henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1 | /* |
| 2 | * libjingle SCTP |
| 3 | * Copyright 2012 Google Inc, and Robin Seggelmann |
| 4 | * |
| 5 | * Redistribution and use in source and binary forms, with or without |
| 6 | * modification, are permitted provided that the following conditions are met: |
| 7 | * |
| 8 | * 1. Redistributions of source code must retain the above copyright notice, |
| 9 | * this list of conditions and the following disclaimer. |
| 10 | * 2. Redistributions in binary form must reproduce the above copyright notice, |
| 11 | * this list of conditions and the following disclaimer in the documentation |
| 12 | * and/or other materials provided with the distribution. |
| 13 | * 3. The name of the author may not be used to endorse or promote products |
| 14 | * derived from this software without specific prior written permission. |
| 15 | * |
| 16 | * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED |
| 17 | * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF |
| 18 | * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO |
| 19 | * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, |
| 20 | * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, |
| 21 | * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; |
| 22 | * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, |
| 23 | * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR |
| 24 | * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF |
| 25 | * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. |
| 26 | */ |
| 27 | |
| 28 | #include "talk/media/sctp/sctpdataengine.h" |
| 29 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 30 | #include <stdarg.h> |
| 31 | #include <stdio.h> |
| 32 | #include <vector> |
| 33 | |
| 34 | #include "talk/base/buffer.h" |
| 35 | #include "talk/base/helpers.h" |
| 36 | #include "talk/base/logging.h" |
| 37 | #include "talk/media/base/codec.h" |
| 38 | #include "talk/media/base/constants.h" |
| 39 | #include "talk/media/base/streamparams.h" |
| 40 | #include "usrsctplib/usrsctp.h" |
| 41 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 42 | namespace cricket { |
| 43 | |
| 44 | // This is the SCTP port to use. It is passed along the wire and the listener |
| 45 | // and connector must be using the same port. It is not related to the ports at |
| 46 | // the IP level. (Corresponds to: sockaddr_conn.sconn_port in usrsctp.h) |
| 47 | // |
| 48 | // TODO(ldixon): Allow port to be set from higher level code. |
| 49 | static const int kSctpDefaultPort = 5001; |
| 50 | // TODO(ldixon): Find where this is defined, and also check is Sctp really |
| 51 | // respects this. |
| 52 | static const size_t kSctpMtu = 1280; |
| 53 | |
| 54 | enum { |
| 55 | MSG_SCTPINBOUNDPACKET = 1, // MessageData is SctpInboundPacket |
| 56 | MSG_SCTPOUTBOUNDPACKET = 2, // MessageData is talk_base:Buffer |
| 57 | }; |
| 58 | |
| 59 | struct SctpInboundPacket { |
| 60 | talk_base::Buffer buffer; |
| 61 | ReceiveDataParams params; |
| 62 | // The |flags| parameter is used by SCTP to distinguish notification packets |
| 63 | // from other types of packets. |
| 64 | int flags; |
| 65 | }; |
| 66 | |
mallinath@webrtc.org | 1112c30 | 2013-09-23 20:34:45 +0000 | [diff] [blame] | 67 | // Helper for logging SCTP messages. |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 68 | static void debug_sctp_printf(const char *format, ...) { |
| 69 | char s[255]; |
| 70 | va_list ap; |
| 71 | va_start(ap, format); |
| 72 | vsnprintf(s, sizeof(s), format, ap); |
mallinath@webrtc.org | 1112c30 | 2013-09-23 20:34:45 +0000 | [diff] [blame] | 73 | LOG(LS_INFO) << "SCTP: " << s; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 74 | va_end(ap); |
| 75 | } |
| 76 | |
mallinath@webrtc.org | 1112c30 | 2013-09-23 20:34:45 +0000 | [diff] [blame] | 77 | // Get the PPID to use for the terminating fragment of this type. |
| 78 | static SctpDataMediaChannel::PayloadProtocolIdentifier GetPpid( |
| 79 | cricket::DataMessageType type) { |
| 80 | switch (type) { |
| 81 | default: |
| 82 | case cricket::DMT_NONE: |
| 83 | return SctpDataMediaChannel::PPID_NONE; |
| 84 | case cricket::DMT_CONTROL: |
| 85 | return SctpDataMediaChannel::PPID_CONTROL; |
| 86 | case cricket::DMT_BINARY: |
| 87 | return SctpDataMediaChannel::PPID_BINARY_LAST; |
| 88 | case cricket::DMT_TEXT: |
| 89 | return SctpDataMediaChannel::PPID_TEXT_LAST; |
