(LS_ERROR) << "Max message size of " << max_message_size
<< " is larger than send bufffer size "
<< kSctpSendBufferSize;
return false;
}
if (max_message_size < 1) {
RTC_LOG(LS_ERROR) << "Max message size of " << max_message_size
<< " is too small";
return false;
}
// We allow changing max_message_size with a second Start() call,
// but not changing the port numbers.
max_message_size_ = max_message_size;
if (started_) {
if (local_sctp_port != local_port_ || remote_sctp_port != remote_port_) {
RTC_LOG(LS_ERROR)
<< "Can't change SCTP port after SCTP association formed.";
return false;
}
return true;
}
local_port_ = local_sctp_port;
remote_port_ = remote_sctp_port;
started_ = true;
RTC_DCHECK(!sock_);
// Only try to connect if the DTLS transport has been writable before
// (indicating that the DTLS handshake is complete).
if (was_ever_writable_) {
return Connect();
}
return true;
}
bool UsrsctpTransport::OpenStream(int sid) {
RTC_DCHECK_RUN_ON(network_thread_);
if (sid > kMaxSctpSid) {
RTC_LOG(LS_WARNING) << debug_name_
<< "->OpenStream(...): "
"Not adding data stream "
"with sid="
<< sid << " because sid is too high.";
return false;
}
auto it = stream_status_by_sid_.find(sid);
if (it == stream_status_by_sid_.end()) {
stream_status_by_sid_[sid] = StreamStatus();
return true;
}
if (it->second.is_open()) {
RTC_LOG(LS_WARNING) << debug_name_
<< "->OpenStream(...): "
"Not adding data stream "
"with sid="
<< sid << " because stream is already open.";
return false;
} else {
RTC_LOG(LS_WARNING) << debug_name_
<< "->OpenStream(...): "
"Not adding data stream "
" with sid="
<< sid << " because stream is still closing.";
return false;
}
}
bool UsrsctpTransport::ResetStream(int sid) {
RTC_DCHECK_RUN_ON(network_thread_);
auto it = stream_status_by_sid_.find(sid);
if (it == stream_status_by_sid_.end() || !it->second.is_open()) {
RTC_LOG(LS_WARNING) << debug_name_ << "->ResetStream(" << sid
<< "): stream not open.";
return false;
}
RTC_LOG(LS_VERBOSE) << debug_name_ << "->ResetStream(" << sid
<< "): "
"Queuing RE-CONFIG chunk.";
it->second.closure_initiated = true;
// Signal our stream-reset logic that it should try to send now, if it can.
SendQueuedStreamResets();
// The stream will actually get removed when we get the acknowledgment.
return true;
}
bool UsrsctpTransport::SendData(int sid,
const webrtc::SendDataParams& params,
const rtc::CopyOnWriteBuffer& payload,
SendDataResult* result) {
RTC_DCHECK_RUN_ON(network_thread_);
if (partial_outgoing_message_.has_value()) {
if (result) {
*result = SDR_BLOCK;
}
// Ready to send should get set only when SendData() call gets blocked.
ready_to_send_data_ = false;
return false;
}
// Do not queue data to send on a closing stream.
auto it = stream_status_by_sid_.find(sid);
if (it == stream_status_by_sid_.end() || !it->second.is_open()) {
RTC_LOG(LS_WARNING)
<< debug_name_
<< "->SendData(...): "
"Not sending data because sid is unknown or closing: "
<< sid;
if (result) {
*result = SDR_ERROR;
}
return false;
}
size_t payload_size = payload.size();
OutgoingMessage message(payload, sid, params);
SendDataResult send_message_result = SendMessageInternal(&message);
if (result) {
*result = send_message_result;
}
if (payload_size == message.size()) {
// Nothing was sent.
return false;
}
// If any data is sent, we accept the message. In the case that data was
// partially accepted by the sctp library, the remaining is buffered. This
// ensures the client does not resend the message.
