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diff --git a/third_party/libwebrtc/api/crypto/crypto_options.h b/third_party/libwebrtc/api/crypto/crypto_options.h
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+/*
+ *  Copyright 2018 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.
+ */
+
+#ifndef API_CRYPTO_CRYPTO_OPTIONS_H_
+#define API_CRYPTO_CRYPTO_OPTIONS_H_
+
+#include <vector>
+
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// CryptoOptions defines advanced cryptographic settings for native WebRTC.
+// These settings must be passed into PeerConnectionFactoryInterface::Options
+// and are only applicable to native use cases of WebRTC.
+struct RTC_EXPORT CryptoOptions {
+  CryptoOptions();
+  CryptoOptions(const CryptoOptions& other);
+  ~CryptoOptions();
+
+  // Helper method to return an instance of the CryptoOptions with GCM crypto
+  // suites disabled. This method should be used instead of depending on current
+  // default values set by the constructor.
+  static CryptoOptions NoGcm();
+
+  // Returns a list of the supported DTLS-SRTP Crypto suites based on this set
+  // of crypto options.
+  std::vector<int> GetSupportedDtlsSrtpCryptoSuites() const;
+
+  bool operator==(const CryptoOptions& other) const;
+  bool operator!=(const CryptoOptions& other) const;
+
+  // SRTP Related Peer Connection options.
+  struct Srtp {
+    // Enable GCM crypto suites from RFC 7714 for SRTP. GCM will only be used
+    // if both sides enable it.
+    bool enable_gcm_crypto_suites = false;
+
+    // If set to true, the (potentially insecure) crypto cipher
+    // kSrtpAes128CmSha1_32 will be included in the list of supported ciphers
+    // during negotiation. It will only be used if both peers support it and no
+    // other ciphers get preferred.
+    bool enable_aes128_sha1_32_crypto_cipher = false;
+
+    // The most commonly used cipher. Can be disabled, mostly for testing
+    // purposes.
+    bool enable_aes128_sha1_80_crypto_cipher = true;
+
+    // If set to true, encrypted RTP header extensions as defined in RFC 6904
+    // will be negotiated. They will only be used if both peers support them.
+    bool enable_encrypted_rtp_header_extensions = false;
+  } srtp;
+
+  // Options to be used when the FrameEncryptor / FrameDecryptor APIs are used.
+  struct SFrame {
+    // If set all RtpSenders must have an FrameEncryptor attached to them before
+    // they are allowed to send packets. All RtpReceivers must have a
+    // FrameDecryptor attached to them before they are able to receive packets.
+    bool require_frame_encryption = false;
+  } sframe;
+};
+
+}  // namespace webrtc
+
+#endif  // API_CRYPTO_CRYPTO_OPTIONS_H_