content/browser/renderer_host/p2p/socket_host.cc

   1 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
   2 // Use of this source code is governed by a BSD-style license that can be
   3 // found in the LICENSE file.
   4
   5 #include "content/browser/renderer_host/p2p/socket_host.h"
   6
   7 #include "base/metrics/histogram.h"
   8 #include "base/sys_byteorder.h"
   9 #include "content/browser/renderer_host/p2p/socket_host_tcp.h"
  10 #include "content/browser/renderer_host/p2p/socket_host_tcp_server.h"
  11 #include "content/browser/renderer_host/p2p/socket_host_udp.h"
  12 #include "content/browser/renderer_host/render_process_host_impl.h"
  13 #include "content/public/browser/browser_thread.h"
  14 #include "crypto/hmac.h"
  15 #include "third_party/webrtc/base/asyncpacketsocket.h"
  16 #include "third_party/webrtc/base/byteorder.h"
  17 #include "third_party/webrtc/base/messagedigest.h"
  18 #include "third_party/webrtc/p2p/base/stun.h"
  19
  20 namespace {
  21
  22 const uint32 kStunMagicCookie = 0x2112A442;
  23 const size_t kMinRtpHeaderLength = 12;
  24 const size_t kMinRtcpHeaderLength = 8;
  25 const size_t kRtpExtensionHeaderLength = 4;
  26 const size_t kDtlsRecordHeaderLength = 13;
  27 const size_t kTurnChannelHeaderLength = 4;
  28 const size_t kAbsSendTimeExtensionLength = 3;
  29 const size_t kOneByteHeaderLength = 1;
  30 const size_t kMaxRtpPacketLength = 2048;
  31
  32 // Fake auth tag written by the render process if external authentication is
  33 // enabled. HMAC in packet will be compared against this value before updating
  34 // packet with actual HMAC value.
  35 static const unsigned char kFakeAuthTag[10] = {
  36     0xba, 0xdd, 0xba, 0xdd, 0xba, 0xdd, 0xba, 0xdd, 0xba, 0xdd
  37 };
  38
  39 bool IsTurnChannelData(const char* data, size_t length) {
  40   return length >= kTurnChannelHeaderLength && ((*data & 0xC0) == 0x40);
  41 }
  42
  43 bool IsDtlsPacket(const char* data, size_t length) {
  44   const uint8* u = reinterpret_cast<const uint8*>(data);
  45   return (length >= kDtlsRecordHeaderLength && (u[0] > 19 && u[0] < 64));
  46 }
  47
  48 bool IsRtcpPacket(const char* data, size_t length) {
  49   if (length < kMinRtcpHeaderLength) {
  50     return false;
  51   }
  52
  53   int type = (static_cast<uint8>(data[1]) & 0x7F);
  54   return (type >= 64 && type < 96);
  55 }
  56
  57 bool IsTurnSendIndicationPacket(const char* data, size_t length) {
  58   if (length < content::P2PSocketHost::kStunHeaderSize) {
  59     return false;
  60   }
  61
  62   uint16 type = rtc::GetBE16(data);
  63   return (type == cricket::TURN_SEND_INDICATION);
  64 }
  65
  66 bool IsRtpPacket(const char* data, size_t length) {
  67   return (length >= kMinRtpHeaderLength) && ((*data & 0xC0) == 0x80);
  68 }
  69
  70 // Verifies rtp header and message length.
  71 bool ValidateRtpHeader(const char* rtp, size_t length, size_t* header_length) {
  72   if (header_length) {
  73     *header_length = 0;
  74   }
  75
  76   if (length < kMinRtpHeaderLength) {
  77     return false;
  78   }
  79
  80   size_t cc_count = rtp[0] & 0x0F;
  81   size_t header_length_without_extension = kMinRtpHeaderLength + 4 * cc_count;
  82   if (header_length_without_extension > length) {
  83     return false;
  84   }
  85
  86   // If extension bit is not set, we are done with header processing, as input
  87   // length is verified above.