RTC_DCHECK_LT(message.size(), payload_size);
if (message.size() > 0) {
RTC_DCHECK(!partial_outgoing_message_.has_value());
RTC_DLOG(LS_VERBOSE) << "Partially sent message. Buffering the remaining"
<< message.size() << "/" << payload_size << " bytes.";
partial_outgoing_message_.emplace(message);
}
return true;
}
SendDataResult UsrsctpTransport::SendMessageInternal(OutgoingMessage* message) {
RTC_DCHECK_RUN_ON(network_thread_);
if (!sock_) {
RTC_LOG(LS_WARNING) << debug_name_
<< "->SendMessageInternal(...): "
"Not sending packet with sid="
<< message->sid() << " len=" << message->size()
<< " before Start().";
return SDR_ERROR;
}
if (message->send_params().type != webrtc::DataMessageType::kControl) {
auto it = stream_status_by_sid_.find(message->sid());
if (it == stream_status_by_sid_.end()) {
RTC_LOG(LS_WARNING) << debug_name_
<< "->SendMessageInternal(...): "
"Not sending data because sid is unknown: "
<< message->sid();
return SDR_ERROR;
}
}
if (message->size() > static_cast<size_t>(max_message_size_)) {
RTC_LOG(LS_ERROR) << "Attempting to send message of size "
<< message->size() << " which is larger than limit "
<< max_message_size_;
return SDR_ERROR;
}
// Send data using SCTP.
sctp_sendv_spa spa = CreateSctpSendParams(
message->sid(), message->send_params(), message->size());
const void* data = message->data();
size_t data_length = message->size();
if (message->size() == 0) {
// Empty messages are replaced by a single NUL byte on the wire as SCTP
// doesn't support empty messages.
// The PPID carries the information that the payload needs to be ignored.
data = kZero;
data_length = 1;
}
// Note: this send call is not atomic because the EOR bit is set. This means
// that usrsctp can partially accept this message and it is our duty to buffer
// the rest.
ssize_t send_res = usrsctp_sendv(sock_, data, data_length, NULL, 0, &spa,
rtc::checked_cast<socklen_t>(sizeof(spa)),
SCTP_SENDV_SPA, 0);
if (send_res < 0) {
if (errno == SCTP_EWOULDBLOCK) {
ready_to_send_data_ = false;
RTC_LOG(LS_INFO) << debug_name_
<< "->SendMessageInternal(...): EWOULDBLOCK returned";
return SDR_BLOCK;
}
RTC_LOG_ERRNO(LS_ERROR) << "ERROR:" << debug_name_
<< "->SendMessageInternal(...): "
" usrsctp_sendv: ";
return SDR_ERROR;
}
size_t amount_sent = static_cast<size_t>(send_res);
RTC_DCHECK_LE(amount_sent, data_length);
if (message->size() != 0)
message->Advance(amount_sent);
// Only way out now is success.
return SDR_SUCCESS;
}
bool UsrsctpTransport::ReadyToSendData() {
RTC_DCHECK_RUN_ON(network_thread_);
return ready_to_send_data_;
}
void UsrsctpTransport::ConnectTransportSignals() {
RTC_DCHECK_RUN_ON(network_thread_);
if (!transport_) {
return;
}
transport_->SignalWritableState.connect(this,
&UsrsctpTransport::OnWritableState);
transport_->SignalReadPacket.connect(this, &UsrsctpTransport::OnPacketRead);
transport_->SignalClosed.connect(this, &UsrsctpTransport::OnClosed);
}
void UsrsctpTransport::DisconnectTransportSignals() {
RTC_DCHECK_RUN_ON(network_thread_);
if (!transport_) {
return;
}
transport_->SignalWritableState.disconnect(this);
transport_->SignalReadPacket.disconnect(this);
transport_->SignalClosed.disconnect(this);
}
bool UsrsctpTransport::Connect() {
RTC_DCHECK_RUN_ON(network_thread_);
RTC_LOG(LS_VERBOSE) << debug_name_ << "->Connect().";
// If we already have a socket connection (which shouldn't ever happen), just
// return.
RTC_DCHECK(!sock_);
if (sock_) {
RTC_LOG(LS_ERROR) << debug_name_
<< "->Connect(): Ignored as socket "
"is already established.";
return true;
}
// If no socket (it was closed) try to start it again. This can happen when
// the socket we are connecting to closes, does an sctp shutdown handshake,
// or behaves unexpectedly causing us to perform a CloseSctpSocket.
if (!OpenSctpSocket()) {
return false;
}
// Note: conversion from int to uint16_t happens on assignment.
sockaddr_conn local_sconn = GetSctpSockAddr(local_port_);
if (usrsctp_bind(sock_, reinterpret_cast<sockaddr*>(&local_sconn),
sizeof(local_sconn)) < 0) {
RTC_LOG_ERRNO(LS_ERROR)
<< debug_name_ << "->Connect(): " << ("Failed usrsctp_bind");
CloseSctpSocket();
return false;
}
// Note: conversion from int to uint16_t happens on assignment.
sockaddr_conn remote_sconn = GetSctpSockAddr(remote_port_);
int connect_result = usrsctp_connect(
sock_, reinterpret_cast<sockaddr*>(&remote_sconn), sizeof(remote_sconn));
if (connect_result < 0 && errno != SCTP_EINPROGRESS) {
RTC_LOG_ERRNO(LS_ERROR) << debug_name_
<< "->Connect(): "
"Failed usrsctp_connect. got errno="
<< errno << ", but wanted " << SCTP_EINPROGRESS;
CloseSctpSocket();
return false;
}
// Set the MTU and disable MTU discovery.
