Repo created

TMessagesProj/jni/voip/webrtc/pc/srtp_transport.cc

@@ -0,0 +1,457 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "pc/srtp_transport.h"
+
+#include <string.h>
+
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/match.h"
+#include "media/base/rtp_utils.h"
+#include "modules/rtp_rtcp/source/rtp_util.h"
+#include "pc/rtp_transport.h"
+#include "pc/srtp_session.h"
+#include "rtc_base/async_packet_socket.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/copy_on_write_buffer.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/ssl_stream_adapter.h"
+#include "rtc_base/third_party/base64/base64.h"
+#include "rtc_base/trace_event.h"
+#include "rtc_base/zero_memory.h"
+
+namespace webrtc {
+
+SrtpTransport::SrtpTransport(bool rtcp_mux_enabled,
+                             const FieldTrialsView& field_trials)
+    : RtpTransport(rtcp_mux_enabled), field_trials_(field_trials) {}
+
+bool SrtpTransport::SendRtpPacket(rtc::CopyOnWriteBuffer* packet,
+                                  const rtc::PacketOptions& options,
+                                  int flags) {
+  if (!IsSrtpActive()) {
+    RTC_LOG(LS_ERROR)
+        << "Failed to send the packet because SRTP transport is inactive.";
+    return false;
+  }
+  rtc::PacketOptions updated_options = options;
+  TRACE_EVENT0("webrtc", "SRTP Encode");
+  bool res;
+  uint8_t* data = packet->MutableData();
+  int len = rtc::checked_cast<int>(packet->size());
+// If ENABLE_EXTERNAL_AUTH flag is on then packet authentication is not done
+// inside libsrtp for a RTP packet. A external HMAC module will be writing
+// a fake HMAC value. This is ONLY done for a RTP packet.
+// Socket layer will update rtp sendtime extension header if present in
+// packet with current time before updating the HMAC.
+#if !defined(ENABLE_EXTERNAL_AUTH)
+  res = ProtectRtp(data, len, static_cast<int>(packet->capacity()), &len);
+#else
+  if (!IsExternalAuthActive()) {
+    res = ProtectRtp(data, len, static_cast<int>(packet->capacity()), &len);
+  } else {
+    updated_options.packet_time_params.rtp_sendtime_extension_id =
+        rtp_abs_sendtime_extn_id_;
+    res = ProtectRtp(data, len, static_cast<int>(packet->capacity()), &len,
+                     &updated_options.packet_time_params.srtp_packet_index);
+    // If protection succeeds, let's get auth params from srtp.
+    if (res) {
+      uint8_t* auth_key = nullptr;
+      int key_len = 0;
+      res = GetRtpAuthParams(
+          &auth_key, &key_len,
+          &updated_options.packet_time_params.srtp_auth_tag_len);
+      if (res) {
+        updated_options.packet_time_params.srtp_auth_key.resize(key_len);
+        updated_options.packet_time_params.srtp_auth_key.assign(
+            auth_key, auth_key + key_len);
+      }
+    }
+  }
+#endif
+  if (!res) {
+    uint16_t seq_num = ParseRtpSequenceNumber(*packet);
+    uint32_t ssrc = ParseRtpSsrc(*packet);
+    RTC_LOG(LS_ERROR) << "Failed to protect RTP packet: size=" << len
+                      << ", seqnum=" << seq_num << ", SSRC=" << ssrc;
+    return false;
+  }
+
+  // Update the length of the packet now that we've added the auth tag.
+  packet->SetSize(len);
+  return SendPacket(/*rtcp=*/false, packet, updated_options, flags);
+}
+
+bool SrtpTransport::SendRtcpPacket(rtc::CopyOnWriteBuffer* packet,
+                                   const rtc::PacketOptions& options,
+                                   int flags) {
+  if (!IsSrtpActive()) {
+    RTC_LOG(LS_ERROR)
+        << "Failed to send the packet because SRTP transport is inactive.";
+    return false;
+  }
+
+  TRACE_EVENT0("webrtc", "SRTP Encode");
+  uint8_t* data = packet->MutableData();
+  int len = rtc::checked_cast<int>(packet->size());
+  if (!ProtectRtcp(data, len, static_cast<int>(packet->capacity()), &len)) {
+    int type = -1;
+    cricket::GetRtcpType(data, len, &type);
+    RTC_LOG(LS_ERROR) << "Failed to protect RTCP packet: size=" << len
+                      << ", type=" << type;
+    return false;
+  }
+  // Update the length of the packet now that we've added the auth tag.