| 90 | }; |
| 91 | } |
| 92 | |
| 93 | static bool GetDataMediaType( |
| 94 | SctpDataMediaChannel::PayloadProtocolIdentifier ppid, |
| 95 | cricket::DataMessageType *dest) { |
| 96 | ASSERT(dest != NULL); |
| 97 | switch (ppid) { |
| 98 | case SctpDataMediaChannel::PPID_BINARY_PARTIAL: |
| 99 | case SctpDataMediaChannel::PPID_BINARY_LAST: |
| 100 | *dest = cricket::DMT_BINARY; |
| 101 | return true; |
| 102 | |
| 103 | case SctpDataMediaChannel::PPID_TEXT_PARTIAL: |
| 104 | case SctpDataMediaChannel::PPID_TEXT_LAST: |
| 105 | *dest = cricket::DMT_TEXT; |
| 106 | return true; |
| 107 | |
| 108 | case SctpDataMediaChannel::PPID_CONTROL: |
| 109 | *dest = cricket::DMT_CONTROL; |
| 110 | return true; |
| 111 | |
| 112 | case SctpDataMediaChannel::PPID_NONE: |
| 113 | *dest = cricket::DMT_NONE; |
| 114 | return true; |
| 115 | |
| 116 | default: |
| 117 | return false; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 118 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 119 | } |
| 120 | |
| 121 | // This is the callback usrsctp uses when there's data to send on the network |
| 122 | // that has been wrapped appropriatly for the SCTP protocol. |
| 123 | static int OnSctpOutboundPacket(void* addr, void* data, size_t length, |
| 124 | uint8_t tos, uint8_t set_df) { |
| 125 | SctpDataMediaChannel* channel = static_cast<SctpDataMediaChannel*>(addr); |
| 126 | LOG(LS_VERBOSE) << "global OnSctpOutboundPacket():" |
| 127 | << "addr: " << addr << "; length: " << length |
| 128 | << "; tos: " << std::hex << static_cast<int>(tos) |
mallinath@webrtc.org | 1112c30 | 2013-09-23 20:34:45 +0000 | [diff] [blame] | 129 | << "; set_df: " << std::hex << static_cast<int>(set_df); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 130 | // Note: We have to copy the data; the caller will delete it. |
| 131 | talk_base::Buffer* buffer = new talk_base::Buffer(data, length); |
| 132 | channel->worker_thread()->Post(channel, MSG_SCTPOUTBOUNDPACKET, |
| 133 | talk_base::WrapMessageData(buffer)); |
| 134 | return 0; |
| 135 | } |
| 136 | |
| 137 | // This is the callback called from usrsctp when data has been received, after |
| 138 | // a packet has been interpreted and parsed by usrsctp and found to contain |
| 139 | // payload data. It is called by a usrsctp thread. It is assumed this function |
| 140 | // will free the memory used by 'data'. |
| 141 | static int OnSctpInboundPacket(struct socket* sock, union sctp_sockstore addr, |
| 142 | void* data, size_t length, |
| 143 | struct sctp_rcvinfo rcv, int flags, |
| 144 | void* ulp_info) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 145 | SctpDataMediaChannel* channel = static_cast<SctpDataMediaChannel*>(ulp_info); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 146 | // Post data to the channel's receiver thread (copying it). |
| 147 | // TODO(ldixon): Unclear if copy is needed as this method is responsible for |
| 148 | // memory cleanup. But this does simplify code. |
mallinath@webrtc.org | 1112c30 | 2013-09-23 20:34:45 +0000 | [diff] [blame] | 149 | const SctpDataMediaChannel::PayloadProtocolIdentifier ppid = |
| 150 | static_cast<SctpDataMediaChannel::PayloadProtocolIdentifier>( |
| 151 | talk_base::HostToNetwork32(rcv.rcv_ppid)); |
| 152 | cricket::DataMessageType type = cricket::DMT_NONE; |
| 153 | if (!GetDataMediaType(ppid, &type) && !(flags & MSG_NOTIFICATION)) { |
| 154 | // It's neither a notification nor a recognized data packet. Drop it. |
| 155 | LOG(LS_ERROR) << "Received an unknown PPID " << ppid |
| 156 | << " on an SCTP packet. Dropping."; |
| 157 | } else { |
| 158 | SctpInboundPacket* packet = new SctpInboundPacket; |
| 159 | packet->buffer.SetData(data, length); |
| 160 | packet->params.ssrc = rcv.rcv_sid; |
| 161 | packet->params.seq_num = rcv.rcv_ssn; |
| 162 | packet->params.timestamp = rcv.rcv_tsn; |
| 163 | packet->params.type = type; |
| 164 | packet->flags = flags; |
| 165 | channel->worker_thread()->Post(channel, MSG_SCTPINBOUNDPACKET, |
| 166 | talk_base::WrapMessageData(packet)); |
| 167 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 168 | free(data); |
| 169 | return 1; |
| 170 | } |
| 171 | |
| 172 | // Set the initial value of the static SCTP Data Engines reference count. |
| 173 | int SctpDataEngine::usrsctp_engines_count = 0; |
| 174 | |
| 175 | SctpDataEngine::SctpDataEngine() { |
| 176 | if (usrsctp_engines_count == 0) { |
