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| 1 /* | |
| 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | |
| 3 * | |
| 4 * Use of this source code is governed by a BSD-style license | |
| 5 * that can be found in the LICENSE file in the root of the source | |
| 6 * tree. An additional intellectual property rights grant can be found | |
| 7 * in the file PATENTS. All contributing project authors may | |
| 8 * be found in the AUTHORS file in the root of the source tree. | |
| 9 */ | |
| 10 | |
| 11 #include <errno.h> | |
| 12 namespace { | |
| 13 // Some ERRNO values get re-#defined to WSA* equivalents in some talk/ | |
| 14 // headers. We save the original ones in an enum. | |
| 15 enum PreservedErrno { | |
| 16 SCTP_EINPROGRESS = EINPROGRESS, | |
| 17 SCTP_EWOULDBLOCK = EWOULDBLOCK | |
| 18 }; | |
| 19 } | |
| 20 | |
| 21 #include "webrtc/media/sctp/sctptransport.h" | |
| 22 | |
| 23 #include <stdarg.h> | |
| 24 #include <stdio.h> | |
| 25 | |
| 26 #include <memory> | |
| 27 #include <sstream> | |
| 28 | |
| 29 #include "usrsctplib/usrsctp.h" | |
| 30 #include "webrtc/base/arraysize.h" | |
| 31 #include "webrtc/base/copyonwritebuffer.h" | |
| 32 #include "webrtc/base/criticalsection.h" | |
| 33 #include "webrtc/base/helpers.h" | |
| 34 #include "webrtc/base/logging.h" | |
| 35 #include "webrtc/base/safe_conversions.h" | |
| 36 #include "webrtc/base/thread_checker.h" | |
| 37 #include "webrtc/base/trace_event.h" | |
| 38 #include "webrtc/media/base/codec.h" | |
| 39 #include "webrtc/media/base/mediaconstants.h" | |
| 40 #include "webrtc/media/base/rtputils.h" // For IsRtpPacket | |
| 41 #include "webrtc/media/base/streamparams.h" | |
| 42 | |
| 43 namespace { | |
| 44 | |
| 45 // The biggest SCTP packet. Starting from a 'safe' wire MTU value of 1280, | |
| 46 // take off 80 bytes for DTLS/TURN/TCP/IP overhead. | |
| 47 static constexpr size_t kSctpMtu = 1200; | |
| 48 | |
| 49 // The size of the SCTP association send buffer. 256kB, the usrsctp default. | |
| 50 static constexpr int kSendBufferSize = 262144; | |
| 51 | |
| 52 // Set the initial value of the static SCTP Data Engines reference count. | |
| 53 int g_usrsctp_usage_count = 0; | |
| 54 rtc::GlobalLockPod g_usrsctp_lock_; | |
| 55 | |
| 56 // DataMessageType is used for the SCTP "Payload Protocol Identifier", as | |
| 57 // defined in http://tools.ietf.org/html/rfc4960#section-14.4 | |
| 58 // | |
| 59 // For the list of IANA approved values see: | |
| 60 // http://www.iana.org/assignments/sctp-parameters/sctp-parameters.xml | |
| 61 // The value is not used by SCTP itself. It indicates the protocol running | |
| 62 // on top of SCTP. | |
| 63 enum PayloadProtocolIdentifier { | |
| 64 PPID_NONE = 0, // No protocol is specified. | |
| 65 // Matches the PPIDs in mozilla source and | |
| 66 // https://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-protocol Sec. 9 | |
| 67 // They're not yet assigned by IANA. | |
| 68 PPID_CONTROL = 50, | |
| 69 PPID_BINARY_PARTIAL = 52, | |
| 70 PPID_BINARY_LAST = 53, | |
| 71 PPID_TEXT_PARTIAL = 54, | |
| 72 PPID_TEXT_LAST = 51 | |
| 73 }; | |
| 74 | |
| 75 typedef std::set<uint32_t> StreamSet; | |
| 76 | |
| 77 // Returns a comma-separated, human-readable list of the stream IDs in 's' | |
| 78 std::string ListStreams(const StreamSet& s) { | |
| 79 std::stringstream result; | |
| 80 bool first = true; | |
| 81 for (StreamSet::const_iterator it = s.begin(); it != s.end(); ++it) { | |
| 82 if (!first) { | |
| 83 result << ", " << *it; | |
| 84 } else { | |
| 85 result << *it; | |
| 86 first = false; | |
| 87 } | |
| 88 } | |
| 89 return result.str(); | |
| 90 } | |
| 91 | |
| 92 // Returns a pipe-separated, human-readable list of the SCTP_STREAM_RESET | |
| 93 // flags in 'flags' | |
| 94 std::string ListFlags(int flags) { | |
| 95 std::stringstream result; | |
| 96 bool first = true; | |
| 97 // Skip past the first 12 chars (strlen("SCTP_STREAM_")) | |
| 98 #define MAKEFLAG(X) \ | |
| 99 { X, #X + 12 } | |
| 100 struct flaginfo_t { | |
| 101 int value; | |
| 102 const char* name; | |
| 103 } flaginfo[] = {MAKEFLAG(SCTP_STREAM_RESET_INCOMING_SSN), | |
| 104 MAKEFLAG(SCTP_STREAM_RESET_OUTGOING_SSN), | |
| 105 MAKEFLAG(SCTP_STREAM_RESET_DENIED), | |
| 106 MAKEFLAG(SCTP_STREAM_RESET_FAILED), | |
| 107 MAKEFLAG(SCTP_STREAM_CHANGE_DENIED)}; | |
| 108 #undef MAKEFLAG | |
| 109 for (uint32_t i = 0; i < arraysize(flaginfo); ++i) { | |
| 110 if (flags & flaginfo[i].value) { | |
| 111 if (!first) | |
| 112 result << " | "; | |
| 113 result << flaginfo[i].name; | |
| 114 first = false; | |
| 115 } | |
| 116 } | |
| 117 return result.str(); | |
| 118 } | |
| 119 | |
| 120 // Returns a comma-separated, human-readable list of the integers in 'array'. | |
| 121 // All 'num_elems' of them. | |
| 122 std::string ListArray(const uint16_t* array, int num_elems) { | |
| 123 std::stringstream result; | |
| 124 for (int i = 0; i < num_elems; ++i) { | |
| 125 if (i) { | |
| 126 result << ", " << array[i]; | |
| 127 } else { | |
| 128 result << array[i]; | |
| 129 } | |
| 130 } | |
| 131 return result.str(); | |
| 132 } | |
| 133 | |
| 134 // Helper for logging SCTP messages. | |
| 135 void DebugSctpPrintf(const char* format, ...) { | |
| 136 #if RTC_DCHECK_IS_ON | |
| 137 char s[255]; | |
| 138 va_list ap; | |
| 139 va_start(ap, format); | |
| 140 vsnprintf(s, sizeof(s), format, ap); | |
| 141 LOG(LS_INFO) << "SCTP: " << s; | |
| 142 va_end(ap); | |
| 143 #endif | |
| 144 } | |
| 145 | |
| 146 // Get the PPID to use for the terminating fragment of this type. | |
| 147 PayloadProtocolIdentifier GetPpid(cricket::DataMessageType type) { | |
| 148 switch (type) { | |
| 149 default: | |
| 150 case cricket::DMT_NONE: | |
| 151 return PPID_NONE; | |
