Created the DtlsSrtpTransport.

webrtc/pc/dtlssrtptransport.cc

Index: webrtc/pc/dtlssrtptransport.cc
diff --git a/webrtc/pc/dtlssrtptransport.cc b/webrtc/pc/dtlssrtptransport.cc
new file mode 100644
index 0000000000000000000000000000000000000000..c0876ab82b172fd5f532e68dfcfcfcbd476a9372
--- /dev/null
+++ b/webrtc/pc/dtlssrtptransport.cc
@@ -0,0 +1,241 @@
+/*
+ *  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 "webrtc/pc/dtlssrtptransport.h"
+
+#include <memory>
+#include <string>
+#include <utility>
+
+#include "webrtc/media/base/rtputils.h"
+#include "webrtc/rtc_base/sslstreamadapter.h"
+
+namespace {
+// Value specified in RFC 5764.
+static const char kDtlsSrtpExporterLabel[] = "EXTRACTOR-dtls_srtp";
+}  // namespace
+
+namespace webrtc {
+
+DtlsSrtpTransport::DtlsSrtpTransport(
+    std::unique_ptr<webrtc::SrtpTransport> transport,
+    const std::string& content_name)
+    : srtp_transport_(std::move(transport)), content_name_(content_name) {
+  ConnectToSrtpTransport();
+}
+
+void DtlsSrtpTransport::ConnectToSrtpTransport() {
+  RTC_DCHECK(srtp_transport_);
+  srtp_transport_->SignalPacketReceived.connect(
+      this, &DtlsSrtpTransport::OnPacketReceived);
+  srtp_transport_->SignalReadyToSend.connect(this,
+                                             &DtlsSrtpTransport::OnReadyToSend);
+}
+
+void DtlsSrtpTransport::OnPacketReceived(bool rtcp,
+                                         rtc::CopyOnWriteBuffer* packet,
+                                         const rtc::PacketTime& packet_time) {
+  SignalPacketReceived(rtcp, packet, packet_time);
+}
+
+void DtlsSrtpTransport::OnReadyToSend(bool ready) {
+  SignalReadyToSend(ready);
+}
+
+bool DtlsSrtpTransport::SetupDtlsSrtp(bool rtcp) {
+  bool ret = false;
+
+  cricket::DtlsTransportInternal* transport =
+      rtcp ? rtcp_dtls_transport_ : rtp_dtls_transport_;
+  RTC_DCHECK(transport);
+  RTC_DCHECK(transport->IsDtlsActive());
+
+  int selected_crypto_suite;
+
+  if (!transport->GetSrtpCryptoSuite(&selected_crypto_suite)) {
+    LOG(LS_ERROR) << "No DTLS-SRTP selected crypto suite";
+    return false;
+  }
+
+  LOG(LS_INFO) << "Installing keys from DTLS-SRTP on " << content_name_ << " "

[pthatcher, 2017/09/12 00:24:44]

I don't think it's worth passing in the content_name just for this comment.

[Zhi Huang, 2017/09/27 00:46:32]

Ok. I'll remove it then.