+  packet->SetSize(len);
+
+  return SendPacket(/*rtcp=*/true, packet, options, flags);
+}
+
+void SrtpTransport::OnRtpPacketReceived(rtc::CopyOnWriteBuffer packet,
+                                        int64_t packet_time_us) {
+  TRACE_EVENT0("webrtc", "SrtpTransport::OnRtpPacketReceived");
+  if (!IsSrtpActive()) {
+    RTC_LOG(LS_WARNING)
+        << "Inactive SRTP transport received an RTP packet. Drop it.";
+    return;
+  }
+  char* data = packet.MutableData<char>();
+  int len = rtc::checked_cast<int>(packet.size());
+  if (!UnprotectRtp(data, len, &len)) {
+    // Limit the error logging to avoid excessive logs when there are lots of
+    // bad packets.
+    const int kFailureLogThrottleCount = 100;
+    if (decryption_failure_count_ % kFailureLogThrottleCount == 0) {
+      RTC_LOG(LS_ERROR) << "Failed to unprotect RTP packet: size=" << len
+                        << ", seqnum=" << ParseRtpSequenceNumber(packet)
+                        << ", SSRC=" << ParseRtpSsrc(packet)
+                        << ", previous failure count: "
+                        << decryption_failure_count_;
+    }
+    ++decryption_failure_count_;
+    return;
+  }
+  packet.SetSize(len);
+  DemuxPacket(std::move(packet), packet_time_us);
+}
+
+void SrtpTransport::OnRtcpPacketReceived(rtc::CopyOnWriteBuffer packet,
+                                         int64_t packet_time_us) {
+  TRACE_EVENT0("webrtc", "SrtpTransport::OnRtcpPacketReceived");
+  if (!IsSrtpActive()) {
+    RTC_LOG(LS_WARNING)
+        << "Inactive SRTP transport received an RTCP packet. Drop it.";
+    return;
+  }
+  char* data = packet.MutableData<char>();
+  int len = rtc::checked_cast<int>(packet.size());
+  if (!UnprotectRtcp(data, len, &len)) {
+    int type = -1;
+    cricket::GetRtcpType(data, len, &type);
+    RTC_LOG(LS_ERROR) << "Failed to unprotect RTCP packet: size=" << len
+                      << ", type=" << type;
+    return;
+  }
+  packet.SetSize(len);
+  SendRtcpPacketReceived(&packet, packet_time_us);
+}
+
+void SrtpTransport::OnNetworkRouteChanged(
+    absl::optional<rtc::NetworkRoute> network_route) {
+  // Only append the SRTP overhead when there is a selected network route.
+  if (network_route) {
+    int srtp_overhead = 0;
+    if (IsSrtpActive()) {
+      GetSrtpOverhead(&srtp_overhead);
+    }
+    network_route->packet_overhead += srtp_overhead;
+  }
+  SendNetworkRouteChanged(network_route);
+}
+
+void SrtpTransport::OnWritableState(
+    rtc::PacketTransportInternal* packet_transport) {
+  SendWritableState(IsWritable(/*rtcp=*/false) && IsWritable(/*rtcp=*/true));
+}
+
+bool SrtpTransport::SetRtpParams(int send_crypto_suite,
+                                 const uint8_t* send_key,
+                                 int send_key_len,
+                                 const std::vector<int>& send_extension_ids,
+                                 int recv_crypto_suite,
+                                 const uint8_t* recv_key,
+                                 int recv_key_len,
+                                 const std::vector<int>& recv_extension_ids) {
+  // If parameters are being set for the first time, we should create new SRTP
+  // sessions and call "SetSend/SetRecv". Otherwise we should call
+  // "UpdateSend"/"UpdateRecv" on the existing sessions, which will internally
+  // call "srtp_update".
