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2ebbff83ee
Move the GCM srtp cipher suites below the default SRTP_AES128_CM_SHA1_80 one. This will not negotiate them by default since they have an impact on packet overhead for audio-only calls. GCM can still be negotiated if the peer offers it as preferred cipher suite or answers with just that cipher suite. BUG=chromium:713701 Change-Id: I79bd4ab827e5c7f55f5550d14db3f4217a7eff86 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158404 Reviewed-by: Justin Uberti <juberti@google.com> Reviewed-by: Justin Uberti <juberti@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Justin Uberti <juberti@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29672}
72 lines
2.7 KiB
C++
72 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 = false;
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// If set to true, the (potentially insecure) crypto cipher
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// SRTP_AES128_CM_SHA1_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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// If set to true, encrypted RTP header extensions as defined in RFC 6904
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// will be negotiated. They will only be used if both peers support them.
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bool enable_encrypted_rtp_header_extensions = false;
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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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