// We can only do this after usrsctp_connect or it has no effect.
sctp_paddrparams params = {};
memcpy(&params.spp_address, &remote_sconn, sizeof(remote_sconn));
params.spp_flags = SPP_PMTUD_DISABLE;
// The MTU value provided specifies the space available for chunks in the
// packet, so we subtract the SCTP header size.
params.spp_pathmtu = kSctpMtu - sizeof(struct sctp_common_header);
if (usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_PEER_ADDR_PARAMS, &params,
sizeof(params))) {
RTC_LOG_ERRNO(LS_ERROR) << debug_name_
<< "->Connect(): "
"Failed to set SCTP_PEER_ADDR_PARAMS.";
}
// Since this is a fresh SCTP association, we'll always start out with empty
// queues, so "ReadyToSendData" should be true.
SetReadyToSendData();
return true;
}
bool UsrsctpTransport::OpenSctpSocket() {
RTC_DCHECK_RUN_ON(network_thread_);
if (sock_) {
RTC_LOG(LS_WARNING) << debug_name_
<< "->OpenSctpSocket(): "
"Ignoring attempt to re-create existing socket.";
return false;
}
UsrSctpWrapper::IncrementUsrSctpUsageCount();
// If kSctpSendBufferSize 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.
// TODO(bugs.webrtc.org/11824): That was previously set to 50%, not 25%, but
// it was reduced to a recent usrsctp regression. Can return to 50% when the
// root cause is fixed.
static const int kSendThreshold = usrsctp_sysctl_get_sctp_sendspace() / 4;
sock_ = usrsctp_socket(
AF_CONN, SOCK_STREAM, IPPROTO_SCTP, &UsrSctpWrapper::OnSctpInboundPacket,
&UsrSctpWrapper::SendThresholdCallback, kSendThreshold, this);
if (!sock_) {
RTC_LOG_ERRNO(LS_ERROR) << debug_name_
<< "->OpenSctpSocket(): "
"Failed to create SCTP socket.";
UsrSctpWrapper::DecrementUsrSctpUsageCount();
return false;
}
if (!ConfigureSctpSocket()) {
usrsctp_close(sock_);
sock_ = nullptr;
UsrSctpWrapper::DecrementUsrSctpUsageCount();
return false;
}
id_ = g_transport_map_->Register(this);
// Register our id 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(reinterpret_cast<void*>(id_));
return true;
}
bool UsrsctpTransport::ConfigureSctpSocket() {
RTC_DCHECK_RUN_ON(network_thread_);
RTC_DCHECK(sock_);
// Make the socket non-blocking. Connect, close, shutdown etc will not block
// the thread waiting for the socket operation to complete.
if (usrsctp_set_non_blocking(sock_, 1) < 0) {
RTC_LOG_ERRNO(LS_ERROR) << debug_name_
<< "->ConfigureSctpSocket(): "
"Failed to set SCTP to non blocking.";
return false;
}
// This ensures that the usrsctp close call deletes the association. This
// prevents usrsctp from calling OnSctpOutboundPacket with references to
// this class as the address.
linger linger_opt;
linger_opt.l_onoff = 1;
linger_opt.l_linger = 0;
if (usrsctp_setsockopt(sock_, SOL_SOCKET, SO_LINGER, &linger_opt,
sizeof(linger_opt))) {
RTC_LOG_ERRNO(LS_ERROR) << debug_name_
<< "->ConfigureSctpSocket(): "
"Failed to set SO_LINGER.";
return false;
}
// Enable stream ID resets.
struct sctp_assoc_value stream_rst;
stream_rst.assoc_id = SCTP_ALL_ASSOC;
stream_rst.assoc_value = 1;
if (usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_ENABLE_STREAM_RESET,
&stream_rst, sizeof(stream_rst))) {
RTC_LOG_ERRNO(LS_ERROR) << debug_name_
<< "->ConfigureSctpSocket(): "
"Failed to set SCTP_ENABLE_STREAM_RESET.";
return false;
}
// Nagle.