+  bool new_sessions = false;
+  if (!send_session_) {
+    RTC_DCHECK(!recv_session_);
+    CreateSrtpSessions();
+    new_sessions = true;
+  }
+  bool ret = new_sessions
+                 ? send_session_->SetSend(send_crypto_suite, send_key,
+                                          send_key_len, send_extension_ids)
+                 : send_session_->UpdateSend(send_crypto_suite, send_key,
+                                             send_key_len, send_extension_ids);
+  if (!ret) {
+    ResetParams();
+    return false;
+  }
+
+  ret = new_sessions
+            ? recv_session_->SetRecv(recv_crypto_suite, recv_key, recv_key_len,
+                                     recv_extension_ids)
+            : recv_session_->UpdateRecv(recv_crypto_suite, recv_key,
+                                        recv_key_len, recv_extension_ids);
+  if (!ret) {
+    ResetParams();
+    return false;
+  }
+
+  RTC_LOG(LS_INFO) << "SRTP " << (new_sessions ? "activated" : "updated")
+                   << " with negotiated parameters: send crypto_suite "
+                   << send_crypto_suite << " recv crypto_suite "
+                   << recv_crypto_suite;
+  MaybeUpdateWritableState();
+  return true;
+}
+
+bool SrtpTransport::SetRtcpParams(int send_crypto_suite,
+                                  const uint8_t* send_key,
+                                  int send_key_len,
+                                  const std::vector<int>& send_extension_ids,
+                                  int recv_crypto_suite,
+                                  const uint8_t* recv_key,
+                                  int recv_key_len,
+                                  const std::vector<int>& recv_extension_ids) {
+  // This can only be called once, but can be safely called after
+  // SetRtpParams
+  if (send_rtcp_session_ || recv_rtcp_session_) {
+    RTC_LOG(LS_ERROR) << "Tried to set SRTCP Params when filter already active";
+    return false;
+  }
+
+  send_rtcp_session_.reset(new cricket::SrtpSession(field_trials_));
+  if (!send_rtcp_session_->SetSend(send_crypto_suite, send_key, send_key_len,
+                                   send_extension_ids)) {
+    return false;
+  }
+
+  recv_rtcp_session_.reset(new cricket::SrtpSession(field_trials_));
+  if (!recv_rtcp_session_->SetRecv(recv_crypto_suite, recv_key, recv_key_len,
+                                   recv_extension_ids)) {
+    return false;
+  }
+
+  RTC_LOG(LS_INFO) << "SRTCP activated with negotiated parameters:"
+                      " send crypto_suite "
+                   << send_crypto_suite << " recv crypto_suite "
+                   << recv_crypto_suite;
+  MaybeUpdateWritableState();
+  return true;
+}
+
+bool SrtpTransport::IsSrtpActive() const {
+  return send_session_ && recv_session_;
+}
+
+bool SrtpTransport::IsWritable(bool rtcp) const {
+  return IsSrtpActive() && RtpTransport::IsWritable(rtcp);
+}
+
+void SrtpTransport::ResetParams() {
+  send_session_ = nullptr;
+  recv_session_ = nullptr;
+  send_rtcp_session_ = nullptr;
+  recv_rtcp_session_ = nullptr;
+  MaybeUpdateWritableState();
+  RTC_LOG(LS_INFO) << "The params in SRTP transport are reset.";
+}
+
+void SrtpTransport::CreateSrtpSessions() {
+  send_session_.reset(new cricket::SrtpSession(field_trials_));
+  recv_session_.reset(new cricket::SrtpSession(field_trials_));
+  if (external_auth_enabled_) {
+    send_session_->EnableExternalAuth();
+  }
+}
+
+bool SrtpTransport::ProtectRtp(void* p, int in_len, int max_len, int* out_len) {
+  if (!IsSrtpActive()) {
+    RTC_LOG(LS_WARNING) << "Failed to ProtectRtp: SRTP not active";
+    return false;
+  }
+  RTC_CHECK(send_session_);
+  return send_session_->ProtectRtp(p, in_len, max_len, out_len);
+}
+
+bool SrtpTransport::ProtectRtp(void* p,
+                               int in_len,
+                               int max_len,
+                               int* out_len,
+                               int64_t* index) {
+  if (!IsSrtpActive()) {
+    RTC_LOG(LS_WARNING) << "Failed to ProtectRtp: SRTP not active";
+    return false;
+  }
+  RTC_CHECK(send_session_);
+  return send_session_->ProtectRtp(p, in_len, max_len, out_len, index);
+}
+
+bool SrtpTransport::ProtectRtcp(void* p,
+                                int in_len,
+                                int max_len,
+                                int* out_len) {
