blob: 545d18ac375701c6def7eac77998067b28cbf0cb [file] [log] [blame]
/*
* 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 "media/sctp/sctptransport.h"
#include <stdarg.h>
#include <stdio.h>
#include <memory>
#include <sstream>
#include "usrsctplib/usrsctp.h"
#include "media/base/codec.h"
#include "media/base/mediaconstants.h"
#include "media/base/streamparams.h"
#include "p2p/base/dtlstransportinternal.h" // For PF_NORMAL
#include "rtc_base/arraysize.h"
#include "rtc_base/copyonwritebuffer.h"
#include "rtc_base/criticalsection.h"
#include "rtc_base/helpers.h"
#include "rtc_base/logging.h"
#include "rtc_base/safe_conversions.h"
#include "rtc_base/thread_checker.h"
#include "rtc_base/trace_event.h"
namespace {
// The biggest SCTP packet. Starting from a 'safe' wire MTU value of 1280,
// take off 80 bytes for DTLS/TURN/TCP/IP overhead.
static constexpr size_t kSctpMtu = 1200;
// The size of the SCTP association send buffer. 256kB, the usrsctp default.
static constexpr int kSendBufferSize = 256 * 1024;
// Set the initial value of the static SCTP Data Engines reference count.
int g_usrsctp_usage_count = 0;
rtc::GlobalLockPod g_usrsctp_lock_;
// DataMessageType is used for the SCTP "Payload Protocol Identifier", as
// defined in http://tools.ietf.org/html/rfc4960#section-14.4
//
// For the list of IANA approved values see:
// http://www.iana.org/assignments/sctp-parameters/sctp-parameters.xml
// The value is not used by SCTP itself. It indicates the protocol running
// on top of SCTP.
enum PayloadProtocolIdentifier {
PPID_NONE = 0, // No protocol is specified.
// Matches the PPIDs in mozilla source and
// https://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-protocol Sec. 9
// They're not yet assigned by IANA.
PPID_CONTROL = 50,
PPID_BINARY_PARTIAL = 52,
PPID_BINARY_LAST = 53,
PPID_TEXT_PARTIAL = 54,
PPID_TEXT_LAST = 51
};
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);
RTC_LOG(LS_INFO) << "SCTP: " << s;
va_end(ap);
#endif
}
// Get the PPID to use for the terminating fragment of this type.
PayloadProtocolIdentifier GetPpid(cricket::DataMessageType type) {
switch (type) {
default:
case cricket::DMT_NONE:
return PPID_NONE;
case cricket::DMT_CONTROL:
return PPID_CONTROL;
case cricket::DMT_BINARY:
return PPID_BINARY_LAST;
case cricket::DMT_TEXT:
return PPID_TEXT_LAST;
}
}
bool GetDataMediaType(PayloadProtocolIdentifier ppid,
cricket::DataMessageType* dest) {
RTC_DCHECK(dest != NULL);
switch (ppid) {
case PPID_BINARY_PARTIAL:
case PPID_BINARY_LAST:
*dest = cricket::DMT_BINARY;
return true;
case PPID_TEXT_PARTIAL:
case PPID_TEXT_LAST:
*dest = cricket::DMT_TEXT;
return true;
case PPID_CONTROL:
*dest = cricket::DMT_CONTROL;
return true;
case PPID_NONE:
*dest = cricket::DMT_NONE;
return true;
default:
return false;
}
}
// Log the packet in text2pcap format, if log level is at LS_VERBOSE.
//
// In order to turn these logs into a pcap file you can use, first filter the
// "SCTP_PACKET" log lines:
//
// cat chrome_debug.log | grep SCTP_PACKET > filtered.log
//
// Then run through text2pcap:
//
// text2pcap -t "%H:%M:%S." -D -u 1024,1024 filtered.log filtered.pcap
//
// The value "1024" isn't important, we just need a port for the dummy UDP
// headers generated. Lastly, you should be able to open filtered.pcap in
// Wireshark, then right click a packet and "Decode As..." SCTP.
