Add rtppacketuil.h/cc

talk/media/base/rtputils.cc

 /*
  * libjingle
  * Copyright 2011 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
  *     this list of conditions and the following disclaimer in the documentation
  *     and/or other materials provided with the distribution.
  *  3. The name of the author may not be used to endorse or promote products
  *     derived from this software without specific prior written permission.
  *
  * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
  * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
  * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
  * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
  * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
  * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
  * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
  * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
  * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
 #include "talk/media/base/rtputils.h"
 
+#include "talk/media/base/turnutils.h"
+// PacketTimeUpdateParams is defined in asyncpacketsocket.h.
+// TODO(sergeyu): Find more appropriate place for PacketTimeUpdateParams.
+#include "webrtc/base/asyncpacketsocket.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/messagedigest.h"
+
 namespace cricket {
 
 static const uint8_t kRtpVersion = 2;
 static const size_t kRtpFlagsOffset = 0;
 static const size_t kRtpPayloadTypeOffset = 1;
 static const size_t kRtpSeqNumOffset = 2;
 static const size_t kRtpTimestampOffset = 4;
 static const size_t kRtpSsrcOffset = 8;
 static const size_t kRtcpPayloadTypeOffset = 1;
+static const size_t kRtpExtensionHeaderLen = 4;
+static const size_t kAbsSendTimeExtensionLen = 3;
+static const size_t kOneByteExtensionHeaderLen = 1;
+
+namespace {
+
+// Fake auth tag written by the sender when external authentication is enabled.
+// HMAC in packet will be compared against this value before updating packet
+// with actual HMAC value.
+static const uint8_t kFakeAuthTag[10] = {
+    0xba, 0xdd, 0xba, 0xdd, 0xba, 0xdd, 0xba, 0xdd, 0xba, 0xdd
+};
+
+void UpdateAbsSendTimeExtensionValue(uint8_t* extension_data,
+                                     size_t length,
+                                     uint64_t time_us) {
+  // Absolute send time in RTP streams.
+  //
+  // The absolute send time is signaled to the receiver in-band using the
+  // general mechanism for RTP header extensions [RFC5285]. The payload
+  // of this extension (the transmitted value) is a 24-bit unsigned integer
+  // containing the sender's current time in seconds as a fixed point number
+  // with 18 bits fractional part.
+  //
+  // The form of the absolute send time extension block:
+  //
+  //    0                   1                   2                   3
+  //    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
+  //   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+  //   |  ID   | len=2 |              absolute send time               |
+  //   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+  if (length != kAbsSendTimeExtensionLen) {
+    RTC_NOTREACHED();
+    return;
+  }
+
+  // Convert microseconds to a 6.18 fixed point value in seconds.
+  uint32_t send_time = ((time_us << 18) / 1000000) & 0x00FFFFFF;
+  extension_data[0] = static_cast<uint8_t>(send_time >> 16);
+  extension_data[1] = static_cast<uint8_t>(send_time >> 8);
+  extension_data[2] = static_cast<uint8_t>(send_time);
+}
+
+// Assumes |length| is actual packet length + tag length. Updates HMAC at end of
+// the RTP packet.
+void UpdateRtpAuthTag(uint8_t* rtp,
+                      size_t length,
+                      const rtc::PacketTimeUpdateParams& packet_time_params) {
+  // If there is no key, return.
+  if (packet_time_params.srtp_auth_key.empty()) {
+    return;
+  }
+
+  size_t tag_length = packet_time_params.srtp_auth_tag_len;
+
+  // ROC (rollover counter) is at the beginning of the auth tag.
+  const size_t kRocLength = 4;
+  if (tag_length < kRocLength || tag_length > length) {
+    RTC_NOTREACHED();
+    return;
+  }
+
+  uint8_t* auth_tag = rtp + (length - tag_length);
+
+  // We should have a fake HMAC value @ auth_tag.
+  RTC_DCHECK_EQ(0, memcmp(auth_tag, kFakeAuthTag, tag_length));
+
+  // Copy ROC after end of rtp packet.
+  memcpy(auth_tag, &packet_time_params.srtp_packet_index, kRocLength);
+  // Authentication of a RTP packet will have RTP packet + ROC size.
+  size_t auth_required_length = length - tag_length + kRocLength;
+
+  uint8_t output[64];
+  size_t result = rtc::ComputeHmac(
+      rtc::DIGEST_SHA_1, &packet_time_params.srtp_auth_key[0],
+      packet_time_params.srtp_auth_key.size(), rtp,
+      auth_required_length, output, sizeof(output));
+
+  if (result < tag_length) {
+    RTC_NOTREACHED();
+    return;
+  }
+
+  // Copy HMAC from output to packet. This is required as auth tag length
+  // may not be equal to the actual HMAC length.
+  memcpy(auth_tag, output, tag_length);
+}
+
+}
 