uint32_t nodelay = 1;
if (usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_NODELAY, &nodelay,
sizeof(nodelay))) {
RTC_LOG_ERRNO(LS_ERROR) << debug_name_
<< "->ConfigureSctpSocket(): "
"Failed to set SCTP_NODELAY.";
return false;
}
// Explicit EOR.
uint32_t eor = 1;
if (usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_EXPLICIT_EOR, &eor,
sizeof(eor))) {
RTC_LOG_ERRNO(LS_ERROR) << debug_name_
<< "->ConfigureSctpSocket(): "
"Failed to set SCTP_EXPLICIT_EOR.";
return false;
}
// Subscribe to SCTP event notifications.
// TODO(crbug.com/1137936): Subscribe to SCTP_SEND_FAILED_EVENT once deadlock
// is fixed upstream, or we switch to the upcall API:
// https://github.com/sctplab/usrsctp/issues/537
int event_types[] = {SCTP_ASSOC_CHANGE, SCTP_PEER_ADDR_CHANGE,
SCTP_SENDER_DRY_EVENT, SCTP_STREAM_RESET_EVENT};
struct sctp_event event = {0};
event.se_assoc_id = SCTP_ALL_ASSOC;
event.se_on = 1;
for (size_t i = 0; i < arraysize(event_types); i++) {
event.se_type = event_types[i];
if (usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_EVENT, &event,
sizeof(event)) < 0) {
RTC_LOG_ERRNO(LS_ERROR) << debug_name_
<< "->ConfigureSctpSocket(): "
"Failed to set SCTP_EVENT type: "
<< event.se_type;
return false;
}
}
return true;
}
void UsrsctpTransport::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(reinterpret_cast<void*>(id_));
RTC_CHECK(g_transport_map_->Deregister(id_));
UsrSctpWrapper::DecrementUsrSctpUsageCount();
ready_to_send_data_ = false;
}
}
bool UsrsctpTransport::SendQueuedStreamResets() {
RTC_DCHECK_RUN_ON(network_thread_);
auto needs_reset =
[this](const std::map<uint32_t, StreamStatus>::value_type& stream) {
// Ignore streams with partial outgoing messages as they are required to
// be fully sent by the WebRTC spec
// https://w3c.github.io/webrtc-pc/#closing-procedure
return stream.second.need_outgoing_reset() &&
(!partial_outgoing_message_.has_value() ||
partial_outgoing_message_.value().sid() !=
static_cast<int>(stream.first));
};
// Figure out how many streams need to be reset. We need to do this so we can
// allocate the right amount of memory for the sctp_reset_streams structure.
size_t num_streams = absl::c_count_if(stream_status_by_sid_, needs_reset);
if (num_streams == 0) {
// Nothing to reset.
return true;
}
RTC_LOG(LS_VERBOSE) << "SendQueuedStreamResets[" << debug_name_
<< "]: Resetting " << num_streams << " outgoing streams.";
const size_t num_bytes =
sizeof(struct sctp_reset_streams) + (num_streams * sizeof(uint16_t));
std::vector<uint8_t> reset_stream_buf(num_bytes, 0);
struct sctp_reset_streams* resetp =
reinterpret_cast<sctp_reset_streams*>(&reset_stream_buf[0]);
resetp->srs_assoc_id = SCTP_ALL_ASSOC;
resetp->srs_flags = SCTP_STREAM_RESET_OUTGOING;
resetp->srs_number_streams = rtc::checked_cast<uint16_t>(num_streams);
int result_idx = 0;
for (const auto& stream : stream_status_by_sid_) {
if (needs_reset(stream)) {
resetp->srs_stream_list[result_idx++] = stream.first;
}
}
int ret =
usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_RESET_STREAMS, resetp,
rtc::checked_cast<socklen_t>(reset_stream_buf.size()));
if (ret < 0) {
// Note that usrsctp only lets us have one reset in progress at a time
// (even though multiple streams can be reset at once). If this happens,
// SendQueuedStreamResets will end up called after the current in-progress
// reset finishes, in OnStreamResetEvent.