| 177 | // First argument is udp_encapsulation_port, which is not releveant for our |
| 178 | // AF_CONN use of sctp. |
| 179 | usrsctp_init(0, cricket::OnSctpOutboundPacket, debug_sctp_printf); |
| 180 | |
| 181 | // To turn on/off detailed SCTP debugging. You will also need to have the |
| 182 | // SCTP_DEBUG cpp defines flag. |
| 183 | // usrsctp_sysctl_set_sctp_debug_on(SCTP_DEBUG_ALL); |
| 184 | |
| 185 | // TODO(ldixon): Consider turning this on/off. |
| 186 | usrsctp_sysctl_set_sctp_ecn_enable(0); |
| 187 | |
| 188 | // TODO(ldixon): Consider turning this on/off. |
| 189 | // This is not needed right now (we don't do dynamic address changes): |
| 190 | // If SCTP Auto-ASCONF is enabled, the peer is informed automatically |
| 191 | // when a new address is added or removed. This feature is enabled by |
| 192 | // default. |
| 193 | // usrsctp_sysctl_set_sctp_auto_asconf(0); |
| 194 | |
| 195 | // TODO(ldixon): Consider turning this on/off. |
| 196 | // Add a blackhole sysctl. Setting it to 1 results in no ABORTs |
| 197 | // being sent in response to INITs, setting it to 2 results |
| 198 | // in no ABORTs being sent for received OOTB packets. |
| 199 | // This is similar to the TCP sysctl. |
| 200 | // |
| 201 | // See: http://lakerest.net/pipermail/sctp-coders/2012-January/009438.html |
| 202 | // See: http://svnweb.freebsd.org/base?view=revision&revision=229805 |
| 203 | // usrsctp_sysctl_set_sctp_blackhole(2); |
mallinath@webrtc.org | 1112c30 | 2013-09-23 20:34:45 +0000 | [diff] [blame] | 204 | |
| 205 | // Set the number of default outgoing streams. This is the number we'll |
| 206 | // send in the SCTP INIT message. The 'appropriate default' in the |
| 207 | // second paragraph of |
| 208 | // http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-05#section-6.2 |
| 209 | // is cricket::kMaxSctpSid. |
| 210 | usrsctp_sysctl_set_sctp_nr_outgoing_streams_default( |
| 211 | cricket::kMaxSctpSid); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 212 | } |
| 213 | usrsctp_engines_count++; |
| 214 | |
| 215 | // We don't put in a codec because we don't want one offered when we |
| 216 | // use the hybrid data engine. |
| 217 | // codecs_.push_back(cricket::DataCodec( kGoogleSctpDataCodecId, |
| 218 | // kGoogleSctpDataCodecName, 0)); |
| 219 | } |
| 220 | |
| 221 | SctpDataEngine::~SctpDataEngine() { |
| 222 | // TODO(ldixon): There is currently a bug in teardown of usrsctp that blocks |
| 223 | // indefintely if a finish call made too soon after close calls. So teardown |
| 224 | // has been skipped. Once the bug is fixed, retest and enable teardown. |
| 225 | // |
| 226 | // usrsctp_engines_count--; |
| 227 | // LOG(LS_VERBOSE) << "usrsctp_engines_count:" << usrsctp_engines_count; |
| 228 | // if (usrsctp_engines_count == 0) { |
| 229 | // if (usrsctp_finish() != 0) { |
| 230 | // LOG(LS_WARNING) << "usrsctp_finish."; |
| 231 | // } |
| 232 | // } |
| 233 | } |
| 234 | |
| 235 | DataMediaChannel* SctpDataEngine::CreateChannel( |
| 236 | DataChannelType data_channel_type) { |
| 237 | if (data_channel_type != DCT_SCTP) { |
| 238 | return NULL; |
| 239 | } |
| 240 | return new SctpDataMediaChannel(talk_base::Thread::Current()); |
| 241 | } |
| 242 | |
| 243 | SctpDataMediaChannel::SctpDataMediaChannel(talk_base::Thread* thread) |
| 244 | : worker_thread_(thread), |
wu@webrtc.org | 7818752 | 2013-10-07 23:32:02 +0000 | [diff] [blame] | 245 | local_port_(-1), |
| 246 | remote_port_(-1), |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 247 | sock_(NULL), |
| 248 | sending_(false), |
| 249 | receiving_(false), |
| 250 | debug_name_("SctpDataMediaChannel") { |
| 251 | } |
| 252 | |
| 253 | SctpDataMediaChannel::~SctpDataMediaChannel() { |
| 254 | CloseSctpSocket(); |
| 255 | } |
| 256 | |
| 257 | sockaddr_conn SctpDataMediaChannel::GetSctpSockAddr(int port) { |
| 258 | sockaddr_conn sconn = {0}; |
| 259 | sconn.sconn_family = AF_CONN; |
| 260 | #ifdef HAVE_SCONN_LEN |
| 261 | sconn.sconn_len = sizeof(sockaddr_conn); |
| 262 | #endif |
| 263 | // Note: conversion from int to uint16_t happens here. |
| 264 | sconn.sconn_port = talk_base::HostToNetwork16(port); |
| 265 | sconn.sconn_addr = this; |
| 266 | return sconn; |
| 267 | } |
| 268 | |
| 269 | bool SctpDataMediaChannel::OpenSctpSocket() { |
| 270 | if (sock_) { |
| 271 | LOG(LS_VERBOSE) << debug_name_ |