| 152 case cricket::DMT_CONTROL: | |
| 153 return PPID_CONTROL; | |
| 154 case cricket::DMT_BINARY: | |
| 155 return PPID_BINARY_LAST; | |
| 156 case cricket::DMT_TEXT: | |
| 157 return PPID_TEXT_LAST; | |
| 158 } | |
| 159 } | |
| 160 | |
| 161 bool GetDataMediaType(PayloadProtocolIdentifier ppid, | |
| 162 cricket::DataMessageType* dest) { | |
| 163 RTC_DCHECK(dest != NULL); | |
| 164 switch (ppid) { | |
| 165 case PPID_BINARY_PARTIAL: | |
| 166 case PPID_BINARY_LAST: | |
| 167 *dest = cricket::DMT_BINARY; | |
| 168 return true; | |
| 169 | |
| 170 case PPID_TEXT_PARTIAL: | |
| 171 case PPID_TEXT_LAST: | |
| 172 *dest = cricket::DMT_TEXT; | |
| 173 return true; | |
| 174 | |
| 175 case PPID_CONTROL: | |
| 176 *dest = cricket::DMT_CONTROL; | |
| 177 return true; | |
| 178 | |
| 179 case PPID_NONE: | |
| 180 *dest = cricket::DMT_NONE; | |
| 181 return true; | |
| 182 | |
| 183 default: | |
| 184 return false; | |
| 185 } | |
| 186 } | |
| 187 | |
| 188 // Log the packet in text2pcap format, if log level is at LS_VERBOSE. | |
| 189 void VerboseLogPacket(const void* data, size_t length, int direction) { | |
| 190 if (LOG_CHECK_LEVEL(LS_VERBOSE) && length > 0) { | |
| 191 char* dump_buf; | |
| 192 // Some downstream project uses an older version of usrsctp that expects | |
| 193 // a non-const "void*" as first parameter when dumping the packet, so we | |
| 194 // need to cast the const away here to avoid a compiler error. | |
| 195 if ((dump_buf = usrsctp_dumppacket(const_cast<void*>(data), length, | |
| 196 direction)) != NULL) { | |
| 197 LOG(LS_VERBOSE) << dump_buf; | |
| 198 usrsctp_freedumpbuffer(dump_buf); | |
| 199 } | |
| 200 } | |
| 201 } | |
| 202 | |
| 203 } // namespace | |
| 204 | |
| 205 namespace cricket { | |
| 206 | |
| 207 // Handles global init/deinit, and mapping from usrsctp callbacks to | |
| 208 // SctpTransport calls. | |
| 209 class SctpTransport::UsrSctpWrapper { | |
| 210 public: | |
| 211 static void InitializeUsrSctp() { | |
| 212 LOG(LS_INFO) << __FUNCTION__; | |
| 213 // First argument is udp_encapsulation_port, which is not releveant for our | |
| 214 // AF_CONN use of sctp. | |
| 215 usrsctp_init(0, &UsrSctpWrapper::OnSctpOutboundPacket, &DebugSctpPrintf); | |
| 216 | |
| 217 // To turn on/off detailed SCTP debugging. You will also need to have the | |
| 218 // SCTP_DEBUG cpp defines flag. | |
| 219 // usrsctp_sysctl_set_sctp_debug_on(SCTP_DEBUG_ALL); | |
| 220 | |
| 221 // TODO(ldixon): Consider turning this on/off. | |
| 222 usrsctp_sysctl_set_sctp_ecn_enable(0); | |
| 223 | |
| 224 // This is harmless, but we should find out when the library default | |
| 225 // changes. | |
| 226 int send_size = usrsctp_sysctl_get_sctp_sendspace(); | |
| 227 if (send_size != kSendBufferSize) { | |
| 228 LOG(LS_ERROR) << "Got different send size than expected: " << send_size; | |
| 229 } | |
| 230 | |
| 231 // TODO(ldixon): Consider turning this on/off. | |
| 232 // This is not needed right now (we don't do dynamic address changes): | |
| 233 // If SCTP Auto-ASCONF is enabled, the peer is informed automatically | |
| 234 // when a new address is added or removed. This feature is enabled by | |
| 235 // default. | |
| 236 // usrsctp_sysctl_set_sctp_auto_asconf(0); | |
| 237 | |
| 238 // TODO(ldixon): Consider turning this on/off. | |
| 239 // Add a blackhole sysctl. Setting it to 1 results in no ABORTs | |
| 240 // being sent in response to INITs, setting it to 2 results | |
| 241 // in no ABORTs being sent for received OOTB packets. | |
| 242 // This is similar to the TCP sysctl. | |
| 243 // | |
| 244 // See: http://lakerest.net/pipermail/sctp-coders/2012-January/009438.html | |
| 245 // See: http://svnweb.freebsd.org/base?view=revision&revision=229805 | |
| 246 // usrsctp_sysctl_set_sctp_blackhole(2); | |
| 247 | |
| 248 // Set the number of default outgoing streams. This is the number we'll | |
| 249 // send in the SCTP INIT message. | |
| 250 usrsctp_sysctl_set_sctp_nr_outgoing_streams_default(kMaxSctpStreams); | |
| 251 } | |
| 252 | |
| 253 static void UninitializeUsrSctp() { | |
| 254 LOG(LS_INFO) << __FUNCTION__; | |
| 255 // usrsctp_finish() may fail if it's called too soon after the transports | |
| 256 // are | |
| 257 // closed. Wait and try again until it succeeds for up to 3 seconds. | |
| 258 for (size_t i = 0; i < 300; ++i) { | |
| 259 if (usrsctp_finish() == 0) { | |
| 260 return; | |
| 261 } | |
| 262 | |
| 263 rtc::Thread::SleepMs(10); | |
| 264 } | |
| 265 LOG(LS_ERROR) << "Failed to shutdown usrsctp."; | |
| 266 } | |
| 267 | |
| 268 static void IncrementUsrSctpUsageCount() { | |
| 269 rtc::GlobalLockScope lock(&g_usrsctp_lock_); | |
| 270 if (!g_usrsctp_usage_count) { | |
| 271 InitializeUsrSctp(); | |
| 272 } | |
| 273 ++g_usrsctp_usage_count; | |
| 274 } | |
| 275 | |
| 276 static void DecrementUsrSctpUsageCount() { | |
| 277 rtc::GlobalLockScope lock(&g_usrsctp_lock_); | |
| 278 --g_usrsctp_usage_count; | |
| 279 if (!g_usrsctp_usage_count) { | |
| 280 UninitializeUsrSctp(); | |
| 281 } | |
| 282 } | |
| 283 | |
| 284 // This is the callback usrsctp uses when there's data to send on the network | |
| 285 // that has been wrapped appropriatly for the SCTP protocol. | |
| 286 static int OnSctpOutboundPacket(void* addr, | |
| 287 void* data, | |
| 288 size_t length, | |
| 289 uint8_t tos, | |
| 290 uint8_t set_df) { | |
| 291 SctpTransport* transport = static_cast<SctpTransport*>(addr); | |
| 292 LOG(LS_VERBOSE) << "global OnSctpOutboundPacket():" | |
| 293 << "addr: " << addr << "; length: " << length | |
| 294 << "; tos: " << std::hex << static_cast<int>(tos) | |
| 295 << "; set_df: " << std::hex << static_cast<int>(set_df); | |
| 296 | |
| 297 VerboseLogPacket(data, length, SCTP_DUMP_OUTBOUND); | |