RTC_LOG_ERRNO(LS_WARNING) << debug_name_
<< "->SendQueuedStreamResets(): "
"Failed to send a stream reset for "
<< num_streams << " streams";
return false;
}
// Since the usrsctp call completed successfully, update our stream status
// map to note that we started the outgoing reset.
for (auto it = stream_status_by_sid_.begin();
it != stream_status_by_sid_.end(); ++it) {
if (it->second.need_outgoing_reset()) {
it->second.outgoing_reset_initiated = true;
}
}
return true;
}
void UsrsctpTransport::SetReadyToSendData() {
RTC_DCHECK_RUN_ON(network_thread_);
if (!ready_to_send_data_) {
ready_to_send_data_ = true;
SignalReadyToSendData();
}
}
bool UsrsctpTransport::SendBufferedMessage() {
RTC_DCHECK_RUN_ON(network_thread_);
RTC_DCHECK(partial_outgoing_message_.has_value());
RTC_DLOG(LS_VERBOSE) << "Sending partially buffered message of size "
<< partial_outgoing_message_->size() << ".";
SendMessageInternal(&partial_outgoing_message_.value());
if (partial_outgoing_message_->size() > 0) {
// Still need to finish sending the message.
return false;
}
RTC_DCHECK_EQ(0u, partial_outgoing_message_->size());
int sid = partial_outgoing_message_->sid();
partial_outgoing_message_.reset();
// Send the queued stream reset if it was pending for this stream.
auto it = stream_status_by_sid_.find(sid);
if (it->second.need_outgoing_reset()) {
SendQueuedStreamResets();
}
return true;
}
void UsrsctpTransport::OnWritableState(
rtc::PacketTransportInternal* transport) {
RTC_DCHECK_RUN_ON(network_thread_);
RTC_DCHECK_EQ(transport_, transport);
if (!was_ever_writable_ && transport->writable()) {
was_ever_writable_ = true;
if (started_) {
Connect();
}
}
}
// Called by network interface when a packet has been received.
void UsrsctpTransport::OnPacketRead(rtc::PacketTransportInternal* transport,
const char* data,
size_t len,
const int64_t& /* packet_time_us */,
int flags) {
RTC_DCHECK_RUN_ON(network_thread_);
RTC_DCHECK_EQ(transport_, transport);
TRACE_EVENT0("webrtc", "UsrsctpTransport::OnPacketRead");
if (flags & PF_SRTP_BYPASS) {
// We are only interested in SCTP packets.
return;
}
RTC_LOG(LS_VERBOSE) << debug_name_
<< "->OnPacketRead(...): "
" length="
<< len << ", started: " << started_;
// Only give receiving packets to usrsctp after if connected. This enables two
// peers to each make a connect call, but for them not to receive an INIT
// packet before they have called connect; least the last receiver of the INIT
// packet will have called connect, and a connection will be established.
if (sock_) {
// Pass received packet to SCTP stack. Once processed by usrsctp, the data
// will be will be given to the global OnSctpInboundData, and then,
// marshalled by the AsyncInvoker.
VerboseLogPacket(data, len, SCTP_DUMP_INBOUND);
usrsctp_conninput(reinterpret_cast<void*>(id_), data, len, 0);
} else {
// TODO(ldixon): Consider caching the packet for very slightly better
// reliability.
}
}
void UsrsctpTransport::OnClosed(rtc::PacketTransportInternal* transport) {
SignalClosedAbruptly();
}
void UsrsctpTransport::OnSendThresholdCallback() {
RTC_DCHECK_RUN_ON(network_thread_);
if (partial_outgoing_message_.has_value()) {
if (!SendBufferedMessage()) {
// Did not finish sending the buffered message.
return;
}
}
SetReadyToSendData();
}
sockaddr_conn UsrsctpTransport::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 = reinterpret_cast<void*>(id_);
return sconn;
}
void UsrsctpTransport::OnPacketFromSctpToNetwork(
const rtc::CopyOnWriteBuffer& buffer) {
RTC_DCHECK_RUN_ON(network_thread_);
if (buffer.size() > (kSctpMtu)) {
RTC_LOG(LS_ERROR) << debug_name_
<< "->OnPacketFromSctpToNetwork(...): "
"SCTP seems to have made a packet that is bigger "
"than its official MTU: "
<< buffer.size() << " vs max of " << kSctpMtu;
}
TRACE_EVENT0("webrtc", "UsrsctpTransport::OnPacketFromSctpToNetwork");
// Don't create noise by trying to send a packet when the DTLS transport isn't
// even writable.
if (!transport_ || !transport_->writable()) {
return;
}
// Bon voyage.
transport_->SendPacket(buffer.data<char>(), buffer.size(),
rtc::PacketOptions(), PF_NORMAL);
}
void UsrsctpTransport::InjectDataOrNotificationFromSctpForTesting(
const void* data,
size_t length,
struct sctp_rcvinfo rcv,
int flags) {
OnDataOrNotificationFromSctp(data, length, rcv, flags);
}
void UsrsctpTransport::OnDataOrNotificationFromSctp(const void* data,
size_t length,
struct sctp_rcvinfo rcv,
int flags) {
RTC_DCHECK_RUN_ON(network_thread_);
// If data is NULL, the SCTP association has been closed.
if (!data) {
RTC_LOG(LS_INFO) << debug_name_
<< "->OnDataOrNotificationFromSctp(...): "
"No data; association closed.";
return;
}
// Handle notifications early.