| 272 | << "->Ignoring attempt to re-create existing socket."; |
| 273 | return false; |
| 274 | } |
| 275 | sock_ = usrsctp_socket(AF_CONN, SOCK_STREAM, IPPROTO_SCTP, |
| 276 | cricket::OnSctpInboundPacket, NULL, 0, this); |
| 277 | if (!sock_) { |
| 278 | LOG_ERRNO(LS_ERROR) << debug_name_ << "Failed to create SCTP socket."; |
| 279 | return false; |
| 280 | } |
| 281 | |
| 282 | // Make the socket non-blocking. Connect, close, shutdown etc will not block |
| 283 | // the thread waiting for the socket operation to complete. |
| 284 | if (usrsctp_set_non_blocking(sock_, 1) < 0) { |
| 285 | LOG_ERRNO(LS_ERROR) << debug_name_ << "Failed to set SCTP to non blocking."; |
| 286 | return false; |
| 287 | } |
| 288 | |
| 289 | // This ensures that the usrsctp close call deletes the association. This |
| 290 | // prevents usrsctp from calling OnSctpOutboundPacket with references to |
| 291 | // this class as the address. |
| 292 | linger linger_opt; |
| 293 | linger_opt.l_onoff = 1; |
| 294 | linger_opt.l_linger = 0; |
| 295 | if (usrsctp_setsockopt(sock_, SOL_SOCKET, SO_LINGER, &linger_opt, |
| 296 | sizeof(linger_opt))) { |
| 297 | LOG_ERRNO(LS_ERROR) << debug_name_ << "Failed to set SO_LINGER."; |
| 298 | return false; |
| 299 | } |
| 300 | |
sergeyu@chromium.org | a59696b | 2013-09-13 23:48:58 +0000 | [diff] [blame] | 301 | uint32_t nodelay = 1; |
| 302 | if (usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_NODELAY, &nodelay, |
| 303 | sizeof(nodelay))) { |
| 304 | LOG_ERRNO(LS_ERROR) << debug_name_ << "Failed to set SCTP_NODELAY."; |
| 305 | return false; |
| 306 | } |
| 307 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 308 | // Subscribe to SCTP event notifications. |
| 309 | int event_types[] = {SCTP_ASSOC_CHANGE, |
| 310 | SCTP_PEER_ADDR_CHANGE, |
wu@webrtc.org | d64719d | 2013-08-01 00:00:07 +0000 | [diff] [blame] | 311 | SCTP_SEND_FAILED_EVENT, |
| 312 | SCTP_SENDER_DRY_EVENT}; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 313 | struct sctp_event event = {0}; |
| 314 | event.se_assoc_id = SCTP_ALL_ASSOC; |
| 315 | event.se_on = 1; |
| 316 | for (size_t i = 0; i < ARRAY_SIZE(event_types); i++) { |
| 317 | event.se_type = event_types[i]; |
| 318 | if (usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_EVENT, &event, |
| 319 | sizeof(event)) < 0) { |
| 320 | LOG_ERRNO(LS_ERROR) << debug_name_ << "Failed to set SCTP_EVENT type: " |
| 321 | << event.se_type; |
| 322 | return false; |
| 323 | } |
| 324 | } |
| 325 | |
| 326 | // Register this class as an address for usrsctp. This is used by SCTP to |
| 327 | // direct the packets received (by the created socket) to this class. |
| 328 | usrsctp_register_address(this); |
| 329 | sending_ = true; |
| 330 | return true; |
| 331 | } |
| 332 | |
| 333 | void SctpDataMediaChannel::CloseSctpSocket() { |
| 334 | sending_ = false; |
| 335 | if (sock_) { |
| 336 | // We assume that SO_LINGER option is set to close the association when |
| 337 | // close is called. This means that any pending packets in usrsctp will be |
| 338 | // discarded instead of being sent. |
| 339 | usrsctp_close(sock_); |
| 340 | sock_ = NULL; |
| 341 | usrsctp_deregister_address(this); |
| 342 | } |
| 343 | } |
| 344 | |
| 345 | bool SctpDataMediaChannel::Connect() { |
| 346 | LOG(LS_VERBOSE) << debug_name_ << "->Connect()."; |
wu@webrtc.org | 7818752 | 2013-10-07 23:32:02 +0000 | [diff] [blame] | 347 | if (remote_port_ < 0) { |
| 348 | remote_port_ = kSctpDefaultPort; |
| 349 | } |
| 350 | if (local_port_ < 0) { |
| 351 | local_port_ = kSctpDefaultPort; |
| 352 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 353 | |
| 354 | // If we already have a socket connection, just return. |
| 355 | if (sock_) { |
| 356 | LOG(LS_WARNING) << debug_name_ << "->Connect(): Ignored as socket " |
| 357 | "is already established."; |
| 358 | return true; |
| 359 | } |
| 360 | |
| 361 | // If no socket (it was closed) try to start it again. This can happen when |
| 362 | // the socket we are connecting to closes, does an sctp shutdown handshake, |
| 363 | // or behaves unexpectedly causing us to perform a CloseSctpSocket. |
| 364 | if (!sock_ && !OpenSctpSocket()) { |
| 365 | return false; |
| 366 | } |
| 367 | |
| 368 | // Note: conversion from int to uint16_t happens on assignment. |
| 369 | sockaddr_conn local_sconn = GetSctpSockAddr(local_port_); |
| 370 | if (usrsctp_bind(sock_, reinterpret_cast<sockaddr *>(&local_sconn), |