  88   if (!(rtp[0] & 0x10)) {
  89     if (header_length)
  90       *header_length = header_length_without_extension;
  91
  92     return true;
  93   }
  94
  95   rtp += header_length_without_extension;
  96
  97   if (header_length_without_extension + kRtpExtensionHeaderLength > length) {
  98     return false;
  99   }
 100
 101   // Getting extension profile length.
 102   // Length is in 32 bit words.
 103   uint16 extension_length_in_32bits = rtc::GetBE16(rtp + 2);
 104   size_t extension_length = extension_length_in_32bits * 4;
 105
 106   size_t rtp_header_length = extension_length +
 107                              header_length_without_extension +
 108                              kRtpExtensionHeaderLength;
 109
 110   // Verify input length against total header size.
 111   if (rtp_header_length > length) {
 112     return false;
 113   }
 114
 115   if (header_length) {
 116     *header_length = rtp_header_length;
 117   }
 118   return true;
 119 }
 120
 121 void UpdateAbsSendTimeExtensionValue(char* extension_data,
 122                                      size_t length,
 123                                      uint32 abs_send_time) {
 124   // Absolute send time in RTP streams.
 125   //
 126   // The absolute send time is signaled to the receiver in-band using the
 127   // general mechanism for RTP header extensions [RFC5285]. The payload
 128   // of this extension (the transmitted value) is a 24-bit unsigned integer
 129   // containing the sender's current time in seconds as a fixed point number
 130   // with 18 bits fractional part.
 131   //
 132   // The form of the absolute send time extension block:
 133   //
 134   //    0                   1                   2                   3
 135   //    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
 136   //   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 137   //   |  ID   | len=2 |              absolute send time               |
 138   //   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 139   if (length != kAbsSendTimeExtensionLength) {
 140     NOTREACHED();
 141     return;
 142   }
 143
 144   // Now() has resolution ~1-15ms, using HighResNow(). But it is warned not to
 145   // use it unless necessary, as it is expensive than Now().
 146   uint32 now_second = abs_send_time;
 147   if (!now_second) {
 148     uint64 now_us =
 149         (base::TimeTicks::HighResNow() - base::TimeTicks()).InMicroseconds();
 150     // Convert second to 24-bit unsigned with 18 bit fractional part
 151     now_second =
 152         ((now_us << 18) / base::Time::kMicrosecondsPerSecond) & 0x00FFFFFF;
 153   }
 154   // TODO(mallinath) - Add SetBE24 to byteorder.h in libjingle.
 155   extension_data[0] = static_cast<uint8>(now_second >> 16);
 156   extension_data[1] = static_cast<uint8>(now_second >> 8);
 157   extension_data[2] = static_cast<uint8>(now_second);
 158 }
 159
 160 // Assumes |length| is actual packet length + tag length. Updates HMAC at end of
 161 // the RTP packet.
 162 void UpdateRtpAuthTag(char* rtp,
 163                       size_t length,
 164                       const rtc::PacketOptions& options) {
 165   // If there is no key, return.
 166   if (options.packet_time_params.srtp_auth_key.empty()) {
 167     return;
 168   }
 169
 170   size_t tag_length = options.packet_time_params.srtp_auth_tag_len;
 171
 172   // ROC (rollover counter) is at the beginning of the auth tag.
 173   const size_t kRocLength = 4;
 174   if (tag_length < kRocLength || tag_length > length) {
 175     NOTREACHED();
 176     return;
 177   }
 178
 179   crypto::HMAC hmac(crypto::HMAC::SHA1);
 180   if (!hmac.Init(reinterpret_cast<const unsigned char*>(
 181         &options.packet_time_params.srtp_auth_key[0]),
 182         options.packet_time_params.srtp_auth_key.size())) {
 183     NOTREACHED();
 184     return;
 185   }
 186
 187   if (tag_length > hmac.DigestLength()) {
 188     NOTREACHED();
 189     return;
 190   }
 191
 192   char* auth_tag = rtp + (length - tag_length);
 193
 194   // We should have a fake HMAC value @ auth_tag.