| 298 // Note: We have to copy the data; the caller will delete it. | |
| 299 rtc::CopyOnWriteBuffer buf(reinterpret_cast<uint8_t*>(data), length); | |
| 300 // TODO(deadbeef): Why do we need an AsyncInvoke here? We're already on the | |
| 301 // right thread and don't need to unwind the stack. | |
| 302 transport->invoker_.AsyncInvoke<void>( | |
| 303 RTC_FROM_HERE, transport->network_thread_, | |
| 304 rtc::Bind(&SctpTransport::OnPacketFromSctpToNetwork, transport, buf)); | |
| 305 return 0; | |
| 306 } | |
| 307 | |
| 308 // This is the callback called from usrsctp when data has been received, after | |
| 309 // a packet has been interpreted and parsed by usrsctp and found to contain | |
| 310 // payload data. It is called by a usrsctp thread. It is assumed this function | |
| 311 // will free the memory used by 'data'. | |
| 312 static int OnSctpInboundPacket(struct socket* sock, | |
| 313 union sctp_sockstore addr, | |
| 314 void* data, | |
| 315 size_t length, | |
| 316 struct sctp_rcvinfo rcv, | |
| 317 int flags, | |
| 318 void* ulp_info) { | |
| 319 SctpTransport* transport = static_cast<SctpTransport*>(ulp_info); | |
| 320 // Post data to the transport's receiver thread (copying it). | |
| 321 // TODO(ldixon): Unclear if copy is needed as this method is responsible for | |
| 322 // memory cleanup. But this does simplify code. | |
| 323 const PayloadProtocolIdentifier ppid = | |
| 324 static_cast<PayloadProtocolIdentifier>( | |
| 325 rtc::HostToNetwork32(rcv.rcv_ppid)); | |
| 326 DataMessageType type = DMT_NONE; | |
| 327 if (!GetDataMediaType(ppid, &type) && !(flags & MSG_NOTIFICATION)) { | |
| 328 // It's neither a notification nor a recognized data packet. Drop it. | |
| 329 LOG(LS_ERROR) << "Received an unknown PPID " << ppid | |
| 330 << " on an SCTP packet. Dropping."; | |
| 331 } else { | |
| 332 rtc::CopyOnWriteBuffer buffer; | |
| 333 ReceiveDataParams params; | |
| 334 buffer.SetData(reinterpret_cast<uint8_t*>(data), length); | |
| 335 params.sid = rcv.rcv_sid; | |
| 336 params.seq_num = rcv.rcv_ssn; | |
| 337 params.timestamp = rcv.rcv_tsn; | |
| 338 params.type = type; | |
| 339 // The ownership of the packet transfers to |invoker_|. Using | |
| 340 // CopyOnWriteBuffer is the most convenient way to do this. | |
| 341 transport->invoker_.AsyncInvoke<void>( | |
| 342 RTC_FROM_HERE, transport->network_thread_, | |
| 343 rtc::Bind(&SctpTransport::OnInboundPacketFromSctpToChannel, transport, | |
| 344 buffer, params, flags)); | |
| 345 } | |
| 346 free(data); | |
| 347 return 1; | |
| 348 } | |
| 349 | |
| 350 static SctpTransport* GetTransportFromSocket(struct socket* sock) { | |
| 351 struct sockaddr* addrs = nullptr; | |
| 352 int naddrs = usrsctp_getladdrs(sock, 0, &addrs); | |
| 353 if (naddrs <= 0 || addrs[0].sa_family != AF_CONN) { | |
| 354 return nullptr; | |
| 355 } | |
| 356 // usrsctp_getladdrs() returns the addresses bound to this socket, which | |
| 357 // contains the SctpTransport* as sconn_addr. Read the pointer, | |
| 358 // then free the list of addresses once we have the pointer. We only open | |
| 359 // AF_CONN sockets, and they should all have the sconn_addr set to the | |
| 360 // pointer that created them, so [0] is as good as any other. | |
| 361 struct sockaddr_conn* sconn = | |
| 362 reinterpret_cast<struct sockaddr_conn*>(&addrs[0]); | |
| 363 SctpTransport* transport = | |
| 364 reinterpret_cast<SctpTransport*>(sconn->sconn_addr); | |
| 365 usrsctp_freeladdrs(addrs); | |
| 366 | |
| 367 return transport; | |
| 368 } | |
| 369 | |
| 370 static int SendThresholdCallback(struct socket* sock, uint32_t sb_free) { | |
| 371 // Fired on our I/O thread. SctpTransport::OnPacketReceived() gets | |
| 372 // a packet containing acknowledgments, which goes into usrsctp_conninput, | |
| 373 // and then back here. | |
| 374 SctpTransport* transport = GetTransportFromSocket(sock); | |
| 375 if (!transport) { | |
| 376 LOG(LS_ERROR) | |
| 377 << "SendThresholdCallback: Failed to get transport for socket " | |
| 378 << sock; | |
| 379 return 0; | |
| 380 } | |
| 381 transport->OnSendThresholdCallback(); | |
| 382 return 0; | |
| 383 } | |
| 384 }; | |
| 385 | |
| 386 SctpTransport::SctpTransport(rtc::Thread* network_thread, | |
| 387 TransportChannel* channel) | |
| 388 : network_thread_(network_thread), | |
| 389 transport_channel_(channel), | |
| 390 was_ever_writable_(channel->writable()) { | |
| 391 RTC_DCHECK(network_thread_); | |
| 392 RTC_DCHECK(transport_channel_); | |
| 393 RTC_DCHECK_RUN_ON(network_thread_); | |
| 394 ConnectTransportChannelSignals(); | |
| 395 } | |
| 396 | |
| 397 SctpTransport::~SctpTransport() { | |
| 398 // Close abruptly; no reset procedure. | |
| 399 CloseSctpSocket(); | |
| 400 } | |
| 401 | |
| 402 void SctpTransport::SetTransportChannel(cricket::TransportChannel* channel) { | |
| 403 RTC_DCHECK_RUN_ON(network_thread_); | |
| 404 RTC_DCHECK(channel); | |
| 405 DisconnectTransportChannelSignals(); | |
| 406 transport_channel_ = channel; | |
| 407 ConnectTransportChannelSignals(); | |
| 408 if (!was_ever_writable_ && channel->writable()) { | |
| 409 was_ever_writable_ = true; | |
| 410 // New channel is writable, now we can start the SCTP connection if Start | |
| 411 // was called already. | |
| 412 if (started_) { | |
| 413 RTC_DCHECK(!sock_); | |
| 414 Connect(); | |
| 415 } | |
| 416 } | |
| 417 } | |
| 418 | |
| 419 bool SctpTransport::Start(int local_sctp_port, int remote_sctp_port) { | |
| 420 RTC_DCHECK_RUN_ON(network_thread_); | |
| 421 if (local_sctp_port == -1) { | |
| 422 local_sctp_port = kSctpDefaultPort; | |
| 423 } | |
| 424 if (remote_sctp_port == -1) { | |
| 425 remote_sctp_port = kSctpDefaultPort; | |
| 426 } | |
| 427 if (started_) { | |
