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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.
#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(const CryptoOptions& other);
// 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 = true;
// 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