 195   DCHECK_EQ(0, memcmp(auth_tag, kFakeAuthTag, tag_length));
 196
 197   // Copy ROC after end of rtp packet.
 198   memcpy(auth_tag, &options.packet_time_params.srtp_packet_index, kRocLength);
 199   // Authentication of a RTP packet will have RTP packet + ROC size.
 200   int auth_required_length = length - tag_length + kRocLength;
 201
 202   unsigned char output[64];
 203   if (!hmac.Sign(base::StringPiece(rtp, auth_required_length),
 204                  output, sizeof(output))) {
 205     NOTREACHED();
 206     return;
 207   }
 208   // Copy HMAC from output to packet. This is required as auth tag length
 209   // may not be equal to the actual HMAC length.
 210   memcpy(auth_tag, output, tag_length);
 211 }
 212
 213 }  // namespace
 214
 215 namespace content {
 216
 217 namespace packet_processing_helpers {
 218
 219 bool ApplyPacketOptions(char* data,
 220                         size_t length,
 221                         const rtc::PacketOptions& options,
 222                         uint32 abs_send_time) {
 223   DCHECK(data != NULL);
 224   DCHECK(length > 0);
 225   // if there is no valid |rtp_sendtime_extension_id| and |srtp_auth_key| in
 226   // PacketOptions, nothing to be updated in this packet.
 227   if (options.packet_time_params.rtp_sendtime_extension_id == -1 &&
 228       options.packet_time_params.srtp_auth_key.empty()) {
 229     return true;
 230   }
 231
 232   DCHECK(!IsDtlsPacket(data, length));
 233   DCHECK(!IsRtcpPacket(data, length));
 234
 235   // If there is a srtp auth key present then packet must be a RTP packet.
 236   // RTP packet may have been wrapped in a TURN Channel Data or
 237   // TURN send indication.
 238   size_t rtp_start_pos;
 239   size_t rtp_length;
 240   if (!GetRtpPacketStartPositionAndLength(
 241       data, length, &rtp_start_pos, &rtp_length)) {
 242     // This method should never return false.
 243     NOTREACHED();
 244     return false;
 245   }
 246
 247   // Skip to rtp packet.
 248   char* start = data + rtp_start_pos;
 249   // If packet option has non default value (-1) for sendtime extension id,
 250   // then we should parse the rtp packet to update the timestamp. Otherwise
 251   // just calculate HMAC and update packet with it.
 252   if (options.packet_time_params.rtp_sendtime_extension_id != -1) {
 253     UpdateRtpAbsSendTimeExtension(
 254         start,
 255         rtp_length,
 256         options.packet_time_params.rtp_sendtime_extension_id,
 257         abs_send_time);
 258   }
 259
 260   UpdateRtpAuthTag(start, rtp_length, options);
 261   return true;
 262 }
 263
 264 bool GetRtpPacketStartPositionAndLength(const char* packet,
 265                                         size_t length,
 266                                         size_t* rtp_start_pos,
 267                                         size_t* rtp_packet_length) {
 268   if (length < kMinRtpHeaderLength || length > kMaxRtpPacketLength) {
 269     return false;
 270   }
 271
 272   size_t rtp_begin;
 273   size_t rtp_length = 0;
 274   if (IsTurnChannelData(packet, length)) {
 275     // Turn Channel Message header format.
 276     //   0                   1                   2                   3
 277     //  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
 278     // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 279     // |         Channel Number        |            Length             |
 280     // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 281     // |                                                               |
 282     // /                       Application Data                        /
 283     // /                                                               /
 284     // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 285     rtp_begin = kTurnChannelHeaderLength;
 286     rtp_length = rtc::GetBE16(&packet[2]);
 287     if (length < rtp_length + kTurnChannelHeaderLength) {
 288       return false;
 289     }
 290   } else if (IsTurnSendIndicationPacket(packet, length)) {
 291     // Validate STUN message length.
 292     const size_t stun_message_length = rtc::GetBE16(&packet[2]);
 293     if (stun_message_length + P2PSocketHost::kStunHeaderSize != length) {
 294       return false;
 295     }
 296
 297     // First skip mandatory stun header which is of 20 bytes.