| 428 if (local_sctp_port != local_port_ || remote_sctp_port != remote_port_) { | |
| 429 LOG(LS_ERROR) << "Can't change SCTP port after SCTP association formed."; | |
| 430 return false; | |
| 431 } | |
| 432 return true; | |
| 433 } | |
| 434 local_port_ = local_sctp_port; | |
| 435 remote_port_ = remote_sctp_port; | |
| 436 started_ = true; | |
| 437 RTC_DCHECK(!sock_); | |
| 438 // Only try to connect if the DTLS channel has been writable before | |
| 439 // (indicating that the DTLS handshake is complete). | |
| 440 if (was_ever_writable_) { | |
| 441 return Connect(); | |
| 442 } | |
| 443 return true; | |
| 444 } | |
| 445 | |
| 446 bool SctpTransport::OpenStream(int sid) { | |
| 447 RTC_DCHECK_RUN_ON(network_thread_); | |
| 448 if (sid > kMaxSctpSid) { | |
| 449 LOG(LS_WARNING) << debug_name_ << "->OpenStream(...): " | |
| 450 << "Not adding data stream " | |
| 451 << "with sid=" << sid << " because sid is too high."; | |
| 452 return false; | |
| 453 } else if (open_streams_.find(sid) != open_streams_.end()) { | |
| 454 LOG(LS_WARNING) << debug_name_ << "->OpenStream(...): " | |
| 455 << "Not adding data stream " | |
| 456 << "with sid=" << sid << " because stream is already open."; | |
| 457 return false; | |
| 458 } else if (queued_reset_streams_.find(sid) != queued_reset_streams_.end() || | |
| 459 sent_reset_streams_.find(sid) != sent_reset_streams_.end()) { | |
| 460 LOG(LS_WARNING) << debug_name_ << "->OpenStream(...): " | |
| 461 << "Not adding data stream " | |
| 462 << " with sid=" << sid | |
| 463 << " because stream is still closing."; | |
| 464 return false; | |
| 465 } | |
| 466 | |
| 467 open_streams_.insert(sid); | |
| 468 return true; | |
| 469 } | |
| 470 | |
| 471 bool SctpTransport::ResetStream(int sid) { | |
| 472 RTC_DCHECK_RUN_ON(network_thread_); | |
| 473 StreamSet::iterator found = open_streams_.find(sid); | |
| 474 if (found == open_streams_.end()) { | |
| 475 LOG(LS_WARNING) << debug_name_ << "->ResetStream(" << sid << "): " | |
| 476 << "stream not found."; | |
| 477 return false; | |
| 478 } else { | |
| 479 LOG(LS_VERBOSE) << debug_name_ << "->ResetStream(" << sid << "): " | |
| 480 << "Removing and queuing RE-CONFIG chunk."; | |
| 481 open_streams_.erase(found); | |
| 482 } | |
| 483 | |
| 484 // SCTP won't let you have more than one stream reset pending at a time, but | |
| 485 // you can close multiple streams in a single reset. So, we keep an internal | |
| 486 // queue of streams-to-reset, and send them as one reset message in | |
| 487 // SendQueuedStreamResets(). | |
| 488 queued_reset_streams_.insert(sid); | |
| 489 | |
| 490 // Signal our stream-reset logic that it should try to send now, if it can. | |
| 491 SendQueuedStreamResets(); | |
| 492 | |
| 493 // The stream will actually get removed when we get the acknowledgment. | |
| 494 return true; | |
| 495 } | |
| 496 | |
| 497 bool SctpTransport::SendData(const SendDataParams& params, | |
| 498 const rtc::CopyOnWriteBuffer& payload, | |
| 499 SendDataResult* result) { | |
| 500 RTC_DCHECK_RUN_ON(network_thread_); | |
| 501 if (result) { | |
| 502 // Preset |result| to assume an error. If SendData succeeds, we'll | |
| 503 // overwrite |*result| once more at the end. | |
| 504 *result = SDR_ERROR; | |
| 505 } | |
| 506 | |
| 507 if (!sock_) { | |
| 508 LOG(LS_WARNING) << debug_name_ << "->SendData(...): " | |
| 509 << "Not sending packet with sid=" << params.sid | |
| 510 << " len=" << payload.size() << " before Start()."; | |
| 511 return false; | |
| 512 } | |
| 513 | |
| 514 if (params.type != DMT_CONTROL && | |
| 515 open_streams_.find(params.sid) == open_streams_.end()) { | |
| 516 LOG(LS_WARNING) << debug_name_ << "->SendData(...): " | |
| 517 << "Not sending data because sid is unknown: " | |
| 518 << params.sid; | |
| 519 return false; | |
| 520 } | |
| 521 | |
| 522 // Send data using SCTP. | |
| 523 ssize_t send_res = 0; // result from usrsctp_sendv. | |
| 524 struct sctp_sendv_spa spa = {0}; | |
| 525 spa.sendv_flags |= SCTP_SEND_SNDINFO_VALID; | |
| 526 spa.sendv_sndinfo.snd_sid = params.sid; | |
| 527 spa.sendv_sndinfo.snd_ppid = rtc::HostToNetwork32(GetPpid(params.type)); | |
| 528 | |
| 529 // Ordered implies reliable. | |
| 530 if (!params.ordered) { | |
| 531 spa.sendv_sndinfo.snd_flags |= SCTP_UNORDERED; | |
| 532 if (params.max_rtx_count >= 0 || params.max_rtx_ms == 0) { | |
| 533 spa.sendv_flags |= SCTP_SEND_PRINFO_VALID; | |
| 534 spa.sendv_prinfo.pr_policy = SCTP_PR_SCTP_RTX; | |
| 535 spa.sendv_prinfo.pr_value = params.max_rtx_count; | |
| 536 } else { | |
| 537 spa.sendv_flags |= SCTP_SEND_PRINFO_VALID; | |
| 538 spa.sendv_prinfo.pr_policy = SCTP_PR_SCTP_TTL; | |
| 539 spa.sendv_prinfo.pr_value = params.max_rtx_ms; | |
| 540 } | |
| 541 } | |
| 542 | |
| 543 // We don't fragment. | |
| 544 send_res = usrsctp_sendv( | |
| 545 sock_, payload.data(), static_cast<size_t>(payload.size()), NULL, 0, &spa, | |
| 546 rtc::checked_cast<socklen_t>(sizeof(spa)), SCTP_SENDV_SPA, 0); | |
| 547 if (send_res < 0) { | |
| 548 if (errno == SCTP_EWOULDBLOCK) { | |
| 549 *result = SDR_BLOCK; | |
| 550 ready_to_send_data_ = false; | |
| 551 LOG(LS_INFO) << debug_name_ << "->SendData(...): EWOULDBLOCK returned"; | |
| 552 } else { | |
| 553 LOG_ERRNO(LS_ERROR) << "ERROR:" << debug_name_ << "->SendData(...): " | |
| 554 << " usrsctp_sendv: "; | |
| 555 } | |
| 556 return false; | |
| 557 } | |
| 558 if (result) { | |
| 559 // Only way out now is success. | |
| 560 *result = SDR_SUCCESS; | |
| 561 } | |
| 562 return true; | |
| 563 } | |
| 564 | |
| 565 bool SctpTransport::ReadyToSendData() { | |
| 566 RTC_DCHECK_RUN_ON(network_thread_); | |
| 567 return ready_to_send_data_; | |
| 568 } | |
| 569 | |
| 570 void SctpTransport::ConnectTransportChannelSignals() { | |
