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bfb444ce2c
This change adds a new subcategory to the public native webrtc::CryptoOptions structure: webrtc::CryptoOptions::Frame. This new structure has a single off by default property: crypto_options.frame.require_frame_encryption. This new flag if set prevents RtpSenders from sending outgoing payloads unless a frame_encryptor_ is attached and prevents RtpReceivers from receiving incoming payloads unless a frame_decryptor_ is attached. This option is important to enforce no unencrypted data can ever leave the device or be received. I have also attached bindings for Java and Objective-C. I have implemented this functionality for E2EE audio but not E2EE video since the changes are still in review. Bug: webrtc:9681 Change-Id: Ie184711190e0cdf5ac781f69e9489ceec904736f Reviewed-on: https://webrtc-review.googlesource.com/c/105540 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Commit-Queue: Benjamin Wright <benwright@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25238}
78 lines
2.6 KiB
C++
78 lines
2.6 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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#include "api/crypto/cryptooptions.h"
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#include "rtc_base/sslstreamadapter.h"
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namespace webrtc {
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CryptoOptions::CryptoOptions() {}
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CryptoOptions::CryptoOptions(const CryptoOptions& other) {
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srtp = other.srtp;
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sframe = other.sframe;
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}
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CryptoOptions::~CryptoOptions() {}
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// static
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CryptoOptions CryptoOptions::NoGcm() {
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CryptoOptions options;
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options.srtp.enable_gcm_crypto_suites = false;
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return options;
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}
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std::vector<int> CryptoOptions::GetSupportedDtlsSrtpCryptoSuites() const {
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std::vector<int> crypto_suites;
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if (srtp.enable_gcm_crypto_suites) {
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crypto_suites.push_back(rtc::SRTP_AEAD_AES_256_GCM);
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crypto_suites.push_back(rtc::SRTP_AEAD_AES_128_GCM);
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}
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// Note: SRTP_AES128_CM_SHA1_80 is what is required to be supported (by
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// draft-ietf-rtcweb-security-arch), but SRTP_AES128_CM_SHA1_32 is allowed as
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// well, and saves a few bytes per packet if it ends up selected.
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// As the cipher suite is potentially insecure, it will only be used if
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// enabled by both peers.
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if (srtp.enable_aes128_sha1_32_crypto_cipher) {
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crypto_suites.push_back(rtc::SRTP_AES128_CM_SHA1_32);
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}
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crypto_suites.push_back(rtc::SRTP_AES128_CM_SHA1_80);
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return crypto_suites;
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}
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bool CryptoOptions::operator==(const CryptoOptions& other) const {
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struct data_being_tested_for_equality {
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struct Srtp {
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bool enable_gcm_crypto_suites;
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bool enable_aes128_sha1_32_crypto_cipher;
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bool enable_encrypted_rtp_header_extensions;
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} srtp;
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struct SFrame {
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bool require_frame_encryption;
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} sframe;
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};
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static_assert(sizeof(data_being_tested_for_equality) == sizeof(*this),
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"Did you add something to CryptoOptions and forget to "
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"update operator==?");
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return srtp.enable_gcm_crypto_suites == other.srtp.enable_gcm_crypto_suites &&
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srtp.enable_aes128_sha1_32_crypto_cipher ==
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other.srtp.enable_aes128_sha1_32_crypto_cipher &&
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srtp.enable_encrypted_rtp_header_extensions ==
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other.srtp.enable_encrypted_rtp_header_extensions &&
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sframe.require_frame_encryption ==
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other.sframe.require_frame_encryption;
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}
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bool CryptoOptions::operator!=(const CryptoOptions& other) const {
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return !(*this == other);
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}
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} // namespace webrtc
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