 298     rtp_begin = P2PSocketHost::kStunHeaderSize;
 299     // Loop through STUN attributes until we find STUN DATA attribute.
 300     bool data_attr_present = false;
 301     while (rtp_begin < length) {
 302       // Keep reading STUN attributes until we hit DATA attribute.
 303       // Attribute will be a TLV structure.
 304       // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 305       // |         Type                  |            Length             |
 306       // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 307       // |                         Value (variable)                ....
 308       // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 309       // The value in the length field MUST contain the length of the Value
 310       // part of the attribute, prior to padding, measured in bytes.  Since
 311       // STUN aligns attributes on 32-bit boundaries, attributes whose content
 312       // is not a multiple of 4 bytes are padded with 1, 2, or 3 bytes of
 313       // padding so that its value contains a multiple of 4 bytes.  The
 314       // padding bits are ignored, and may be any value.
 315       uint16 attr_type, attr_length;
 316       const int kAttrHeaderLength = sizeof(attr_type) + sizeof(attr_length);
 317
 318       if (length < rtp_begin + kAttrHeaderLength) {
 319         return false;
 320       }
 321
 322       // Getting attribute type and length.
 323       attr_type = rtc::GetBE16(&packet[rtp_begin]);
 324       attr_length = rtc::GetBE16(
 325           &packet[rtp_begin + sizeof(attr_type)]);
 326
 327       rtp_begin += kAttrHeaderLength;  // Skip STUN_DATA_ATTR header.
 328
 329       // Checking for bogus attribute length.
 330       if (length < rtp_begin + attr_length) {
 331         return false;
 332       }
 333
 334       if (attr_type != cricket::STUN_ATTR_DATA) {
 335         rtp_begin += attr_length;
 336         if ((attr_length % 4) != 0) {
 337           rtp_begin += (4 - (attr_length % 4));
 338         }
 339         continue;
 340       }
 341
 342       data_attr_present = true;
 343       rtp_length = attr_length;
 344
 345       // We found STUN_DATA_ATTR. We can skip parsing rest of the packet.
 346       break;
 347     }
 348
 349     if (!data_attr_present) {
 350       // There is no data attribute present in the message. We can't do anything
 351       // with the message.
 352       return false;
 353     }
 354
 355   } else {
 356     // This is a raw RTP packet.
 357     rtp_begin = 0;
 358     rtp_length = length;
 359   }
 360
 361   // Making sure we have a valid RTP packet at the end.
 362   if (IsRtpPacket(packet + rtp_begin, rtp_length) &&
 363       ValidateRtpHeader(packet + rtp_begin, rtp_length, NULL)) {
 364     *rtp_start_pos = rtp_begin;
 365     *rtp_packet_length = rtp_length;
 366     return true;
 367   }
 368   return false;
 369 }
 370
 371 // ValidateRtpHeader must be called before this method to make sure, we have
 372 // a sane rtp packet.
 373 bool UpdateRtpAbsSendTimeExtension(char* rtp,
 374                                    size_t length,
 375                                    int extension_id,
 376                                    uint32 abs_send_time) {
 377   //  0                   1                   2                   3
 378   //  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
 379   // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 380   // |V=2|P|X|  CC   |M|     PT      |       sequence number         |
 381   // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 382   // |                           timestamp                           |
 383   // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 384   // |           synchronization source (SSRC) identifier            |
 385   // +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
 386   // |            contributing source (CSRC) identifiers             |
 387   // |                             ....                              |
 388   // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 389
 390   // Return if extension bit is not set.
 391   if (!(rtp[0] & 0x10)) {
 392     return true;
 393   }
 394
 395   size_t cc_count = rtp[0] & 0x0F;
 396   size_t header_length_without_extension = kMinRtpHeaderLength + 4 * cc_count;
 397
 398   rtp += header_length_without_extension;
 399
 400   // Getting extension profile ID and length.
 401   uint16 profile_id = rtc::GetBE16(rtp);
 402   // Length is in 32 bit words.