| 571 RTC_DCHECK_RUN_ON(network_thread_); | |
| 572 transport_channel_->SignalWritableState.connect( | |
| 573 this, &SctpTransport::OnWritableState); | |
| 574 transport_channel_->SignalReadPacket.connect(this, | |
| 575 &SctpTransport::OnPacketRead); | |
| 576 } | |
| 577 | |
| 578 void SctpTransport::DisconnectTransportChannelSignals() { | |
| 579 RTC_DCHECK_RUN_ON(network_thread_); | |
| 580 transport_channel_->SignalWritableState.disconnect(this); | |
| 581 transport_channel_->SignalReadPacket.disconnect(this); | |
| 582 } | |
| 583 | |
| 584 bool SctpTransport::Connect() { | |
| 585 RTC_DCHECK_RUN_ON(network_thread_); | |
| 586 LOG(LS_VERBOSE) << debug_name_ << "->Connect()."; | |
| 587 | |
| 588 // If we already have a socket connection (which shouldn't ever happen), just | |
| 589 // return. | |
| 590 RTC_DCHECK(!sock_); | |
| 591 if (sock_) { | |
| 592 LOG(LS_ERROR) << debug_name_ << "->Connect(): Ignored as socket " | |
| 593 "is already established."; | |
| 594 return true; | |
| 595 } | |
| 596 | |
| 597 // If no socket (it was closed) try to start it again. This can happen when | |
| 598 // the socket we are connecting to closes, does an sctp shutdown handshake, | |
| 599 // or behaves unexpectedly causing us to perform a CloseSctpSocket. | |
| 600 if (!OpenSctpSocket()) { | |
| 601 return false; | |
| 602 } | |
| 603 | |
| 604 // Note: conversion from int to uint16_t happens on assignment. | |
| 605 sockaddr_conn local_sconn = GetSctpSockAddr(local_port_); | |
| 606 if (usrsctp_bind(sock_, reinterpret_cast<sockaddr*>(&local_sconn), | |
| 607 sizeof(local_sconn)) < 0) { | |
| 608 LOG_ERRNO(LS_ERROR) << debug_name_ | |
| 609 << "->Connect(): " << ("Failed usrsctp_bind"); | |
| 610 CloseSctpSocket(); | |
| 611 return false; | |
| 612 } | |
| 613 | |
| 614 // Note: conversion from int to uint16_t happens on assignment. | |
| 615 sockaddr_conn remote_sconn = GetSctpSockAddr(remote_port_); | |
| 616 int connect_result = usrsctp_connect( | |
| 617 sock_, reinterpret_cast<sockaddr*>(&remote_sconn), sizeof(remote_sconn)); | |
| 618 if (connect_result < 0 && errno != SCTP_EINPROGRESS) { | |
| 619 LOG_ERRNO(LS_ERROR) << debug_name_ << "->Connect(): " | |
| 620 << "Failed usrsctp_connect. got errno=" << errno | |
| 621 << ", but wanted " << SCTP_EINPROGRESS; | |
| 622 CloseSctpSocket(); | |
| 623 return false; | |
| 624 } | |
| 625 // Set the MTU and disable MTU discovery. | |
| 626 // We can only do this after usrsctp_connect or it has no effect. | |
| 627 sctp_paddrparams params = {{0}}; | |
| 628 memcpy(¶ms.spp_address, &remote_sconn, sizeof(remote_sconn)); | |
| 629 params.spp_flags = SPP_PMTUD_DISABLE; | |
| 630 params.spp_pathmtu = kSctpMtu; | |
| 631 if (usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_PEER_ADDR_PARAMS, ¶ms, | |
| 632 sizeof(params))) { | |
| 633 LOG_ERRNO(LS_ERROR) << debug_name_ << "->Connect(): " | |
| 634 << "Failed to set SCTP_PEER_ADDR_PARAMS."; | |
| 635 } | |
| 636 // Since this is a fresh SCTP association, we'll always start out with empty | |
| 637 // queues, so "ReadyToSendData" should be true. | |
| 638 SetReadyToSendData(); | |
| 639 return true; | |
| 640 } | |
| 641 | |
| 642 bool SctpTransport::OpenSctpSocket() { | |
| 643 RTC_DCHECK_RUN_ON(network_thread_); | |
| 644 if (sock_) { | |
| 645 LOG(LS_WARNING) << debug_name_ << "->OpenSctpSocket(): " | |
| 646 << "Ignoring attempt to re-create existing socket."; | |
| 647 return false; | |
| 648 } | |
| 649 | |
| 650 UsrSctpWrapper::IncrementUsrSctpUsageCount(); | |
| 651 | |
| 652 // If kSendBufferSize isn't reflective of reality, we log an error, but we | |
| 653 // still have to do something reasonable here. Look up what the buffer's | |
| 654 // real size is and set our threshold to something reasonable. | |
| 655 static const int kSendThreshold = usrsctp_sysctl_get_sctp_sendspace() / 2; | |
| 656 | |
| 657 sock_ = usrsctp_socket( | |
| 658 AF_CONN, SOCK_STREAM, IPPROTO_SCTP, &UsrSctpWrapper::OnSctpInboundPacket, | |
| 659 &UsrSctpWrapper::SendThresholdCallback, kSendThreshold, this); | |
| 660 if (!sock_) { | |
| 661 LOG_ERRNO(LS_ERROR) << debug_name_ << "->OpenSctpSocket(): " | |
| 662 << "Failed to create SCTP socket."; | |
| 663 UsrSctpWrapper::DecrementUsrSctpUsageCount(); | |
| 664 return false; | |
| 665 } | |
| 666 | |
| 667 if (!ConfigureSctpSocket()) { | |
| 668 usrsctp_close(sock_); | |
| 669 sock_ = nullptr; | |
| 670 UsrSctpWrapper::DecrementUsrSctpUsageCount(); | |
| 671 return false; | |
| 672 } | |
| 673 // Register this class as an address for usrsctp. This is used by SCTP to | |
| 674 // direct the packets received (by the created socket) to this class. | |
| 675 usrsctp_register_address(this); | |
| 676 return true; | |
| 677 } | |
| 678 | |
| 679 bool SctpTransport::ConfigureSctpSocket() { | |
| 680 RTC_DCHECK_RUN_ON(network_thread_); | |
| 681 RTC_DCHECK(sock_); | |
| 682 // Make the socket non-blocking. Connect, close, shutdown etc will not block | |
| 683 // the thread waiting for the socket operation to complete. | |
| 684 if (usrsctp_set_non_blocking(sock_, 1) < 0) { | |
| 685 LOG_ERRNO(LS_ERROR) << debug_name_ << "->ConfigureSctpSocket(): " | |
| 686 << "Failed to set SCTP to non blocking."; | |
| 687 return false; | |
| 688 } | |
| 689 | |
| 690 // This ensures that the usrsctp close call deletes the association. This | |
| 691 // prevents usrsctp from calling OnSctpOutboundPacket with references to | |
| 692 // this class as the address. | |
| 693 linger linger_opt; | |
| 694 linger_opt.l_onoff = 1; | |
| 695 linger_opt.l_linger = 0; | |
| 696 if (usrsctp_setsockopt(sock_, SOL_SOCKET, SO_LINGER, &linger_opt, | |
| 697 sizeof(linger_opt))) { | |