//
// Why do all this? Because SCTP goes over DTLS, which is encrypted. So just
// getting a normal packet capture won't help you, unless you have the DTLS
// keying material.
void VerboseLogPacket(const void* data, size_t length, int direction) {
if (RTC_LOG_CHECK_LEVEL(LS_VERBOSE) && length > 0) {
char* dump_buf;
// Some downstream project uses an older version of usrsctp that expects
// a non-const "void*" as first parameter when dumping the packet, so we
// need to cast the const away here to avoid a compiler error.
if ((dump_buf = usrsctp_dumppacket(const_cast<void*>(data), length,
direction)) != NULL) {
RTC_LOG(LS_VERBOSE) << dump_buf;
usrsctp_freedumpbuffer(dump_buf);
}
}
}
} // namespace
namespace cricket {
// Handles global init/deinit, and mapping from usrsctp callbacks to
// SctpTransport calls.
class SctpTransport::UsrSctpWrapper {
public:
static void InitializeUsrSctp() {
RTC_LOG(LS_INFO) << __FUNCTION__;
// First argument is udp_encapsulation_port, which is not releveant for our
// AF_CONN use of sctp.
usrsctp_init(0, &UsrSctpWrapper::OnSctpOutboundPacket, &DebugSctpPrintf);
// To turn on/off detailed SCTP debugging. You will also need to have the
// SCTP_DEBUG cpp defines flag.
// usrsctp_sysctl_set_sctp_debug_on(SCTP_DEBUG_ALL);
// TODO(ldixon): Consider turning this on/off.
usrsctp_sysctl_set_sctp_ecn_enable(0);
// This is harmless, but we should find out when the library default
// changes.
int send_size = usrsctp_sysctl_get_sctp_sendspace();
if (send_size != kSendBufferSize) {
RTC_LOG(LS_ERROR) << "Got different send size than expected: "
<< send_size;
}
// TODO(ldixon): Consider turning this on/off.
// This is not needed right now (we don't do dynamic address changes):
// If SCTP Auto-ASCONF is enabled, the peer is informed automatically
// when a new address is added or removed. This feature is enabled by
// default.
// usrsctp_sysctl_set_sctp_auto_asconf(0);
// TODO(ldixon): Consider turning this on/off.
// Add a blackhole sysctl. Setting it to 1 results in no ABORTs
// being sent in response to INITs, setting it to 2 results
// in no ABORTs being sent for received OOTB packets.
// This is similar to the TCP sysctl.
//
// See: http://lakerest.net/pipermail/sctp-coders/2012-January/009438.html
// See: http://svnweb.freebsd.org/base?view=revision&revision=229805
// usrsctp_sysctl_set_sctp_blackhole(2);
// Set the number of default outgoing streams. This is the number we'll
// send in the SCTP INIT message.
usrsctp_sysctl_set_sctp_nr_outgoing_streams_default(kMaxSctpStreams);
}
static void UninitializeUsrSctp() {
RTC_LOG(LS_INFO) << __FUNCTION__;
// usrsctp_finish() may fail if it's called too soon after the transports
// are
// closed. Wait and try again until it succeeds for up to 3 seconds.
for (size_t i = 0; i < 300; ++i) {
if (usrsctp_finish() == 0) {
return;
}
rtc::Thread::SleepMs(10);
}
RTC_LOG(LS_ERROR) << "Failed to shutdown usrsctp.";
}
static void IncrementUsrSctpUsageCount() {
rtc::GlobalLockScope lock(&g_usrsctp_lock_);
if (!g_usrsctp_usage_count) {
InitializeUsrSctp();
}
++g_usrsctp_usage_count;
}
static void DecrementUsrSctpUsageCount() {
rtc::GlobalLockScope lock(&g_usrsctp_lock_);
--g_usrsctp_usage_count;
if (!g_usrsctp_usage_count) {
UninitializeUsrSctp();
}
}
// This is the callback usrsctp uses when there's data to send on the network
// that has been wrapped appropriatly for the SCTP protocol.
static int OnSctpOutboundPacket(void* addr,
void* data,
size_t length,
uint8_t tos,
uint8_t set_df) {
SctpTransport* transport = static_cast<SctpTransport*>(addr);
RTC_LOG(LS_VERBOSE) << "global OnSctpOutboundPacket():"
<< "addr: " << addr << "; length: " << length
<< "; tos: " << 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.
RTC_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) {
RTC_LOG(LS_ERROR)
<< "SendThresholdCallback: Failed to get transport for socket "
<< sock;
return 0;
}
transport->OnSendThresholdCallback();
return 0;
}
};
SctpTransport::SctpTransport(rtc::Thread* network_thread,
rtc::PacketTransportInternal* 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(rtc::PacketTransportInternal* 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_) {
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 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) {
RTC_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()) {
RTC_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()) {
RTC_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()) {
RTC_LOG(LS_WARNING) << debug_name_ << "->ResetStream(" << sid << "): "
<< "stream not found.";
return false;
} else {
RTC_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_) {
RTC_LOG(LS_WARNING) << debug_name_ << "->SendData(...): "
<< "Not sending packet with sid=" << params.sid
<< " len=" << payload.size() << " before Start().";
return false;
}
if (params.type != DMT_CONTROL &&
open_streams_.find(params.sid) == open_streams_.end()) {
RTC_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;
RTC_LOG(LS_INFO) << debug_name_
<< "->SendData(...): EWOULDBLOCK returned";
} else {
RTC_LOG_ERRNO(LS_ERROR) << "ERROR:" << debug_name_ << "->SendData(...): "
<< " usrsctp_sendv: ";
}
return false;
}
if (result) {
// Only way out now is success.