 403   uint16 extension_length_in_32bits = rtc::GetBE16(rtp + 2);
 404   size_t extension_length = extension_length_in_32bits * 4;
 405
 406   rtp += kRtpExtensionHeaderLength;  // Moving past extension header.
 407
 408   bool found = false;
 409   // WebRTC is using one byte header extension.
 410   // TODO(mallinath) - Handle two byte header extension.
 411   if (profile_id == 0xBEDE) {  // OneByte extension header
 412     //  0
 413     //  0 1 2 3 4 5 6 7
 414     // +-+-+-+-+-+-+-+-+
 415     // |  ID   |length |
 416     // +-+-+-+-+-+-+-+-+
 417
 418     //  0                   1                   2                   3
 419     //  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
 420     // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 421     // |       0xBE    |    0xDE       |           length=3            |
 422     // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 423     // |  ID   | L=0   |     data      |  ID   |  L=1  |   data...
 424     // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 425     //       ...data   |    0 (pad)    |    0 (pad)    |  ID   | L=3   |
 426     // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 427     // |                          data                                 |
 428     // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 429     const char* extension_start = rtp;
 430     const char* extension_end = extension_start + extension_length;
 431
 432     while (rtp < extension_end) {
 433       const int id = (*rtp & 0xF0) >> 4;
 434       const size_t length = (*rtp & 0x0F) + 1;
 435       if (rtp + kOneByteHeaderLength + length > extension_end) {
 436         return false;
 437       }
 438       // The 4-bit length is the number minus one of data bytes of this header
 439       // extension element following the one-byte header.
 440       if (id == extension_id) {
 441         UpdateAbsSendTimeExtensionValue(
 442             rtp + kOneByteHeaderLength, length, abs_send_time);
 443         found = true;
 444         break;
 445       }
 446       rtp += kOneByteHeaderLength + length;
 447       // Counting padding bytes.
 448       while ((rtp < extension_end) && (*rtp == 0)) {
 449         ++rtp;
 450       }
 451     }
 452   }
 453   return found;
 454 }
 455
 456 }  // packet_processing_helpers
 457
 458 P2PSocketHost::P2PSocketHost(IPC::Sender* message_sender,
 459                              int socket_id,
 460                              ProtocolType protocol_type)
 461     : message_sender_(message_sender),
 462       id_(socket_id),
 463       state_(STATE_UNINITIALIZED),
 464       dump_incoming_rtp_packet_(false),
 465       dump_outgoing_rtp_packet_(false),
 466       weak_ptr_factory_(this),
 467       protocol_type_(protocol_type),
 468       send_packets_delayed_total_(0),
 469       send_packets_total_(0),
 470       send_bytes_delayed_max_(0),
 471       send_bytes_delayed_cur_(0) {
 472 }
 473
 474 P2PSocketHost::~P2PSocketHost() {
 475   if (protocol_type_ == P2PSocketHost::UDP) {
 476     UMA_HISTOGRAM_COUNTS_10000("WebRTC.SystemMaxConsecutiveBytesDelayed_UDP",
 477                                send_bytes_delayed_max_);
 478   } else {
 479     UMA_HISTOGRAM_COUNTS_10000("WebRTC.SystemMaxConsecutiveBytesDelayed_TCP",
 480                                send_bytes_delayed_max_);
 481   }
 482
 483   if (send_packets_total_ > 0) {
 484     int delay_rate = (send_packets_delayed_total_ * 100) / send_packets_total_;
 485     if (protocol_type_ == P2PSocketHost::UDP) {
 486       UMA_HISTOGRAM_PERCENTAGE("WebRTC.SystemPercentPacketsDelayed_UDP",
 487                                delay_rate);
 488     } else {
 489       UMA_HISTOGRAM_PERCENTAGE("WebRTC.SystemPercentPacketsDelayed_TCP",
 490                                delay_rate);
 491     }
 492   }
 493 }
 494
 495 // Verifies that the packet |data| has a valid STUN header.