| 698 LOG_ERRNO(LS_ERROR) << debug_name_ << "->ConfigureSctpSocket(): " | |
| 699 << "Failed to set SO_LINGER."; | |
| 700 return false; | |
| 701 } | |
| 702 | |
| 703 // Enable stream ID resets. | |
| 704 struct sctp_assoc_value stream_rst; | |
| 705 stream_rst.assoc_id = SCTP_ALL_ASSOC; | |
| 706 stream_rst.assoc_value = 1; | |
| 707 if (usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_ENABLE_STREAM_RESET, | |
| 708 &stream_rst, sizeof(stream_rst))) { | |
| 709 LOG_ERRNO(LS_ERROR) << debug_name_ << "->ConfigureSctpSocket(): " | |
| 710 | |
| 711 << "Failed to set SCTP_ENABLE_STREAM_RESET."; | |
| 712 return false; | |
| 713 } | |
| 714 | |
| 715 // Nagle. | |
| 716 uint32_t nodelay = 1; | |
| 717 if (usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_NODELAY, &nodelay, | |
| 718 sizeof(nodelay))) { | |
| 719 LOG_ERRNO(LS_ERROR) << debug_name_ << "->ConfigureSctpSocket(): " | |
| 720 << "Failed to set SCTP_NODELAY."; | |
| 721 return false; | |
| 722 } | |
| 723 | |
| 724 // Subscribe to SCTP event notifications. | |
| 725 int event_types[] = {SCTP_ASSOC_CHANGE, SCTP_PEER_ADDR_CHANGE, | |
| 726 SCTP_SEND_FAILED_EVENT, SCTP_SENDER_DRY_EVENT, | |
| 727 SCTP_STREAM_RESET_EVENT}; | |
| 728 struct sctp_event event = {0}; | |
| 729 event.se_assoc_id = SCTP_ALL_ASSOC; | |
| 730 event.se_on = 1; | |
| 731 for (size_t i = 0; i < arraysize(event_types); i++) { | |
| 732 event.se_type = event_types[i]; | |
| 733 if (usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_EVENT, &event, | |
| 734 sizeof(event)) < 0) { | |
| 735 LOG_ERRNO(LS_ERROR) << debug_name_ << "->ConfigureSctpSocket(): " | |
| 736 | |
| 737 << "Failed to set SCTP_EVENT type: " << event.se_type; | |
| 738 return false; | |
| 739 } | |
| 740 } | |
| 741 return true; | |
| 742 } | |
| 743 | |
| 744 void SctpTransport::CloseSctpSocket() { | |
| 745 RTC_DCHECK_RUN_ON(network_thread_); | |
| 746 if (sock_) { | |
| 747 // We assume that SO_LINGER option is set to close the association when | |
| 748 // close is called. This means that any pending packets in usrsctp will be | |
| 749 // discarded instead of being sent. | |
| 750 usrsctp_close(sock_); | |
| 751 sock_ = nullptr; | |
| 752 usrsctp_deregister_address(this); | |
| 753 UsrSctpWrapper::DecrementUsrSctpUsageCount(); | |
| 754 ready_to_send_data_ = false; | |
| 755 } | |
| 756 } | |
| 757 | |
| 758 bool SctpTransport::SendQueuedStreamResets() { | |
| 759 RTC_DCHECK_RUN_ON(network_thread_); | |
| 760 if (!sent_reset_streams_.empty() || queued_reset_streams_.empty()) { | |
| 761 return true; | |
| 762 } | |
| 763 | |
| 764 LOG(LS_VERBOSE) << "SendQueuedStreamResets[" << debug_name_ << "]: Sending [" | |
| 765 << ListStreams(queued_reset_streams_) << "], Open: [" | |
| 766 << ListStreams(open_streams_) << "], Sent: [" | |
| 767 << ListStreams(sent_reset_streams_) << "]"; | |
| 768 | |
| 769 const size_t num_streams = queued_reset_streams_.size(); | |
| 770 const size_t num_bytes = | |
| 771 sizeof(struct sctp_reset_streams) + (num_streams * sizeof(uint16_t)); | |
| 772 | |
| 773 std::vector<uint8_t> reset_stream_buf(num_bytes, 0); | |
| 774 struct sctp_reset_streams* resetp = | |
| 775 reinterpret_cast<sctp_reset_streams*>(&reset_stream_buf[0]); | |
| 776 resetp->srs_assoc_id = SCTP_ALL_ASSOC; | |
| 777 resetp->srs_flags = SCTP_STREAM_RESET_INCOMING | SCTP_STREAM_RESET_OUTGOING; | |
| 778 resetp->srs_number_streams = rtc::checked_cast<uint16_t>(num_streams); | |
| 779 int result_idx = 0; | |
| 780 for (StreamSet::iterator it = queued_reset_streams_.begin(); | |
| 781 it != queued_reset_streams_.end(); ++it) { | |
| 782 resetp->srs_stream_list[result_idx++] = *it; | |
| 783 } | |
| 784 | |
| 785 int ret = | |
| 786 usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_RESET_STREAMS, resetp, | |
| 787 rtc::checked_cast<socklen_t>(reset_stream_buf.size())); | |
| 788 if (ret < 0) { | |
| 789 LOG_ERRNO(LS_ERROR) << debug_name_ << "->SendQueuedStreamResets(): " | |
| 790 "Failed to send a stream reset for " | |
| 791 << num_streams << " streams"; | |
| 792 return false; | |
| 793 } | |
| 794 | |
| 795 // sent_reset_streams_ is empty, and all the queued_reset_streams_ go into | |
| 796 // it now. | |
| 797 queued_reset_streams_.swap(sent_reset_streams_); | |
| 798 return true; | |
| 799 } | |
| 800 | |
| 801 void SctpTransport::SetReadyToSendData() { | |
| 802 RTC_DCHECK_RUN_ON(network_thread_); | |
| 803 if (!ready_to_send_data_) { | |
| 804 ready_to_send_data_ = true; | |
| 805 SignalReadyToSendData(); | |
| 806 } | |
| 807 } | |
| 808 | |
| 809 void SctpTransport::OnWritableState(rtc::PacketTransportInterface* transport) { | |
| 810 RTC_DCHECK_RUN_ON(network_thread_); | |
| 811 RTC_DCHECK_EQ(transport_channel_, transport); | |
| 812 if (!was_ever_writable_ && transport->writable()) { | |
| 813 was_ever_writable_ = true; | |
| 814 if (started_) { | |
| 815 Connect(); | |
| 816 } | |
| 817 } | |
| 818 } | |
| 819 | |
| 820 // Called by network interface when a packet has been received. | |
| 821 void SctpTransport::OnPacketRead(rtc::PacketTransportInterface* transport, | |
| 822 const char* data, | |
| 823 size_t len, | |
| 824 const rtc::PacketTime& packet_time, | |
| 825 int flags) { | |
| 826 RTC_DCHECK_RUN_ON(network_thread_); | |
| 827 RTC_DCHECK_EQ(transport_channel_, transport); | |
| 828 TRACE_EVENT0("webrtc", "SctpTransport::OnPacketRead"); | |
| 829 | |
| 830 // TODO(pthatcher): Do this in a more robust way by checking for | |
| 831 // SCTP or DTLS. | |
| 832 if (IsRtpPacket(data, len)) { | |
| 833 return; | |
| 834 } | |
| 835 | |
| 836 LOG(LS_VERBOSE) << debug_name_ << "->OnPacketRead(...): " | |
| 837 << " length=" << len << ", started: " << started_; | |
| 838 // Only give receiving packets to usrsctp after if connected. This enables two | |