 bool GetUint8(const void* data, size_t offset, int* value) {
   if (!data || !value) {
     return false;
   }
   *value = *(static_cast<const uint8_t*>(data) + offset);
   return true;
 }
 
 bool GetUint16(const void* data, size_t offset, int* value) {
@@ skipping 144 matching lines @@
   if (len < kMinRtpPacketLen)
     return false;
 
   return (static_cast<const uint8_t*>(data)[0] >> 6) == kRtpVersion;
 }
 
 bool IsValidRtpPayloadType(int payload_type) {
   return payload_type >= 0 && payload_type <= 127;
 }
 
+bool ValidateRtpHeader(const uint8_t* rtp,
+                       size_t length,
+                       size_t* header_length) {
+  if (header_length) {
+    *header_length = 0;
+  }
+
+  if (length < kMinRtpPacketLen) {
+    return false;
+  }
+
+  size_t cc_count = rtp[0] & 0x0F;
+  size_t header_length_without_extension = kMinRtpPacketLen + 4 * cc_count;
+  if (header_length_without_extension > length) {
+    return false;
+  }
+
+  // If extension bit is not set, we are done with header processing, as input
+  // length is verified above.
+  if (!(rtp[0] & 0x10)) {
+    if (header_length)
+      *header_length = header_length_without_extension;
+
+    return true;
+  }
+
+  rtp += header_length_without_extension;
+
+  if (header_length_without_extension + kRtpExtensionHeaderLen > length) {
+    return false;
+  }
+
+  // Getting extension profile length.
+  // Length is in 32 bit words.
+  uint16_t extension_length_in_32bits = rtc::GetBE16(rtp + 2);
+  size_t extension_length = extension_length_in_32bits * 4;
+
+  size_t rtp_header_length = extension_length +
+                             header_length_without_extension +
+                             kRtpExtensionHeaderLen;
+
+  // Verify input length against total header size.
+  if (rtp_header_length > length) {
+    return false;
+  }
+
+  if (header_length) {
+    *header_length = rtp_header_length;
+  }
+  return true;
+}
+
+// ValidateRtpHeader() must be called before this method to make sure, we have
+// a sane rtp packet.
+bool UpdateRtpAbsSendTimeExtension(uint8_t* rtp,
+                                   size_t length,
+                                   int extension_id,
+                                   uint64_t time_us) {
+  //  0                   1                   2                   3
+  //  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
+  // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+  // |V=2|P|X|  CC   |M|     PT      |       sequence number         |
+  // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+  // |                           timestamp                           |
+  // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+  // |           synchronization source (SSRC) identifier            |
+  // +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+  // |            contributing source (CSRC) identifiers             |
+  // |                             ....                              |
+  // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+  // Return if extension bit is not set.
+  if (!(rtp[0] & 0x10)) {
+    return true;
+  }
+
+  size_t cc_count = rtp[0] & 0x0F;
+  size_t header_length_without_extension = kMinRtpPacketLen + 4 * cc_count;
+
+  rtp += header_length_without_extension;
+
+  // Getting extension profile ID and length.
+  uint16_t profile_id = rtc::GetBE16(rtp);
+  // Length is in 32 bit words.
+  uint16_t extension_length_in_32bits = rtc::GetBE16(rtp + 2);
+  size_t extension_length = extension_length_in_32bits * 4;
+
+  rtp += kRtpExtensionHeaderLen;  // Moving past extension header.
+
+  bool found = false;