 496 // static
 497 bool P2PSocketHost::GetStunPacketType(
 498     const char* data, int data_size, StunMessageType* type) {
 499
 500   if (data_size < kStunHeaderSize) {
 501     return false;
 502   }
 503
 504   uint32 cookie = base::NetToHost32(*reinterpret_cast<const uint32*>(data + 4));
 505   if (cookie != kStunMagicCookie) {
 506     return false;
 507   }
 508
 509   uint16 length = base::NetToHost16(*reinterpret_cast<const uint16*>(data + 2));
 510   if (length != data_size - kStunHeaderSize) {
 511     return false;
 512   }
 513
 514   int message_type = base::NetToHost16(*reinterpret_cast<const uint16*>(data));
 515
 516   // Verify that the type is known:
 517   switch (message_type) {
 518     case STUN_BINDING_REQUEST:
 519     case STUN_BINDING_RESPONSE:
 520     case STUN_BINDING_ERROR_RESPONSE:
 521     case STUN_SHARED_SECRET_REQUEST:
 522     case STUN_SHARED_SECRET_RESPONSE:
 523     case STUN_SHARED_SECRET_ERROR_RESPONSE:
 524     case STUN_ALLOCATE_REQUEST:
 525     case STUN_ALLOCATE_RESPONSE:
 526     case STUN_ALLOCATE_ERROR_RESPONSE:
 527     case STUN_SEND_REQUEST:
 528     case STUN_SEND_RESPONSE:
 529     case STUN_SEND_ERROR_RESPONSE:
 530     case STUN_DATA_INDICATION:
 531       *type = static_cast<StunMessageType>(message_type);
 532       return true;
 533
 534     default:
 535       return false;
 536   }
 537 }
 538
 539 // static
 540 bool P2PSocketHost::IsRequestOrResponse(StunMessageType type) {
 541   return type == STUN_BINDING_REQUEST || type == STUN_BINDING_RESPONSE ||
 542       type == STUN_ALLOCATE_REQUEST || type == STUN_ALLOCATE_RESPONSE;
 543 }
 544
 545 // static
 546 P2PSocketHost* P2PSocketHost::Create(IPC::Sender* message_sender,
 547                                      int socket_id,
 548                                      P2PSocketType type,
 549                                      net::URLRequestContextGetter* url_context,
 550                                      P2PMessageThrottler* throttler) {
 551   switch (type) {
 552     case P2P_SOCKET_UDP:
 553       return new P2PSocketHostUdp(message_sender, socket_id, throttler);
 554     case P2P_SOCKET_TCP_SERVER:
 555       return new P2PSocketHostTcpServer(
 556           message_sender, socket_id, P2P_SOCKET_TCP_CLIENT);
 557
 558     case P2P_SOCKET_STUN_TCP_SERVER:
 559       return new P2PSocketHostTcpServer(
 560           message_sender, socket_id, P2P_SOCKET_STUN_TCP_CLIENT);
 561
 562     case P2P_SOCKET_TCP_CLIENT:
 563     case P2P_SOCKET_SSLTCP_CLIENT:
 564     case P2P_SOCKET_TLS_CLIENT:
 565       return new P2PSocketHostTcp(message_sender, socket_id, type, url_context);
 566
 567     case P2P_SOCKET_STUN_TCP_CLIENT:
 568     case P2P_SOCKET_STUN_SSLTCP_CLIENT:
 569     case P2P_SOCKET_STUN_TLS_CLIENT:
 570       return new P2PSocketHostStunTcp(
 571           message_sender, socket_id, type, url_context);
 572   }
 573
 574   NOTREACHED();
 575   return NULL;
 576 }
 577
 578 void P2PSocketHost::StartRtpDump(
 579     bool incoming,
 580     bool outgoing,
 581     const RenderProcessHost::WebRtcRtpPacketCallback& packet_callback) {
 582   DCHECK(BrowserThread::CurrentlyOn(BrowserThread::IO));
 583   DCHECK(!packet_callback.is_null());
 584   DCHECK(incoming || outgoing);
 585
 586   if (incoming) {
 587     dump_incoming_rtp_packet_ = true;
 588   }
 589
 590   if (outgoing) {
 591     dump_outgoing_rtp_packet_ = true;
 592   }
 593
 594   packet_dump_callback_ = packet_callback;
 595 }
 596
 597 void P2PSocketHost::StopRtpDump(bool incoming, bool outgoing) {