| 839 // peers to each make a connect call, but for them not to receive an INIT | |
| 840 // packet before they have called connect; least the last receiver of the INIT | |
| 841 // packet will have called connect, and a connection will be established. | |
| 842 if (sock_) { | |
| 843 // Pass received packet to SCTP stack. Once processed by usrsctp, the data | |
| 844 // will be will be given to the global OnSctpInboundData, and then, | |
| 845 // marshalled by the AsyncInvoker. | |
| 846 VerboseLogPacket(data, len, SCTP_DUMP_INBOUND); | |
| 847 usrsctp_conninput(this, data, len, 0); | |
| 848 } else { | |
| 849 // TODO(ldixon): Consider caching the packet for very slightly better | |
| 850 // reliability. | |
| 851 } | |
| 852 } | |
| 853 | |
| 854 void SctpTransport::OnSendThresholdCallback() { | |
| 855 RTC_DCHECK_RUN_ON(network_thread_); | |
| 856 SetReadyToSendData(); | |
| 857 } | |
| 858 | |
| 859 sockaddr_conn SctpTransport::GetSctpSockAddr(int port) { | |
| 860 sockaddr_conn sconn = {0}; | |
| 861 sconn.sconn_family = AF_CONN; | |
| 862 #ifdef HAVE_SCONN_LEN | |
| 863 sconn.sconn_len = sizeof(sockaddr_conn); | |
| 864 #endif | |
| 865 // Note: conversion from int to uint16_t happens here. | |
| 866 sconn.sconn_port = rtc::HostToNetwork16(port); | |
| 867 sconn.sconn_addr = this; | |
| 868 return sconn; | |
| 869 } | |
| 870 | |
| 871 void SctpTransport::OnPacketFromSctpToNetwork( | |
| 872 const rtc::CopyOnWriteBuffer& buffer) { | |
| 873 RTC_DCHECK_RUN_ON(network_thread_); | |
| 874 if (buffer.size() > (kSctpMtu)) { | |
| 875 LOG(LS_ERROR) << debug_name_ << "->OnPacketFromSctpToNetwork(...): " | |
| 876 << "SCTP seems to have made a packet that is bigger " | |
| 877 << "than its official MTU: " << buffer.size() << " vs max of " | |
| 878 << kSctpMtu; | |
| 879 } | |
| 880 TRACE_EVENT0("webrtc", "SctpTransport::OnPacketFromSctpToNetwork"); | |
| 881 | |
| 882 // Don't create noise by trying to send a packet when the DTLS channel isn't | |
| 883 // even writable. | |
| 884 if (!transport_channel_->writable()) { | |
| 885 return; | |
| 886 } | |
| 887 | |
| 888 // Bon voyage. | |
| 889 transport_channel_->SendPacket(buffer.data<char>(), buffer.size(), | |
| 890 rtc::PacketOptions(), PF_NORMAL); | |
| 891 } | |
| 892 | |
| 893 void SctpTransport::OnInboundPacketFromSctpToChannel( | |
| 894 const rtc::CopyOnWriteBuffer& buffer, | |
| 895 ReceiveDataParams params, | |
| 896 int flags) { | |
| 897 RTC_DCHECK_RUN_ON(network_thread_); | |
| 898 LOG(LS_VERBOSE) << debug_name_ << "->OnInboundPacketFromSctpToChannel(...): " | |
| 899 << "Received SCTP data:" | |
| 900 << " sid=" << params.sid | |
| 901 << " notification: " << (flags & MSG_NOTIFICATION) | |
| 902 << " length=" << buffer.size(); | |
| 903 // Sending a packet with data == NULL (no data) is SCTPs "close the | |
| 904 // connection" message. This sets sock_ = NULL; | |
| 905 if (!buffer.size() || !buffer.data()) { | |
| 906 LOG(LS_INFO) << debug_name_ << "->OnInboundPacketFromSctpToChannel(...): " | |
| 907 "No data, closing."; | |
| 908 return; | |
| 909 } | |
| 910 if (flags & MSG_NOTIFICATION) { | |
| 911 OnNotificationFromSctp(buffer); | |
| 912 } else { | |
| 913 OnDataFromSctpToChannel(params, buffer); | |
| 914 } | |
| 915 } | |
| 916 | |
| 917 void SctpTransport::OnDataFromSctpToChannel( | |
| 918 const ReceiveDataParams& params, | |
| 919 const rtc::CopyOnWriteBuffer& buffer) { | |
| 920 RTC_DCHECK_RUN_ON(network_thread_); | |
| 921 LOG(LS_VERBOSE) << debug_name_ << "->OnDataFromSctpToChannel(...): " | |
| 922 << "Posting with length: " << buffer.size() << " on stream " | |
| 923 << params.sid; | |
| 924 // Reports all received messages to upper layers, no matter whether the sid | |
| 925 // is known. | |
| 926 SignalDataReceived(params, buffer); | |
| 927 } | |
| 928 | |
| 929 void SctpTransport::OnNotificationFromSctp( | |
| 930 const rtc::CopyOnWriteBuffer& buffer) { | |
| 931 RTC_DCHECK_RUN_ON(network_thread_); | |
| 932 const sctp_notification& notification = | |
| 933 reinterpret_cast<const sctp_notification&>(*buffer.data()); | |
| 934 RTC_DCHECK(notification.sn_header.sn_length == buffer.size()); | |
| 935 | |
| 936 // TODO(ldixon): handle notifications appropriately. | |
| 937 switch (notification.sn_header.sn_type) { | |
| 938 case SCTP_ASSOC_CHANGE: | |
| 939 LOG(LS_VERBOSE) << "SCTP_ASSOC_CHANGE"; | |
| 940 OnNotificationAssocChange(notification.sn_assoc_change); | |
| 941 break; | |
| 942 case SCTP_REMOTE_ERROR: | |
| 943 LOG(LS_INFO) << "SCTP_REMOTE_ERROR"; | |
| 944 break; | |
| 945 case SCTP_SHUTDOWN_EVENT: | |
| 946 LOG(LS_INFO) << "SCTP_SHUTDOWN_EVENT"; | |
| 947 break; | |
| 948 case SCTP_ADAPTATION_INDICATION: | |
| 949 LOG(LS_INFO) << "SCTP_ADAPTATION_INDICATION"; | |
| 950 break; | |
| 951 case SCTP_PARTIAL_DELIVERY_EVENT: | |
| 952 LOG(LS_INFO) << "SCTP_PARTIAL_DELIVERY_EVENT"; | |
| 953 break; | |
| 954 case SCTP_AUTHENTICATION_EVENT: | |
| 955 LOG(LS_INFO) << "SCTP_AUTHENTICATION_EVENT"; | |
| 956 break; | |
| 957 case SCTP_SENDER_DRY_EVENT: | |
| 958 LOG(LS_VERBOSE) << "SCTP_SENDER_DRY_EVENT"; | |
| 959 SetReadyToSendData(); | |
| 960 break; | |
| 961 // TODO(ldixon): Unblock after congestion. | |
| 962 case SCTP_NOTIFICATIONS_STOPPED_EVENT: | |
| 963 LOG(LS_INFO) << "SCTP_NOTIFICATIONS_STOPPED_EVENT"; | |
| 964 break; | |
| 965 case SCTP_SEND_FAILED_EVENT: | |
| 966 LOG(LS_INFO) << "SCTP_SEND_FAILED_EVENT"; | |
| 967 break; | |
| 968 case SCTP_STREAM_RESET_EVENT: | |
| 969 OnStreamResetEvent(¬ification.sn_strreset_event); | |
| 970 break; | |
| 971 case SCTP_ASSOC_RESET_EVENT: | |
| 972 LOG(LS_INFO) << "SCTP_ASSOC_RESET_EVENT"; | |
| 973 break; | |
| 974 case SCTP_STREAM_CHANGE_EVENT: | |
| 975 LOG(LS_INFO) << "SCTP_STREAM_CHANGE_EVENT"; | |