// Note: Notifications are never split into chunks, so they can and should
// be handled early and entirely separate from the reassembly
// process.
if (flags & MSG_NOTIFICATION) {
RTC_LOG(LS_VERBOSE)
<< debug_name_
<< "->OnDataOrNotificationFromSctp(...): SCTP notification"
<< " length=" << length;
rtc::CopyOnWriteBuffer notification(reinterpret_cast<const uint8_t*>(data),
length);
OnNotificationFromSctp(notification);
return;
}
// Log data chunk
const uint32_t ppid = rtc::NetworkToHost32(rcv.rcv_ppid);
RTC_LOG(LS_VERBOSE) << debug_name_
<< "->OnDataOrNotificationFromSctp(...): SCTP data chunk"
<< " length=" << length << ", sid=" << rcv.rcv_sid
<< ", ppid=" << ppid << ", ssn=" << rcv.rcv_ssn
<< ", cum-tsn=" << rcv.rcv_cumtsn
<< ", eor=" << ((flags & MSG_EOR) ? "y" : "n");
// Validate payload protocol identifier
webrtc::DataMessageType type;
if (!GetDataMediaType(ppid, &type)) {
// Unexpected PPID, dropping
RTC_LOG(LS_ERROR) << "Received an unknown PPID " << ppid
<< " on an SCTP packet. Dropping.";
return;
}
// Expect only continuation messages belonging to the same SID. The SCTP
// stack is expected to ensure this as long as the User Message
// Interleaving extension (RFC 8260) is not explicitly enabled, so this
// merely acts as a safeguard.
if ((partial_incoming_message_.size() != 0) &&
(rcv.rcv_sid != partial_params_.sid)) {
RTC_LOG(LS_ERROR) << "Received a new SID without EOR in the previous"
<< " SCTP packet. Discarding the previous packet.";
partial_incoming_message_.Clear();
}
// Copy metadata of interest
ReceiveDataParams params;
params.type = type;
params.sid = rcv.rcv_sid;
// Note that the SSN is identical for each chunk of the same message.
// Furthermore, it is increased per stream and not on the whole
// association.
params.seq_num = rcv.rcv_ssn;
// Append the chunk's data to the message buffer unless we have a chunk with a
// PPID marking an empty message.
// See: https://tools.ietf.org/html/rfc8831#section-6.6
if (!IsEmptyPPID(ppid))
partial_incoming_message_.AppendData(reinterpret_cast<const uint8_t*>(data),
length);
partial_params_ = params;
partial_flags_ = flags;
// If the message is not yet complete...
if (!(flags & MSG_EOR)) {
if (partial_incoming_message_.size() < kSctpSendBufferSize) {
// We still have space in the buffer. Continue buffering chunks until
// the message is complete before handing it out.
return;
} else {
// The sender is exceeding the maximum message size that we announced.
// Spit out a warning but still hand out the partial message. Note that
// this behaviour is undesirable, see the discussion in issue 7774.
//
// TODO(lgrahl): Once sufficient time has passed and all supported
// browser versions obey the announced maximum message size, we should
// abort the SCTP association instead to prevent message integrity
// violation.
RTC_LOG(LS_ERROR) << "Handing out partial SCTP message.";
}
}
// Dispatch the complete message and reset the message buffer.
OnDataFromSctpToTransport(params, partial_incoming_message_);
partial_incoming_message_.Clear();
}
void UsrsctpTransport::OnDataFromSctpToTransport(
const ReceiveDataParams& params,
const rtc::CopyOnWriteBuffer& buffer) {
RTC_DCHECK_RUN_ON(network_thread_);
RTC_LOG(LS_VERBOSE) << debug_name_
<< "->OnDataFromSctpToTransport(...): "
"Posting with length: "
<< buffer.size() << " on stream " << params.sid;
// Reports all received messages to upper layers, no matter whether the sid
// is known.