*result = SDR_SUCCESS;
}
return true;
}
bool SctpTransport::ReadyToSendData() {
RTC_DCHECK_RUN_ON(network_thread_);
return ready_to_send_data_;
}
void SctpTransport::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_);
RTC_LOG(LS_VERBOSE) << debug_name_ << "->Connect().";
// If we already have a socket connection (which shouldn't ever happen), just
// return.
RTC_DCHECK(!sock_);
if (sock_) {
RTC_LOG(LS_ERROR) << debug_name_
<< "->Connect(): Ignored as socket "
"is already established.";
return true;
}
// If no socket (it was closed) try to start it again. This can happen when
// the socket we are connecting to closes, does an sctp shutdown handshake,
// or behaves unexpectedly causing us to perform a CloseSctpSocket.
if (!OpenSctpSocket()) {
return false;
}
// Note: conversion from int to uint16_t happens on assignment.
sockaddr_conn local_sconn = GetSctpSockAddr(local_port_);
if (usrsctp_bind(sock_, reinterpret_cast<sockaddr*>(&local_sconn),
sizeof(local_sconn)) < 0) {
RTC_LOG_ERRNO(LS_ERROR)
<< debug_name_ << "->Connect(): " << ("Failed usrsctp_bind");
CloseSctpSocket();
return false;
}
// Note: conversion from int to uint16_t happens on assignment.
sockaddr_conn remote_sconn = GetSctpSockAddr(remote_port_);
int connect_result = usrsctp_connect(
sock_, reinterpret_cast<sockaddr*>(&remote_sconn), sizeof(remote_sconn));
if (connect_result < 0 && errno != SCTP_EINPROGRESS) {
RTC_LOG_ERRNO(LS_ERROR) << debug_name_ << "->Connect(): "
<< "Failed usrsctp_connect. got errno=" << errno
<< ", but wanted " << SCTP_EINPROGRESS;
CloseSctpSocket();
return false;
}
// Set the MTU and disable MTU discovery.
// We can only do this after usrsctp_connect or it has no effect.
sctp_paddrparams params = {{0}};
memcpy(&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))) {
RTC_LOG_ERRNO(LS_ERROR) << debug_name_ << "->Connect(): "
<< "Failed to set SCTP_PEER_ADDR_PARAMS.";
}
// Since this is a fresh SCTP association, we'll always start out with empty
// queues, so "ReadyToSendData" should be true.
SetReadyToSendData();
return true;
}
bool SctpTransport::OpenSctpSocket() {
RTC_DCHECK_RUN_ON(network_thread_);
if (sock_) {
RTC_LOG(LS_WARNING) << debug_name_ << "->OpenSctpSocket(): "
<< "Ignoring attempt to re-create existing socket.";
return false;
}
UsrSctpWrapper::IncrementUsrSctpUsageCount();
// If kSendBufferSize isn't reflective of reality, we log an error, but we
// still have to do something reasonable here. Look up what the buffer's
// real size is and set our threshold to something reasonable.
static const int kSendThreshold = usrsctp_sysctl_get_sctp_sendspace() / 2;
sock_ = usrsctp_socket(
AF_CONN, SOCK_STREAM, IPPROTO_SCTP, &UsrSctpWrapper::OnSctpInboundPacket,
&UsrSctpWrapper::SendThresholdCallback, kSendThreshold, this);
if (!sock_) {
RTC_LOG_ERRNO(LS_ERROR) << debug_name_ << "->OpenSctpSocket(): "
<< "Failed to create SCTP socket.";
UsrSctpWrapper::DecrementUsrSctpUsageCount();
return false;
}
if (!ConfigureSctpSocket()) {
usrsctp_close(sock_);
sock_ = nullptr;
UsrSctpWrapper::DecrementUsrSctpUsageCount();
return false;
}
// Register this class as an address for usrsctp. This is used by SCTP to
// direct the packets received (by the created socket) to this class.