| 976 // An acknowledgment we get after our stream resets have gone through, | |
| 977 // if they've failed. We log the message, but don't react -- we don't | |
| 978 // keep around the last-transmitted set of SSIDs we wanted to close for | |
| 979 // error recovery. It doesn't seem likely to occur, and if so, likely | |
| 980 // harmless within the lifetime of a single SCTP association. | |
| 981 break; | |
| 982 default: | |
| 983 LOG(LS_WARNING) << "Unknown SCTP event: " | |
| 984 << notification.sn_header.sn_type; | |
| 985 break; | |
| 986 } | |
| 987 } | |
| 988 | |
| 989 void SctpTransport::OnNotificationAssocChange(const sctp_assoc_change& change) { | |
| 990 RTC_DCHECK_RUN_ON(network_thread_); | |
| 991 switch (change.sac_state) { | |
| 992 case SCTP_COMM_UP: | |
| 993 LOG(LS_VERBOSE) << "Association change SCTP_COMM_UP"; | |
| 994 break; | |
| 995 case SCTP_COMM_LOST: | |
| 996 LOG(LS_INFO) << "Association change SCTP_COMM_LOST"; | |
| 997 break; | |
| 998 case SCTP_RESTART: | |
| 999 LOG(LS_INFO) << "Association change SCTP_RESTART"; | |
| 1000 break; | |
| 1001 case SCTP_SHUTDOWN_COMP: | |
| 1002 LOG(LS_INFO) << "Association change SCTP_SHUTDOWN_COMP"; | |
| 1003 break; | |
| 1004 case SCTP_CANT_STR_ASSOC: | |
| 1005 LOG(LS_INFO) << "Association change SCTP_CANT_STR_ASSOC"; | |
| 1006 break; | |
| 1007 default: | |
| 1008 LOG(LS_INFO) << "Association change UNKNOWN"; | |
| 1009 break; | |
| 1010 } | |
| 1011 } | |
| 1012 | |
| 1013 void SctpTransport::OnStreamResetEvent( | |
| 1014 const struct sctp_stream_reset_event* evt) { | |
| 1015 RTC_DCHECK_RUN_ON(network_thread_); | |
| 1016 // A stream reset always involves two RE-CONFIG chunks for us -- we always | |
| 1017 // simultaneously reset a sid's sequence number in both directions. The | |
| 1018 // requesting side transmits a RE-CONFIG chunk and waits for the peer to send | |
| 1019 // one back. Both sides get this SCTP_STREAM_RESET_EVENT when they receive | |
| 1020 // RE-CONFIGs. | |
| 1021 const int num_sids = (evt->strreset_length - sizeof(*evt)) / | |
| 1022 sizeof(evt->strreset_stream_list[0]); | |
| 1023 LOG(LS_VERBOSE) << "SCTP_STREAM_RESET_EVENT(" << debug_name_ | |
| 1024 << "): Flags = 0x" << std::hex << evt->strreset_flags << " (" | |
| 1025 << ListFlags(evt->strreset_flags) << ")"; | |
| 1026 LOG(LS_VERBOSE) << "Assoc = " << evt->strreset_assoc_id << ", Streams = [" | |
| 1027 << ListArray(evt->strreset_stream_list, num_sids) | |
| 1028 << "], Open: [" << ListStreams(open_streams_) << "], Q'd: [" | |
| 1029 << ListStreams(queued_reset_streams_) << "], Sent: [" | |
| 1030 << ListStreams(sent_reset_streams_) << "]"; | |
| 1031 | |
| 1032 // If both sides try to reset some streams at the same time (even if they're | |
| 1033 // disjoint sets), we can get reset failures. | |
| 1034 if (evt->strreset_flags & SCTP_STREAM_RESET_FAILED) { | |
| 1035 // OK, just try again. The stream IDs sent over when the RESET_FAILED flag | |
| 1036 // is set seem to be garbage values. Ignore them. | |
| 1037 queued_reset_streams_.insert(sent_reset_streams_.begin(), | |
| 1038 sent_reset_streams_.end()); | |
| 1039 sent_reset_streams_.clear(); | |
| 1040 | |
| 1041 } else if (evt->strreset_flags & SCTP_STREAM_RESET_INCOMING_SSN) { | |
| 1042 // Each side gets an event for each direction of a stream. That is, | |
| 1043 // closing sid k will make each side receive INCOMING and OUTGOING reset | |
| 1044 // events for k. As per RFC6525, Section 5, paragraph 2, each side will | |
| 1045 // get an INCOMING event first. | |
| 1046 for (int i = 0; i < num_sids; i++) { | |
| 1047 const int stream_id = evt->strreset_stream_list[i]; | |
| 1048 | |
| 1049 // See if this stream ID was closed by our peer or ourselves. | |
| 1050 StreamSet::iterator it = sent_reset_streams_.find(stream_id); | |
| 1051 | |
| 1052 // The reset was requested locally. | |
| 1053 if (it != sent_reset_streams_.end()) { | |
| 1054 LOG(LS_VERBOSE) << "SCTP_STREAM_RESET_EVENT(" << debug_name_ | |
| 1055 << "): local sid " << stream_id << " acknowledged."; | |
| 1056 sent_reset_streams_.erase(it); | |
| 1057 | |
| 1058 } else if ((it = open_streams_.find(stream_id)) != open_streams_.end()) { | |
| 1059 // The peer requested the reset. | |
| 1060 LOG(LS_VERBOSE) << "SCTP_STREAM_RESET_EVENT(" << debug_name_ | |
| 1061 << "): closing sid " << stream_id; | |
| 1062 open_streams_.erase(it); | |
| 1063 SignalStreamClosedRemotely(stream_id); | |
| 1064 | |
| 1065 } else if ((it = queued_reset_streams_.find(stream_id)) != | |
| 1066 queued_reset_streams_.end()) { | |
| 1067 // The peer requested the reset, but there was a local reset | |
| 1068 // queued. | |
| 1069 LOG(LS_VERBOSE) << "SCTP_STREAM_RESET_EVENT(" << debug_name_ | |
| 1070 << "): double-sided close for sid " << stream_id; | |
| 1071 // Both sides want the stream closed, and the peer got to send the | |
| 1072 // RE-CONFIG first. Treat it like the local Remove(Send|Recv)Stream | |
| 1073 // finished quickly. | |
| 1074 queued_reset_streams_.erase(it); | |
| 1075 | |
| 1076 } else { | |
| 1077 // This stream is unknown. Sometimes this can be from an | |
| 1078 // RESET_FAILED-related retransmit. | |
| 1079 LOG(LS_VERBOSE) << "SCTP_STREAM_RESET_EVENT(" << debug_name_ | |
| 1080 << "): Unknown sid " << stream_id; | |
| 1081 } | |
| 1082 } | |
| 1083 } | |
| 1084 | |
| 1085 // Always try to send the queued RESET because this call indicates that the | |
| 1086 // last local RESET or remote RESET has made some progress. | |
| 1087 SendQueuedStreamResets(); | |
| 1088 } | |
| 1089 | |
| 1090 } // namespace cricket | |
| OLD | NEW |
|---|
Chromium Code Reviews
Issue 2614813003:
Revert of Separating SCTP code from BaseChannel/MediaChannel. (Closed)