+  // WebRTC is using one byte header extension.
+  // TODO(mallinath) - Handle two byte header extension.
+  if (profile_id == 0xBEDE) {  // OneByte extension header
+    //  0
+    //  0 1 2 3 4 5 6 7
+    // +-+-+-+-+-+-+-+-+
+    // |  ID   |length |
+    // +-+-+-+-+-+-+-+-+
+
+    //  0                   1                   2                   3
+    //  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
+    // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    // |       0xBE    |    0xDE       |           length=3            |
+    // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    // |  ID   | L=0   |     data      |  ID   |  L=1  |   data...
+    // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    //       ...data   |    0 (pad)    |    0 (pad)    |  ID   | L=3   |
+    // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    // |                          data                                 |
+    // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    const uint8_t* extension_start = rtp;
+    const uint8_t* extension_end = extension_start + extension_length;
+
+    while (rtp < extension_end) {
+      const int id = (*rtp & 0xF0) >> 4;
+      const size_t length = (*rtp & 0x0F) + 1;
+      if (rtp + kOneByteExtensionHeaderLen + length > extension_end) {
+        return false;
+      }
+      // The 4-bit length is the number minus one of data bytes of this header
+      // extension element following the one-byte header.
+      if (id == extension_id) {
+        UpdateAbsSendTimeExtensionValue(rtp + kOneByteExtensionHeaderLen,
+                                        length, time_us);
+        found = true;
+        break;
+      }
+      rtp += kOneByteExtensionHeaderLen + length;
+      // Counting padding bytes.
+      while ((rtp < extension_end) && (*rtp == 0)) {
+        ++rtp;
+      }
+    }
+  }
+  return found;
+}
+
+bool ApplyPacketOptions(uint8_t* data,
+                        size_t length,
+                        const rtc::PacketTimeUpdateParams& packet_time_params,
+                        uint64_t time_us) {
+  RTC_DCHECK(data);
+  RTC_DCHECK(length);
+
+  // if there is no valid |rtp_sendtime_extension_id| and |srtp_auth_key| in
+  // PacketOptions, nothing to be updated in this packet.
+  if (packet_time_params.rtp_sendtime_extension_id == -1 &&
+      packet_time_params.srtp_auth_key.empty()) {
+    return true;
+  }
+
+  // If there is a srtp auth key present then the packet must be an RTP packet.
+  // RTP packet may have been wrapped in a TURN Channel Data or TURN send
+  // indication.
+  size_t rtp_start_pos;
+  size_t rtp_length;
+  if (!UnwrapTurnPacket(data, length, &rtp_start_pos, &rtp_length)) {
+    RTC_NOTREACHED();
+    return false;
+  }
+
+  // Making sure we have a valid RTP packet at the end.
+  if (!IsRtpPacket(data + rtp_start_pos, rtp_length) ||
+      !ValidateRtpHeader(data + rtp_start_pos, rtp_length, nullptr)) {
+    RTC_NOTREACHED();
+    return false;
+  }
+
+  uint8_t* start = data + rtp_start_pos;
+  // If packet option has non default value (-1) for sendtime extension id,
+  // then we should parse the rtp packet to update the timestamp. Otherwise
+  // just calculate HMAC and update packet with it.
+  if (packet_time_params.rtp_sendtime_extension_id != -1) {
+    UpdateRtpAbsSendTimeExtension(start, rtp_length,
+                                  packet_time_params.rtp_sendtime_extension_id,
+                                  time_us);
+  }
+
+  UpdateRtpAuthTag(start, rtp_length, packet_time_params);
+  return true;
+}
+
 }  // namespace cricket