+               << cricket::RtpRtcpStringLiteral(rtcp);
+
+  int key_len;
+  int salt_len;
+  if (!rtc::GetSrtpKeyAndSaltLengths(selected_crypto_suite, &key_len,
+                                     &salt_len)) {
+    LOG(LS_ERROR) << "Unknown DTLS-SRTP crypto suite" << selected_crypto_suite;
+    return false;
+  }
+
+  // OK, we're now doing DTLS (RFC 5764)
+  std::vector<unsigned char> dtls_buffer(key_len * 2 + salt_len * 2);
+
+  // RFC 5705 exporter using the RFC 5764 parameters
+  if (!transport->ExportKeyingMaterial(kDtlsSrtpExporterLabel, NULL, 0, false,
+                                       &dtls_buffer[0], dtls_buffer.size())) {
+    LOG(LS_WARNING) << "DTLS-SRTP key export failed";
+    RTC_NOTREACHED();  // This should never happen
+    return false;
+  }
+
+  // Sync up the keys with the DTLS-SRTP interface
+  std::vector<unsigned char> client_write_key(key_len + salt_len);
+  std::vector<unsigned char> server_write_key(key_len + salt_len);
+  size_t offset = 0;
+  memcpy(&client_write_key[0], &dtls_buffer[offset], key_len);
+  offset += key_len;
+  memcpy(&server_write_key[0], &dtls_buffer[offset], key_len);
+  offset += key_len;
+  memcpy(&client_write_key[key_len], &dtls_buffer[offset], salt_len);
+  offset += salt_len;
+  memcpy(&server_write_key[key_len], &dtls_buffer[offset], salt_len);
+
+  std::vector<unsigned char>*send_key, *recv_key;
+  rtc::SSLRole role;
+  if (!transport->GetSslRole(&role)) {
+    LOG(LS_WARNING) << "GetSslRole failed";
+    return false;
+  }
+
+  if (role == rtc::SSL_SERVER) {
+    send_key = &server_write_key;
+    recv_key = &client_write_key;
+  } else {
+    send_key = &client_write_key;
+    recv_key = &server_write_key;
+  }
+
+  if (rtcp) {
+    if (!IsActive()) {
+      RTC_DCHECK(srtp_transport_);
+      ret = srtp_transport_->SetRtcpParams(
+          selected_crypto_suite, &(*send_key)[0],
+          static_cast<int>(send_key->size()), selected_crypto_suite,
+          &(*recv_key)[0], static_cast<int>(recv_key->size()));
+    } else {
+      // RTCP doesn't need to call SetRtpParam because it is only used
+      // to make the updated encrypted RTP header extension IDs take effect.
+      ret = true;
+    }
+  } else {
+    RTC_DCHECK(srtp_transport_);
+    ret = srtp_transport_->SetRtpParams(selected_crypto_suite, &(*send_key)[0],
+                                        static_cast<int>(send_key->size()),
+                                        selected_crypto_suite, &(*recv_key)[0],
+                                        static_cast<int>(recv_key->size()));
+  }
+
+  if (!ret) {
+    LOG(LS_WARNING) << "DTLS-SRTP key installation failed";
+  }
+  active_ = ret;
+  return ret;
+}
+
+void DtlsSrtpTransport::SetDtlsTransport(
+    bool rtcp,
+    cricket::DtlsTransportInternal* dtls_transport) {
+  rtcp ? rtcp_dtls_transport_ = dtls_transport
+       : rtp_dtls_transport_ = dtls_transport;
+}
+
+void DtlsSrtpTransport::SetEncryptedHeaderExtensionIds(
+    cricket::ContentSource source,
+    const std::vector<int>& extension_ids) {
+  RTC_DCHECK(srtp_transport_);
+  srtp_transport_->SetEncryptedHeaderExtensionIds(source, extension_ids);
+}
+
+void DtlsSrtpTransport::SetRtcpMuxEnabled(bool enable) {
+  RTC_DCHECK(srtp_transport_);
+  srtp_transport_->SetRtcpMuxEnabled(enable);
+}
+
+rtc::PacketTransportInternal* DtlsSrtpTransport::rtp_packet_transport() const {
+  RTC_DCHECK(srtp_transport_);
+  return srtp_transport_->rtp_packet_transport();
+}
+
+void DtlsSrtpTransport::SetRtpPacketTransport(
+    rtc::PacketTransportInternal* rtp) {
+  RTC_DCHECK(srtp_transport_);
+  srtp_transport_->SetRtpPacketTransport(rtp);
+}
+
+PacketTransportInterface* DtlsSrtpTransport::GetRtpPacketTransport() const {
+  RTC_DCHECK(srtp_transport_);
+  return srtp_transport_->GetRtpPacketTransport();
+}
+
+rtc::PacketTransportInternal* DtlsSrtpTransport::rtcp_packet_transport() const {
+  RTC_DCHECK(srtp_transport_);
+  return srtp_transport_->rtcp_packet_transport();
+}
+
+void DtlsSrtpTransport::SetRtcpPacketTransport(
+    rtc::PacketTransportInternal* rtcp) {
+  RTC_DCHECK(srtp_transport_);
+  srtp_transport_->SetRtcpPacketTransport(rtcp);
+}
+
+PacketTransportInterface* DtlsSrtpTransport::GetRtcpPacketTransport() const {
+  RTC_DCHECK(srtp_transport_);
+  return srtp_transport_->GetRtcpPacketTransport();
+}
+
+bool DtlsSrtpTransport::IsWritable(bool rtcp) const {
+  RTC_DCHECK(srtp_transport_);
+  return srtp_transport_->IsWritable(rtcp);
+}
+
+bool DtlsSrtpTransport::SendRtpPacket(rtc::CopyOnWriteBuffer* packet,
+                                      const rtc::PacketOptions& options,
+                                      int flags) {
+  RTC_DCHECK(srtp_transport_);
+  return srtp_transport_->SendRtpPacket(packet, options, flags);
+}
+
+bool DtlsSrtpTransport::SendRtcpPacket(rtc::CopyOnWriteBuffer* packet,
+                                       const rtc::PacketOptions& options,
+                                       int flags) {
+  RTC_DCHECK(srtp_transport_);
+  return srtp_transport_->SendRtcpPacket(packet, options, flags);
+}
+
+bool DtlsSrtpTransport::HandlesPayloadType(int payload_type) const {
+  RTC_DCHECK(srtp_transport_);
+  return srtp_transport_->HandlesPayloadType(payload_type);
+}
+
+void DtlsSrtpTransport::AddHandledPayloadType(int payload_type) {
+  RTC_DCHECK(srtp_transport_);
+  srtp_transport_->AddHandledPayloadType(payload_type);
+}
+
+void DtlsSrtpTransport::ResetParams() {
+  active_ = false;
+  if (srtp_transport_) {
+    srtp_transport_->ResetParams();
+  }
+}
+
+RTCError DtlsSrtpTransport::SetParameters(
+    const RtpTransportParameters& parameters) {
+  RTC_DCHECK(srtp_transport_);
+  return srtp_transport_->SetParameters(parameters);
+}
+
+RtpTransportParameters DtlsSrtpTransport::GetParameters() const {
+  RTC_DCHECK(srtp_transport_);
+  return srtp_transport_->GetParameters();
+}
+
+}  // namespace webrtc