SignalDataReceived(params, buffer);
}
void UsrsctpTransport::OnNotificationFromSctp(
const rtc::CopyOnWriteBuffer& buffer) {
RTC_DCHECK_RUN_ON(network_thread_);
if (buffer.size() < sizeof(sctp_notification::sn_header)) {
RTC_LOG(LS_ERROR) << "SCTP notification is shorter than header size: "
<< buffer.size();
return;
}
const sctp_notification& notification =
reinterpret_cast<const sctp_notification&>(*buffer.data());
if (buffer.size() != notification.sn_header.sn_length) {
RTC_LOG(LS_ERROR) << "SCTP notification length (" << buffer.size()
<< ") does not match sn_length field ("
<< notification.sn_header.sn_length << ").";
return;
}
// TODO(ldixon): handle notifications appropriately.
switch (notification.sn_header.sn_type) {
case SCTP_ASSOC_CHANGE:
RTC_LOG(LS_VERBOSE) << "SCTP_ASSOC_CHANGE";
if (buffer.size() < sizeof(notification.sn_assoc_change)) {
RTC_LOG(LS_ERROR)
<< "SCTP_ASSOC_CHANGE notification has less than required length: "
<< buffer.size();
return;
}
OnNotificationAssocChange(notification.sn_assoc_change);
break;
case SCTP_REMOTE_ERROR:
RTC_LOG(LS_INFO) << "SCTP_REMOTE_ERROR";
break;
case SCTP_SHUTDOWN_EVENT:
RTC_LOG(LS_INFO) << "SCTP_SHUTDOWN_EVENT";
break;
case SCTP_ADAPTATION_INDICATION:
RTC_LOG(LS_INFO) << "SCTP_ADAPTATION_INDICATION";
break;
case SCTP_PARTIAL_DELIVERY_EVENT:
RTC_LOG(LS_INFO) << "SCTP_PARTIAL_DELIVERY_EVENT";
break;
case SCTP_AUTHENTICATION_EVENT:
RTC_LOG(LS_INFO) << "SCTP_AUTHENTICATION_EVENT";
break;
case SCTP_SENDER_DRY_EVENT:
RTC_LOG(LS_VERBOSE) << "SCTP_SENDER_DRY_EVENT";
SetReadyToSendData();
break;
// TODO(ldixon): Unblock after congestion.
case SCTP_NOTIFICATIONS_STOPPED_EVENT:
RTC_LOG(LS_INFO) << "SCTP_NOTIFICATIONS_STOPPED_EVENT";
break;
case SCTP_SEND_FAILED_EVENT: {
if (buffer.size() < sizeof(notification.sn_send_failed_event)) {
RTC_LOG(LS_ERROR) << "SCTP_SEND_FAILED_EVENT notification has less "
"than required length: "
<< buffer.size();
return;
}
const struct sctp_send_failed_event& ssfe =
notification.sn_send_failed_event;
RTC_LOG(LS_WARNING) << "SCTP_SEND_FAILED_EVENT: message with"
" PPID = "
<< rtc::NetworkToHost32(ssfe.ssfe_info.snd_ppid)
<< " SID = " << ssfe.ssfe_info.snd_sid
<< " flags = " << rtc::ToHex(ssfe.ssfe_info.snd_flags)
<< " failed to sent due to error = "
<< rtc::ToHex(ssfe.ssfe_error);
break;
}
case SCTP_STREAM_RESET_EVENT:
if (buffer.size() < sizeof(notification.sn_strreset_event)) {
RTC_LOG(LS_ERROR) << "SCTP_STREAM_RESET_EVENT notification has less "
"than required length: "
<< buffer.size();
return;
}
OnStreamResetEvent(&notification.sn_strreset_event);
break;
case SCTP_ASSOC_RESET_EVENT:
RTC_LOG(LS_INFO) << "SCTP_ASSOC_RESET_EVENT";
break;
case SCTP_STREAM_CHANGE_EVENT:
RTC_LOG(LS_INFO) << "SCTP_STREAM_CHANGE_EVENT";
// An acknowledgment we get after our stream resets have gone through,
// if they've failed. We log the message, but don't react -- we don't
// keep around the last-transmitted set of SSIDs we wanted to close for
// error recovery. It doesn't seem likely to occur, and if so, likely
// harmless within the lifetime of a single SCTP association.