| 371 | sizeof(local_sconn)) < 0) { |
| 372 | LOG_ERRNO(LS_ERROR) << debug_name_ << "->Connect(): " |
| 373 | << ("Failed usrsctp_bind"); |
| 374 | CloseSctpSocket(); |
| 375 | return false; |
| 376 | } |
| 377 | |
| 378 | // Note: conversion from int to uint16_t happens on assignment. |
| 379 | sockaddr_conn remote_sconn = GetSctpSockAddr(remote_port_); |
| 380 | int connect_result = usrsctp_connect( |
| 381 | sock_, reinterpret_cast<sockaddr *>(&remote_sconn), sizeof(remote_sconn)); |
henrike@webrtc.org | 28654cb | 2013-07-22 21:07:49 +0000 | [diff] [blame] | 382 | if (connect_result < 0 && errno != SCTP_EINPROGRESS) { |
| 383 | LOG_ERRNO(LS_ERROR) << debug_name_ << "Failed usrsctp_connect. got errno=" |
| 384 | << errno << ", but wanted " << SCTP_EINPROGRESS; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 385 | CloseSctpSocket(); |
| 386 | return false; |
| 387 | } |
| 388 | return true; |
| 389 | } |
| 390 | |
| 391 | void SctpDataMediaChannel::Disconnect() { |
| 392 | // TODO(ldixon): Consider calling |usrsctp_shutdown(sock_, ...)| to do a |
| 393 | // shutdown handshake and remove the association. |
| 394 | CloseSctpSocket(); |
| 395 | } |
| 396 | |
| 397 | bool SctpDataMediaChannel::SetSend(bool send) { |
| 398 | if (!sending_ && send) { |
| 399 | return Connect(); |
| 400 | } |
| 401 | if (sending_ && !send) { |
| 402 | Disconnect(); |
| 403 | } |
| 404 | return true; |
| 405 | } |
| 406 | |
| 407 | bool SctpDataMediaChannel::SetReceive(bool receive) { |
| 408 | receiving_ = receive; |
| 409 | return true; |
| 410 | } |
| 411 | |
| 412 | bool SctpDataMediaChannel::AddSendStream(const StreamParams& stream) { |
| 413 | if (!stream.has_ssrcs()) { |
| 414 | return false; |
| 415 | } |
| 416 | |
| 417 | StreamParams found_stream; |
wu@webrtc.org | 91053e7 | 2013-08-10 07:18:04 +0000 | [diff] [blame] | 418 | // TODO(lally): Consider keeping this sorted. |
| 419 | if (GetStreamBySsrc(streams_, stream.first_ssrc(), &found_stream)) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 420 | LOG(LS_WARNING) << debug_name_ << "->AddSendStream(...): " |
| 421 | << "Not adding data send stream '" << stream.id |
| 422 | << "' with ssrc=" << stream.first_ssrc() |
| 423 | << " because stream already exists."; |
| 424 | return false; |
| 425 | } |
| 426 | |
wu@webrtc.org | 91053e7 | 2013-08-10 07:18:04 +0000 | [diff] [blame] | 427 | streams_.push_back(stream); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 428 | return true; |
| 429 | } |
| 430 | |
| 431 | bool SctpDataMediaChannel::RemoveSendStream(uint32 ssrc) { |
| 432 | StreamParams found_stream; |
wu@webrtc.org | 91053e7 | 2013-08-10 07:18:04 +0000 | [diff] [blame] | 433 | if (!GetStreamBySsrc(streams_, ssrc, &found_stream)) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 434 | return false; |
| 435 | } |
| 436 | |
wu@webrtc.org | 91053e7 | 2013-08-10 07:18:04 +0000 | [diff] [blame] | 437 | RemoveStreamBySsrc(&streams_, ssrc); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 438 | return true; |
| 439 | } |
| 440 | |
| 441 | // Note: expects exactly one ssrc. If none are given, it will fail. If more |
| 442 | // than one are given, it will use the first. |
| 443 | bool SctpDataMediaChannel::AddRecvStream(const StreamParams& stream) { |
| 444 | if (!stream.has_ssrcs()) { |
| 445 | return false; |
| 446 | } |
| 447 | |
| 448 | StreamParams found_stream; |
wu@webrtc.org | 91053e7 | 2013-08-10 07:18:04 +0000 | [diff] [blame] | 449 | if (GetStreamBySsrc(streams_, stream.first_ssrc(), &found_stream)) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 450 | LOG(LS_WARNING) << debug_name_ << "->AddRecvStream(...): " |
| 451 | << "Not adding data recv stream '" << stream.id |
| 452 | << "' with ssrc=" << stream.first_ssrc() |
| 453 | << " because stream already exists."; |
| 454 | return false; |
| 455 | } |
| 456 | |
wu@webrtc.org | 91053e7 | 2013-08-10 07:18:04 +0000 | [diff] [blame] | 457 | streams_.push_back(stream); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 458 | LOG(LS_VERBOSE) << debug_name_ << "->AddRecvStream(...): " |
| 459 | << "Added data recv stream '" << stream.id |
| 460 | << "' with ssrc=" << stream.first_ssrc(); |
| 461 | return true; |
| 462 | } |
| 463 | |
| 464 | bool SctpDataMediaChannel::RemoveRecvStream(uint32 ssrc) { |