usrsctp_register_address(this);
return true;
}
bool SctpTransport::ConfigureSctpSocket() {
RTC_DCHECK_RUN_ON(network_thread_);
RTC_DCHECK(sock_);
// Make the socket non-blocking. Connect, close, shutdown etc will not block
// the thread waiting for the socket operation to complete.
if (usrsctp_set_non_blocking(sock_, 1) < 0) {
RTC_LOG_ERRNO(LS_ERROR) << debug_name_ << "->ConfigureSctpSocket(): "
<< "Failed to set SCTP to non blocking.";
return false;
}
// This ensures that the usrsctp close call deletes the association. This
// prevents usrsctp from calling OnSctpOutboundPacket with references to
// this class as the address.
linger linger_opt;
linger_opt.l_onoff = 1;
linger_opt.l_linger = 0;
if (usrsctp_setsockopt(sock_, SOL_SOCKET, SO_LINGER, &linger_opt,
sizeof(linger_opt))) {
RTC_LOG_ERRNO(LS_ERROR) << debug_name_ << "->ConfigureSctpSocket(): "
<< "Failed to set SO_LINGER.";
return false;
}
// Enable stream ID resets.
struct sctp_assoc_value stream_rst;
stream_rst.assoc_id = SCTP_ALL_ASSOC;
stream_rst.assoc_value = 1;
if (usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_ENABLE_STREAM_RESET,
&stream_rst, sizeof(stream_rst))) {
RTC_LOG_ERRNO(LS_ERROR) << debug_name_ << "->ConfigureSctpSocket(): "
<< "Failed to set SCTP_ENABLE_STREAM_RESET.";
return false;
}
// Nagle.
uint32_t nodelay = 1;
if (usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_NODELAY, &nodelay,
sizeof(nodelay))) {
RTC_LOG_ERRNO(LS_ERROR) << debug_name_ << "->ConfigureSctpSocket(): "
<< "Failed to set SCTP_NODELAY.";
return false;
}
// Subscribe to SCTP event notifications.
int event_types[] = {SCTP_ASSOC_CHANGE, SCTP_PEER_ADDR_CHANGE,
SCTP_SEND_FAILED_EVENT, SCTP_SENDER_DRY_EVENT,
SCTP_STREAM_RESET_EVENT};
struct sctp_event event = {0};
event.se_assoc_id = SCTP_ALL_ASSOC;
event.se_on = 1;
for (size_t i = 0; i < arraysize(event_types); i++) {
event.se_type = event_types[i];
if (usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_EVENT, &event,
sizeof(event)) < 0) {
RTC_LOG_ERRNO(LS_ERROR)
<< debug_name_ << "->ConfigureSctpSocket(): "
<< "Failed to set SCTP_EVENT type: " << event.se_type;
return false;
}
}
return true;
}
void SctpTransport::CloseSctpSocket() {
RTC_DCHECK_RUN_ON(network_thread_);
if (sock_) {
// We assume that SO_LINGER option is set to close the association when
// close is called. This means that any pending packets in usrsctp will be
// discarded instead of being sent.
usrsctp_close(sock_);
sock_ = nullptr;
usrsctp_deregister_address(this);
UsrSctpWrapper::DecrementUsrSctpUsageCount();
ready_to_send_data_ = false;
}
}
bool SctpTransport::SendQueuedStreamResets() {
RTC_DCHECK_RUN_ON(network_thread_);
if (!sent_reset_streams_.empty() || queued_reset_streams_.empty()) {
return true;
}
RTC_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) {
RTC_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::PacketTransportInternal* 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::PacketTransportInternal* 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");
if (flags & PF_SRTP_BYPASS) {
// We are only interested in SCTP packets.
return;
}
RTC_LOG(LS_VERBOSE) << debug_name_ << "->OnPacketRead(...): "
<< " length=" << len << ", started: " << started_;
// Only give receiving packets to usrsctp after if connected. This enables two
// peers to each make a connect call, but for them not to receive an INIT
// packet before they have called connect; least the last receiver of the INIT
// packet will have called connect, and a connection will be established.
if (sock_) {
// Pass received packet to SCTP stack. Once processed by usrsctp, the data
// will be will be given to the global OnSctpInboundData, and then,
// marshalled by the AsyncInvoker.
VerboseLogPacket(data, len, SCTP_DUMP_INBOUND);
usrsctp_conninput(this, data, len, 0);
} else {
// TODO(ldixon): Consider caching the packet for very slightly better
// reliability.