 598   DCHECK(BrowserThread::CurrentlyOn(BrowserThread::IO));
 599   DCHECK(incoming || outgoing);
 600
 601   if (incoming) {
 602     dump_incoming_rtp_packet_ = false;
 603   }
 604
 605   if (outgoing) {
 606     dump_outgoing_rtp_packet_ = false;
 607   }
 608
 609   if (!dump_incoming_rtp_packet_ && !dump_outgoing_rtp_packet_) {
 610     packet_dump_callback_.Reset();
 611   }
 612 }
 613
 614 void P2PSocketHost::DumpRtpPacket(const char* packet,
 615                                   size_t length,
 616                                   bool incoming) {
 617   if (IsDtlsPacket(packet, length) || IsRtcpPacket(packet, length)) {
 618     return;
 619   }
 620
 621   size_t rtp_packet_pos = 0;
 622   size_t rtp_packet_length = length;
 623   if (!packet_processing_helpers::GetRtpPacketStartPositionAndLength(
 624           packet, length, &rtp_packet_pos, &rtp_packet_length)) {
 625     return;
 626   }
 627
 628   packet += rtp_packet_pos;
 629
 630   size_t header_length = 0;
 631   bool valid = ValidateRtpHeader(packet, rtp_packet_length, &header_length);
 632   if (!valid) {
 633     DCHECK(false);
 634     return;
 635   }
 636
 637   scoped_ptr<uint8[]> header_buffer(new uint8[header_length]);
 638   memcpy(header_buffer.get(), packet, header_length);
 639
 640   // Posts to the IO thread as the data members should be accessed on the IO
 641   // thread only.
 642   BrowserThread::PostTask(BrowserThread::IO,
 643                           FROM_HERE,
 644                           base::Bind(&P2PSocketHost::DumpRtpPacketOnIOThread,
 645                                      weak_ptr_factory_.GetWeakPtr(),
 646                                      Passed(&header_buffer),
 647                                      header_length,
 648                                      rtp_packet_length,
 649                                      incoming));
 650 }
 651
 652 void P2PSocketHost::DumpRtpPacketOnIOThread(scoped_ptr<uint8[]> packet_header,
 653                                             size_t header_length,
 654                                             size_t packet_length,
 655                                             bool incoming) {
 656   DCHECK(BrowserThread::CurrentlyOn(BrowserThread::IO));
 657
 658   if ((incoming && !dump_incoming_rtp_packet_) ||
 659       (!incoming && !dump_outgoing_rtp_packet_) ||
 660       packet_dump_callback_.is_null()) {
 661     return;
 662   }
 663
 664   // |packet_dump_callback_| must be called on the UI thread.
 665   BrowserThread::PostTask(BrowserThread::UI,
 666                           FROM_HERE,
 667                           base::Bind(packet_dump_callback_,
 668                                      Passed(&packet_header),
 669                                      header_length,
 670                                      packet_length,
 671                                      incoming));
 672 }
 673
 674 void P2PSocketHost::IncrementDelayedPackets() {
 675   send_packets_delayed_total_++;
 676 }
 677
 678 void P2PSocketHost::IncrementTotalSentPackets() {
 679   send_packets_total_++;
 680 }
 681
 682 void P2PSocketHost::IncrementDelayedBytes(uint32 size) {
 683   send_bytes_delayed_cur_ += size;
 684   if (send_bytes_delayed_cur_ > send_bytes_delayed_max_) {
 685     send_bytes_delayed_max_ = send_bytes_delayed_cur_;
 686   }
 687 }
 688
 689 void P2PSocketHost::DecrementDelayedBytes(uint32 size) {
 690   send_bytes_delayed_cur_ -= size;
 691   DCHECK_GE(send_bytes_delayed_cur_, 0);
 692 }
 693
 694 }  // namespace content