78ab1cf39c
The negotiation of encrypted header extensions has already been enabled in Chromium, https://chromium-review.googlesource.com/c/chromium/src/+/5933829. Hence, it make sense to enable the encryption of header extensions by default also in webRTC environment so that all the tests run by taking this into considiration when new changes are made. Bug: webrtc:358039777 Change-Id: I141fac01b0eb0f2ce5a0a365736f0dcf9f21ddcd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366420 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Emil Vardar (xWF) <vardar@google.com> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43573}
74 lines
2.7 KiB
C++
74 lines
2.7 KiB
C++
/*
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* Copyright 2018 The WebRTC Project Authors. All rights reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef API_CRYPTO_CRYPTO_OPTIONS_H_
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#define API_CRYPTO_CRYPTO_OPTIONS_H_
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#include <vector>
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#include "rtc_base/system/rtc_export.h"
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namespace webrtc {
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// CryptoOptions defines advanced cryptographic settings for native WebRTC.
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// These settings must be passed into PeerConnectionFactoryInterface::Options
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// and are only applicable to native use cases of WebRTC.
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struct RTC_EXPORT CryptoOptions {
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CryptoOptions();
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CryptoOptions(const CryptoOptions& other);
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~CryptoOptions();
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// Helper method to return an instance of the CryptoOptions with GCM crypto
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// suites disabled. This method should be used instead of depending on current
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// default values set by the constructor.
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static CryptoOptions NoGcm();
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// Returns a list of the supported DTLS-SRTP Crypto suites based on this set
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// of crypto options.
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std::vector<int> GetSupportedDtlsSrtpCryptoSuites() const;
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bool operator==(const CryptoOptions& other) const;
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bool operator!=(const CryptoOptions& other) const;
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// SRTP Related Peer Connection options.
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struct Srtp {
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// Enable GCM crypto suites from RFC 7714 for SRTP. GCM will only be used
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// if both sides enable it.
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bool enable_gcm_crypto_suites = true;
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// If set to true, the (potentially insecure) crypto cipher
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// kSrtpAes128CmSha1_32 will be included in the list of supported ciphers
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// during negotiation. It will only be used if both peers support it and no
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// other ciphers get preferred.
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bool enable_aes128_sha1_32_crypto_cipher = false;
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// The most commonly used cipher. Can be disabled, mostly for testing
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// purposes.
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bool enable_aes128_sha1_80_crypto_cipher = true;
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// This feature enables encrypting RTP header extensions using RFC 6904, if
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// requested. For this to work the Chromium field trial
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// `kWebRtcEncryptedRtpHeaderExtensions` must be enabled.
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bool enable_encrypted_rtp_header_extensions = true;
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} srtp;
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// Options to be used when the FrameEncryptor / FrameDecryptor APIs are used.
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struct SFrame {
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// If set all RtpSenders must have an FrameEncryptor attached to them before
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// they are allowed to send packets. All RtpReceivers must have a
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// FrameDecryptor attached to them before they are able to receive packets.
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bool require_frame_encryption = false;
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} sframe;
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};
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} // namespace webrtc
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#endif // API_CRYPTO_CRYPTO_OPTIONS_H_
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