}
}
void SctpTransport::OnSendThresholdCallback() {
RTC_DCHECK_RUN_ON(network_thread_);
SetReadyToSendData();
}
sockaddr_conn SctpTransport::GetSctpSockAddr(int port) {
sockaddr_conn sconn = {0};
sconn.sconn_family = AF_CONN;
#ifdef HAVE_SCONN_LEN
sconn.sconn_len = sizeof(sockaddr_conn);
#endif
// Note: conversion from int to uint16_t happens here.
sconn.sconn_port = rtc::HostToNetwork16(port);
sconn.sconn_addr = this;
return sconn;
}
void SctpTransport::OnPacketFromSctpToNetwork(
const rtc::CopyOnWriteBuffer& buffer) {
RTC_DCHECK_RUN_ON(network_thread_);
if (buffer.size() > (kSctpMtu)) {
RTC_LOG(LS_ERROR) << debug_name_ << "->OnPacketFromSctpToNetwork(...): "
<< "SCTP seems to have made a packet that is bigger "
<< "than its official MTU: " << buffer.size()
<< " vs max of " << kSctpMtu;
}
TRACE_EVENT0("webrtc", "SctpTransport::OnPacketFromSctpToNetwork");
// Don't create noise by trying to send a packet when the DTLS 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_);
RTC_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()) {
RTC_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_);
RTC_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:
RTC_LOG(LS_VERBOSE) << "SCTP_ASSOC_CHANGE";
OnNotificationAssocChange(notification.sn_assoc_change);
break;
case SCTP_REMOTE_ERROR:
RTC_LOG(LS_INFO) << "SCTP_REMOTE_ERROR";
break;
case SCTP_SHUTDOWN_EVENT:
RTC_LOG(LS_INFO) << "SCTP_SHUTDOWN_EVENT";
break;
case SCTP_ADAPTATION_INDICATION:
RTC_LOG(LS_INFO) << "SCTP_ADAPTATION_INDICATION";
break;
case SCTP_PARTIAL_DELIVERY_EVENT:
RTC_LOG(LS_INFO) << "SCTP_PARTIAL_DELIVERY_EVENT";
break;
case SCTP_AUTHENTICATION_EVENT:
RTC_LOG(LS_INFO) << "SCTP_AUTHENTICATION_EVENT";
break;
case SCTP_SENDER_DRY_EVENT:
RTC_LOG(LS_VERBOSE) << "SCTP_SENDER_DRY_EVENT";
SetReadyToSendData();
break;
// TODO(ldixon): Unblock after congestion.
case SCTP_NOTIFICATIONS_STOPPED_EVENT:
RTC_LOG(LS_INFO) << "SCTP_NOTIFICATIONS_STOPPED_EVENT";
break;
case SCTP_SEND_FAILED_EVENT:
RTC_LOG(LS_INFO) << "SCTP_SEND_FAILED_EVENT";
break;
case SCTP_STREAM_RESET_EVENT:
OnStreamResetEvent(&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;
default:
RTC_LOG(LS_WARNING) << "Unknown SCTP event: "
<< notification.sn_header.sn_type;
break;
}
}
void SctpTransport::OnNotificationAssocChange(const sctp_assoc_change& change) {
RTC_DCHECK_RUN_ON(network_thread_);
switch (change.sac_state) {
case SCTP_COMM_UP:
RTC_LOG(LS_VERBOSE) << "Association change SCTP_COMM_UP";
break;
case SCTP_COMM_LOST:
RTC_LOG(LS_INFO) << "Association change SCTP_COMM_LOST";
break;
case SCTP_RESTART:
RTC_LOG(LS_INFO) << "Association change SCTP_RESTART";
break;
case SCTP_SHUTDOWN_COMP:
RTC_LOG(LS_INFO) << "Association change SCTP_SHUTDOWN_COMP";
break;
case SCTP_CANT_STR_ASSOC:
RTC_LOG(LS_INFO) << "Association change SCTP_CANT_STR_ASSOC";
break;
default:
RTC_LOG(LS_INFO) << "Association change UNKNOWN";
break;
}
}
void SctpTransport::OnStreamResetEvent(
const struct sctp_stream_reset_event* evt) {
RTC_DCHECK_RUN_ON(network_thread_);
// 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]);
RTC_LOG(LS_VERBOSE) << "SCTP_STREAM_RESET_EVENT(" << debug_name_
<< "): Flags = 0x" << std::hex << evt->strreset_flags
<< " (" << ListFlags(evt->strreset_flags) << ")";
RTC_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()) {
RTC_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.
RTC_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.
RTC_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.
RTC_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