From 6bf0a5cb5034a7e684dcc3500e841785237ce2dd Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Sun, 7 Apr 2024 19:32:43 +0200 Subject: Adding upstream version 1:115.7.0. Signed-off-by: Daniel Baumann --- third_party/libwebrtc/api/crypto/crypto_options.h | 72 +++++++++++++++++++++++ 1 file changed, 72 insertions(+) create mode 100644 third_party/libwebrtc/api/crypto/crypto_options.h (limited to 'third_party/libwebrtc/api/crypto/crypto_options.h') diff --git a/third_party/libwebrtc/api/crypto/crypto_options.h b/third_party/libwebrtc/api/crypto/crypto_options.h new file mode 100644 index 0000000000..317995134a --- /dev/null +++ b/third_party/libwebrtc/api/crypto/crypto_options.h @@ -0,0 +1,72 @@ +/* + * 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 + +#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 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_ -- cgit v1.2.3