wu@webrtc.org | 91053e7 | 2013-08-10 07:18:04 +0000 | [diff] [blame] | 465 | RemoveStreamBySsrc(&streams_, ssrc); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 466 | return true; |
| 467 | } |
| 468 | |
| 469 | bool SctpDataMediaChannel::SendData( |
| 470 | const SendDataParams& params, |
| 471 | const talk_base::Buffer& payload, |
| 472 | SendDataResult* result) { |
| 473 | if (result) { |
mallinath@webrtc.org | 1112c30 | 2013-09-23 20:34:45 +0000 | [diff] [blame] | 474 | // Preset |result| to assume an error. If SendData succeeds, we'll |
| 475 | // overwrite |*result| once more at the end. |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 476 | *result = SDR_ERROR; |
| 477 | } |
| 478 | |
| 479 | if (!sending_) { |
| 480 | LOG(LS_WARNING) << debug_name_ << "->SendData(...): " |
| 481 | << "Not sending packet with ssrc=" << params.ssrc |
| 482 | << " len=" << payload.length() << " before SetSend(true)."; |
| 483 | return false; |
| 484 | } |
| 485 | |
| 486 | StreamParams found_stream; |
wu@webrtc.org | 91053e7 | 2013-08-10 07:18:04 +0000 | [diff] [blame] | 487 | if (params.type != cricket::DMT_CONTROL && |
| 488 | !GetStreamBySsrc(streams_, params.ssrc, &found_stream)) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 489 | LOG(LS_WARNING) << debug_name_ << "->SendData(...): " |
| 490 | << "Not sending data because ssrc is unknown: " |
| 491 | << params.ssrc; |
| 492 | return false; |
| 493 | } |
| 494 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 495 | // |
| 496 | // Send data using SCTP. |
mallinath@webrtc.org | 1112c30 | 2013-09-23 20:34:45 +0000 | [diff] [blame] | 497 | ssize_t send_res = 0; // result from usrsctp_sendv. |
| 498 | struct sctp_sendv_spa spa = {0}; |
| 499 | spa.sendv_flags |= SCTP_SEND_SNDINFO_VALID; |
| 500 | spa.sendv_sndinfo.snd_sid = params.ssrc; |
| 501 | spa.sendv_sndinfo.snd_ppid = talk_base::HostToNetwork32( |
| 502 | GetPpid(params.type)); |
| 503 | |
| 504 | // Ordered implies reliable. |
| 505 | if (!params.ordered) { |
| 506 | spa.sendv_sndinfo.snd_flags |= SCTP_UNORDERED; |
| 507 | if (params.max_rtx_count >= 0 || params.max_rtx_ms == 0) { |
| 508 | spa.sendv_flags |= SCTP_SEND_PRINFO_VALID; |
| 509 | spa.sendv_prinfo.pr_policy = SCTP_PR_SCTP_RTX; |
| 510 | spa.sendv_prinfo.pr_value = params.max_rtx_count; |
| 511 | } else { |
| 512 | spa.sendv_flags |= SCTP_SEND_PRINFO_VALID; |
| 513 | spa.sendv_prinfo.pr_policy = SCTP_PR_SCTP_TTL; |
| 514 | spa.sendv_prinfo.pr_value = params.max_rtx_ms; |
| 515 | } |
| 516 | } |
| 517 | |
| 518 | // We don't fragment. |
| 519 | send_res = usrsctp_sendv(sock_, payload.data(), |
| 520 | static_cast<size_t>(payload.length()), |
| 521 | NULL, 0, &spa, |
| 522 | static_cast<socklen_t>(sizeof(spa)), |
| 523 | SCTP_SENDV_SPA, 0); |
| 524 | if (send_res < 0) { |
wu@webrtc.org | d64719d | 2013-08-01 00:00:07 +0000 | [diff] [blame] | 525 | if (errno == EWOULDBLOCK) { |
| 526 | *result = SDR_BLOCK; |
| 527 | LOG(LS_INFO) << debug_name_ << "->SendData(...): EWOULDBLOCK returned"; |
| 528 | } else { |
| 529 | LOG_ERRNO(LS_ERROR) << "ERROR:" << debug_name_ |
| 530 | << "->SendData(...): " |
| 531 | << " usrsctp_sendv: "; |
| 532 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 533 | return false; |
| 534 | } |
| 535 | if (result) { |
mallinath@webrtc.org | 1112c30 | 2013-09-23 20:34:45 +0000 | [diff] [blame] | 536 | // Only way out now is success. |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 537 | *result = SDR_SUCCESS; |
| 538 | } |
| 539 | return true; |
| 540 | } |
| 541 | |
| 542 | // Called by network interface when a packet has been received. |
| 543 | void SctpDataMediaChannel::OnPacketReceived(talk_base::Buffer* packet) { |
mallinath@webrtc.org | 1112c30 | 2013-09-23 20:34:45 +0000 | [diff] [blame] | 544 | LOG(LS_VERBOSE) << debug_name_ << "->OnPacketReceived(...): " << " length=" |
| 545 | << packet->length() << ", sending: " << sending_; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 546 | // Only give receiving packets to usrsctp after if connected. This enables two |
| 547 | // peers to each make a connect call, but for them not to receive an INIT |
| 548 | // packet before they have called connect; least the last receiver of the INIT |
| 549 | // packet will have called connect, and a connection will be established. |
| 550 | if (sending_) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 551 | // Pass received packet to SCTP stack. Once processed by usrsctp, the data |