break;
case SCTP_PEER_ADDR_CHANGE:
RTC_LOG(LS_INFO) << "SCTP_PEER_ADDR_CHANGE";
break;
default:
RTC_LOG(LS_WARNING) << "Unknown SCTP event: "
<< notification.sn_header.sn_type;
break;
}
}
void UsrsctpTransport::OnNotificationAssocChange(
const sctp_assoc_change& change) {
RTC_DCHECK_RUN_ON(network_thread_);
switch (change.sac_state) {
case SCTP_COMM_UP:
RTC_LOG(LS_VERBOSE) << "Association change SCTP_COMM_UP, stream # is "
<< change.sac_outbound_streams << " outbound, "
<< change.sac_inbound_streams << " inbound.";
max_outbound_streams_ = change.sac_outbound_streams;
max_inbound_streams_ = change.sac_inbound_streams;
SignalAssociationChangeCommunicationUp();
// In case someone tried to close a stream before communication
// came up, send any queued resets.
SendQueuedStreamResets();
break;
case SCTP_COMM_LOST:
RTC_LOG(LS_INFO) << "Association change SCTP_COMM_LOST";
break;
case SCTP_RESTART:
RTC_LOG(LS_INFO) << "Association change SCTP_RESTART";
break;
case SCTP_SHUTDOWN_COMP:
RTC_LOG(LS_INFO) << "Association change SCTP_SHUTDOWN_COMP";
break;
case SCTP_CANT_STR_ASSOC:
RTC_LOG(LS_INFO) << "Association change SCTP_CANT_STR_ASSOC";
break;
default:
RTC_LOG(LS_INFO) << "Association change UNKNOWN";
break;
}
}
void UsrsctpTransport::OnStreamResetEvent(
const struct sctp_stream_reset_event* evt) {
RTC_DCHECK_RUN_ON(network_thread_);
// This callback indicates that a reset is complete for incoming and/or
// outgoing streams. The reset may have been initiated by us or the remote
// side.
const int num_sids = (evt->strreset_length - sizeof(*evt)) /
sizeof(evt->strreset_stream_list[0]);
if (evt->strreset_flags & SCTP_STREAM_RESET_FAILED) {
// OK, just try sending any previously sent stream resets again. The stream
// IDs sent over when the RESET_FIALED flag is set seem to be garbage
// values. Ignore them.
for (std::map<uint32_t, StreamStatus>::value_type& stream :
stream_status_by_sid_) {
stream.second.outgoing_reset_initiated = false;
}
SendQueuedStreamResets();
// TODO(deadbeef): If this happens, the entire SCTP association is in quite
// crippled state. The SCTP session should be dismantled, and the WebRTC
// connectivity errored because is clear that the distant party is not
// playing ball: malforms the transported data.
return;
}
// Loop over the received events and properly update the StreamStatus map.
for (int i = 0; i < num_sids; i++) {
const uint32_t sid = evt->strreset_stream_list[i];
auto it = stream_status_by_sid_.find(sid);
if (it == stream_status_by_sid_.end()) {
// This stream is unknown. Sometimes this can be from a
// RESET_FAILED-related retransmit.
RTC_LOG(LS_VERBOSE) << "SCTP_STREAM_RESET_EVENT(" << debug_name_
<< "): Unknown sid " << sid;
continue;
}
StreamStatus& status = it->second;
if (evt->strreset_flags & SCTP_STREAM_RESET_INCOMING_SSN) {
RTC_LOG(LS_VERBOSE) << "SCTP_STREAM_RESET_INCOMING_SSN(" << debug_name_
<< "): sid " << sid;
status.incoming_reset_complete = true;
// If we receive an incoming stream reset and we haven't started the
// closing procedure ourselves, this means the remote side started the
// closing procedure; fire a signal so that the relevant data channel
// can change to "closing" (we still need to reset the outgoing stream
// before it changes to "closed").
if (!status.closure_initiated) {
SignalClosingProcedureStartedRemotely(sid);
}
}
if (evt->strreset_flags & SCTP_STREAM_RESET_OUTGOING_SSN) {
RTC_LOG(LS_VERBOSE) << "SCTP_STREAM_RESET_OUTGOING_SSN(" << debug_name_
<< "): sid " << sid;
status.outgoing_reset_complete = true;
}
// If this reset completes the closing procedure, remove the stream from
// our map so we can consider it closed, and fire a signal such that the
// relevant DataChannel will change its state to "closed" and its ID can be
// re-used.
if (status.reset_complete()) {
stream_status_by_sid_.erase(it);
SignalClosingProcedureComplete(sid);
}
}
// Always try to send any queued resets because this call indicates that the
// last outgoing or incoming reset has made some progress.
SendQueuedStreamResets();
}
} // namespace cricket