+  if (!IsSrtpActive()) {
+    RTC_LOG(LS_WARNING) << "Failed to ProtectRtcp: SRTP not active";
+    return false;
+  }
+  if (send_rtcp_session_) {
+    return send_rtcp_session_->ProtectRtcp(p, in_len, max_len, out_len);
+  } else {
+    RTC_CHECK(send_session_);
+    return send_session_->ProtectRtcp(p, in_len, max_len, out_len);
+  }
+}
+
+bool SrtpTransport::UnprotectRtp(void* p, int in_len, int* out_len) {
+  if (!IsSrtpActive()) {
+    RTC_LOG(LS_WARNING) << "Failed to UnprotectRtp: SRTP not active";
+    return false;
+  }
+  RTC_CHECK(recv_session_);
+  return recv_session_->UnprotectRtp(p, in_len, out_len);
+}
+
+bool SrtpTransport::UnprotectRtcp(void* p, int in_len, int* out_len) {
+  if (!IsSrtpActive()) {
+    RTC_LOG(LS_WARNING) << "Failed to UnprotectRtcp: SRTP not active";
+    return false;
+  }
+  if (recv_rtcp_session_) {
+    return recv_rtcp_session_->UnprotectRtcp(p, in_len, out_len);
+  } else {
+    RTC_CHECK(recv_session_);
+    return recv_session_->UnprotectRtcp(p, in_len, out_len);
+  }
+}
+
+bool SrtpTransport::GetRtpAuthParams(uint8_t** key,
+                                     int* key_len,
+                                     int* tag_len) {
+  if (!IsSrtpActive()) {
+    RTC_LOG(LS_WARNING) << "Failed to GetRtpAuthParams: SRTP not active";
+    return false;
+  }
+
+  RTC_CHECK(send_session_);
+  return send_session_->GetRtpAuthParams(key, key_len, tag_len);
+}
+
+bool SrtpTransport::GetSrtpOverhead(int* srtp_overhead) const {
+  if (!IsSrtpActive()) {
+    RTC_LOG(LS_WARNING) << "Failed to GetSrtpOverhead: SRTP not active";
+    return false;
+  }
+
+  RTC_CHECK(send_session_);
+  *srtp_overhead = send_session_->GetSrtpOverhead();
+  return true;
+}
+
+void SrtpTransport::EnableExternalAuth() {
+  RTC_DCHECK(!IsSrtpActive());
+  external_auth_enabled_ = true;
+}
+
+bool SrtpTransport::IsExternalAuthEnabled() const {
+  return external_auth_enabled_;
+}
+
+bool SrtpTransport::IsExternalAuthActive() const {
+  if (!IsSrtpActive()) {
+    RTC_LOG(LS_WARNING)
+        << "Failed to check IsExternalAuthActive: SRTP not active";
+    return false;
+  }
+
+  RTC_CHECK(send_session_);
+  return send_session_->IsExternalAuthActive();
+}
+
+bool SrtpTransport::MaybeSetKeyParams() {
+  if (!send_crypto_suite_ || !recv_crypto_suite_) {
+    return true;
+  }
+
+  return SetRtpParams(*send_crypto_suite_, send_key_.data(),
+                      static_cast<int>(send_key_.size()), std::vector<int>(),
+                      *recv_crypto_suite_, recv_key_.data(),
+                      static_cast<int>(recv_key_.size()), std::vector<int>());
+}
+
+bool SrtpTransport::ParseKeyParams(const std::string& key_params,
+                                   uint8_t* key,
+                                   size_t len) {
+  // example key_params: "inline:YUJDZGVmZ2hpSktMbW9QUXJzVHVWd3l6MTIzNDU2"
+
+  // Fail if key-method is wrong.
+  if (!absl::StartsWith(key_params, "inline:")) {
+    return false;
+  }
+
+  // Fail if base64 decode fails, or the key is the wrong size.
+  std::string key_b64(key_params.substr(7)), key_str;
+  if (!rtc::Base64::Decode(key_b64, rtc::Base64::DO_STRICT, &key_str,
+                           nullptr) ||
+      key_str.size() != len) {
+    return false;
+  }
+
+  memcpy(key, key_str.c_str(), len);
+  // TODO(bugs.webrtc.org/8905): Switch to ZeroOnFreeBuffer for storing
+  // sensitive data.
+  rtc::ExplicitZeroMemory(&key_str[0], key_str.size());
+  return true;
+}
+
+void SrtpTransport::MaybeUpdateWritableState() {
+  bool writable = IsWritable(/*rtcp=*/true) && IsWritable(/*rtcp=*/false);
+  // Only fire the signal if the writable state changes.
+  if (writable_ != writable) {
+    writable_ = writable;
+    SendWritableState(writable_);
+  }
+}
+
+bool SrtpTransport::UnregisterRtpDemuxerSink(RtpPacketSinkInterface* sink) {
+  if (recv_session_ &&
+      field_trials_.IsEnabled("WebRTC-SrtpRemoveReceiveStream")) {
+    // Remove the SSRCs explicitly registered with the demuxer
+    // (via SDP negotiation) from the SRTP session.
+    for (const auto ssrc : GetSsrcsForSink(sink)) {
+      if (!recv_session_->RemoveSsrcFromSession(ssrc)) {
+        RTC_LOG(LS_WARNING)
+            << "Could not remove SSRC " << ssrc << " from SRTP session.";
+      }
+    }
+  }
+  return RtpTransport::UnregisterRtpDemuxerSink(sink);
+}
+
+}  // namespace webrtc