| 552 | // will be will be given to the global OnSctpInboundData, and then, |
| 553 | // marshalled by a Post and handled with OnMessage. |
| 554 | usrsctp_conninput(this, packet->data(), packet->length(), 0); |
| 555 | } else { |
| 556 | // TODO(ldixon): Consider caching the packet for very slightly better |
| 557 | // reliability. |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 558 | } |
| 559 | } |
| 560 | |
| 561 | void SctpDataMediaChannel::OnInboundPacketFromSctpToChannel( |
| 562 | SctpInboundPacket* packet) { |
| 563 | LOG(LS_VERBOSE) << debug_name_ << "->OnInboundPacketFromSctpToChannel(...): " |
| 564 | << "Received SCTP data:" |
| 565 | << " ssrc=" << packet->params.ssrc |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 566 | << " notification: " << (packet->flags & MSG_NOTIFICATION) |
mallinath@webrtc.org | 1112c30 | 2013-09-23 20:34:45 +0000 | [diff] [blame] | 567 | << " length=" << packet->buffer.length(); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 568 | // Sending a packet with data == NULL (no data) is SCTPs "close the |
| 569 | // connection" message. This sets sock_ = NULL; |
| 570 | if (!packet->buffer.length() || !packet->buffer.data()) { |
| 571 | LOG(LS_INFO) << debug_name_ << "->OnInboundPacketFromSctpToChannel(...): " |
| 572 | "No data, closing."; |
| 573 | return; |
| 574 | } |
| 575 | if (packet->flags & MSG_NOTIFICATION) { |
| 576 | OnNotificationFromSctp(&packet->buffer); |
| 577 | } else { |
| 578 | OnDataFromSctpToChannel(packet->params, &packet->buffer); |
| 579 | } |
| 580 | } |
| 581 | |
| 582 | void SctpDataMediaChannel::OnDataFromSctpToChannel( |
| 583 | const ReceiveDataParams& params, talk_base::Buffer* buffer) { |
| 584 | StreamParams found_stream; |
wu@webrtc.org | 91053e7 | 2013-08-10 07:18:04 +0000 | [diff] [blame] | 585 | if (!GetStreamBySsrc(streams_, params.ssrc, &found_stream)) { |
| 586 | if (params.type == DMT_CONTROL) { |
| 587 | SignalDataReceived(params, buffer->data(), buffer->length()); |
| 588 | } else { |
| 589 | LOG(LS_WARNING) << debug_name_ << "->OnDataFromSctpToChannel(...): " |
| 590 | << "Received packet for unknown ssrc: " << params.ssrc; |
| 591 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 592 | return; |
| 593 | } |
| 594 | |
| 595 | if (receiving_) { |
| 596 | LOG(LS_VERBOSE) << debug_name_ << "->OnDataFromSctpToChannel(...): " |
| 597 | << "Posting with length: " << buffer->length(); |
| 598 | SignalDataReceived(params, buffer->data(), buffer->length()); |
| 599 | } else { |
| 600 | LOG(LS_WARNING) << debug_name_ << "->OnDataFromSctpToChannel(...): " |
| 601 | << "Not receiving packet with sid=" << params.ssrc |
| 602 | << " len=" << buffer->length() |
| 603 | << " before SetReceive(true)."; |
| 604 | } |
| 605 | } |
| 606 | |
wu@webrtc.org | d64719d | 2013-08-01 00:00:07 +0000 | [diff] [blame] | 607 | void SctpDataMediaChannel::OnNotificationFromSctp(talk_base::Buffer* buffer) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 608 | const sctp_notification& notification = |
| 609 | reinterpret_cast<const sctp_notification&>(*buffer->data()); |
| 610 | ASSERT(notification.sn_header.sn_length == buffer->length()); |
| 611 | |
| 612 | // TODO(ldixon): handle notifications appropriately. |
| 613 | switch (notification.sn_header.sn_type) { |
| 614 | case SCTP_ASSOC_CHANGE: |
| 615 | LOG(LS_VERBOSE) << "SCTP_ASSOC_CHANGE"; |
| 616 | OnNotificationAssocChange(notification.sn_assoc_change); |
| 617 | break; |
| 618 | case SCTP_REMOTE_ERROR: |
| 619 | LOG(LS_INFO) << "SCTP_REMOTE_ERROR"; |
| 620 | break; |
| 621 | case SCTP_SHUTDOWN_EVENT: |
| 622 | LOG(LS_INFO) << "SCTP_SHUTDOWN_EVENT"; |
| 623 | break; |
| 624 | case SCTP_ADAPTATION_INDICATION: |
| 625 | LOG(LS_INFO) << "SCTP_ADAPTATION_INIDICATION"; |
| 626 | break; |
| 627 | case SCTP_PARTIAL_DELIVERY_EVENT: |
| 628 | LOG(LS_INFO) << "SCTP_PARTIAL_DELIVERY_EVENT"; |
| 629 | break; |
| 630 | case SCTP_AUTHENTICATION_EVENT: |
| 631 | LOG(LS_INFO) << "SCTP_AUTHENTICATION_EVENT"; |
| 632 | break; |
| 633 | case SCTP_SENDER_DRY_EVENT: |
| 634 | LOG(LS_INFO) << "SCTP_SENDER_DRY_EVENT"; |
wu@webrtc.org | d64719d | 2013-08-01 00:00:07 +0000 | [diff] [blame] | 635 | SignalReadyToSend(true); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 636 | break; |
| 637 | // TODO(ldixon): Unblock after congestion. |
| 638 | case SCTP_NOTIFICATIONS_STOPPED_EVENT: |
| 639 | LOG(LS_INFO) << "SCTP_NOTIFICATIONS_STOPPED_EVENT"; |
| 640 | break; |
| 641 | case SCTP_SEND_FAILED_EVENT: |
| 642 | LOG(LS_INFO) << "SCTP_SEND_FAILED_EVENT"; |
| 643 | break; |
| 644 | case SCTP_STREAM_RESET_EVENT: |
| 645 | LOG(LS_INFO) << "SCTP_STREAM_RESET_EVENT"; |
| 646 | // TODO(ldixon): Notify up to channel that stream resent has happened, |
| 647 | // and write unit test for this case. |
| 648 | break; |
| 649 | case SCTP_ASSOC_RESET_EVENT: |
| 650 | LOG(LS_INFO) << "SCTP_ASSOC_RESET_EVENT"; |
| 651 | break; |
| 652 | case SCTP_STREAM_CHANGE_EVENT: |
| 653 | LOG(LS_INFO) << "SCTP_STREAM_CHANGE_EVENT"; |
| 654 | break; |
| 655 | default: |
| 656 | LOG(LS_WARNING) << "Unknown SCTP event: " |
| 657 | << notification.sn_header.sn_type; |
| 658 | break; |
| 659 | } |
| 660 | } |
| 661 | |
| 662 | void SctpDataMediaChannel::OnNotificationAssocChange( |
| 663 | const sctp_assoc_change& change) { |
| 664 | switch (change.sac_state) { |
| 665 | case SCTP_COMM_UP: |
| 666 | LOG(LS_VERBOSE) << "Association change SCTP_COMM_UP"; |
| 667 | break; |
| 668 | case SCTP_COMM_LOST: |
| 669 | LOG(LS_INFO) << "Association change SCTP_COMM_LOST"; |
| 670 | break; |
| 671 | case SCTP_RESTART: |
| 672 | LOG(LS_INFO) << "Association change SCTP_RESTART"; |
| 673 | break; |
| 674 | case SCTP_SHUTDOWN_COMP: |
| 675 | LOG(LS_INFO) << "Association change SCTP_SHUTDOWN_COMP"; |
| 676 | break; |
| 677 | case SCTP_CANT_STR_ASSOC: |
| 678 | LOG(LS_INFO) << "Association change SCTP_CANT_STR_ASSOC"; |
| 679 | break; |
| 680 | default: |
| 681 | LOG(LS_INFO) << "Association change UNKNOWN"; |
| 682 | break; |
| 683 | } |
| 684 | } |
| 685 | |
wu@webrtc.org | 7818752 | 2013-10-07 23:32:02 +0000 | [diff] [blame] | 686 | // Puts the specified |param| from the codec identified by |id| into |dest| |
| 687 | // and returns true. Or returns false if it wasn't there, leaving |dest| |
| 688 | // untouched. |
| 689 | static bool GetCodecIntParameter(const std::vector<DataCodec>& codecs, |
| 690 | int id, const std::string& name, |
| 691 | const std::string& param, int* dest) { |
| 692 | std::string value; |
| 693 | Codec match_pattern; |
| 694 | match_pattern.id = id; |
| 695 | match_pattern.name = name; |
| 696 | for (size_t i = 0; i < codecs.size(); ++i) { |
| 697 | if (codecs[i].Matches(match_pattern)) { |
| 698 | if (codecs[i].GetParam(param, &value)) { |
| 699 | *dest = talk_base::FromString<int>(value); |
| 700 | return true; |
| 701 | } |
| 702 | } |
| 703 | } |
| 704 | return false; |
| 705 | } |
| 706 | |
| 707 | bool SctpDataMediaChannel::SetSendCodecs(const std::vector<DataCodec>& codecs) { |
| 708 | return GetCodecIntParameter( |
| 709 | codecs, kGoogleSctpDataCodecId, kGoogleSctpDataCodecName, kCodecParamPort, |
| 710 | &remote_port_); |
| 711 | } |
| 712 | |
| 713 | bool SctpDataMediaChannel::SetRecvCodecs(const std::vector<DataCodec>& codecs) { |
| 714 | return GetCodecIntParameter( |
| 715 | codecs, kGoogleSctpDataCodecId, kGoogleSctpDataCodecName, kCodecParamPort, |
| 716 | &local_port_); |
| 717 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 718 | |
| 719 | void SctpDataMediaChannel::OnPacketFromSctpToNetwork( |
| 720 | talk_base::Buffer* buffer) { |
| 721 | if (buffer->length() > kSctpMtu) { |
| 722 | LOG(LS_ERROR) << debug_name_ << "->OnPacketFromSctpToNetwork(...): " |
| 723 | << "SCTP seems to have made a poacket that is bigger " |
| 724 | "than its official MTU."; |
| 725 | } |
henrike@webrtc.org | 1e09a71 | 2013-07-26 19:17:59 +0000 | [diff] [blame] | 726 | MediaChannel::SendPacket(buffer); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 727 | } |
| 728 | |
| 729 | void SctpDataMediaChannel::OnMessage(talk_base::Message* msg) { |
| 730 | switch (msg->message_id) { |
| 731 | case MSG_SCTPINBOUNDPACKET: { |
| 732 | SctpInboundPacket* packet = |
| 733 | static_cast<talk_base::TypedMessageData<SctpInboundPacket*>*>( |
| 734 | msg->pdata)->data(); |
| 735 | OnInboundPacketFromSctpToChannel(packet); |
| 736 | delete packet; |
| 737 | break; |
| 738 | } |
| 739 | case MSG_SCTPOUTBOUNDPACKET: { |
| 740 | talk_base::Buffer* buffer = |
| 741 | static_cast<talk_base::TypedMessageData<talk_base::Buffer*>*>( |
| 742 | msg->pdata)->data(); |
| 743 | OnPacketFromSctpToNetwork(buffer); |
| 744 | delete buffer; |
| 745 | break; |
| 746 | } |
| 747 | } |
| 748 | } |
| 749 | |
| 750 | } // namespace cricket |