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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/api/crypto/crypto_options.cc | |
parent | Initial commit. (diff) | |
download | thunderbird-upstream.tar.xz thunderbird-upstream.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to '')
-rw-r--r-- | third_party/libwebrtc/api/crypto/crypto_options.cc | 89 |
1 files changed, 89 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/crypto/crypto_options.cc b/third_party/libwebrtc/api/crypto/crypto_options.cc new file mode 100644 index 0000000000..22c5dd464b --- /dev/null +++ b/third_party/libwebrtc/api/crypto/crypto_options.cc @@ -0,0 +1,89 @@ +/* + * 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 "api/crypto/crypto_options.h" + +#include "rtc_base/ssl_stream_adapter.h" + +namespace webrtc { + +CryptoOptions::CryptoOptions() {} + +CryptoOptions::CryptoOptions(const CryptoOptions& other) { + srtp = other.srtp; + sframe = other.sframe; +} + +CryptoOptions::~CryptoOptions() {} + +// static +CryptoOptions CryptoOptions::NoGcm() { + CryptoOptions options; + options.srtp.enable_gcm_crypto_suites = false; + return options; +} + +std::vector<int> CryptoOptions::GetSupportedDtlsSrtpCryptoSuites() const { + std::vector<int> crypto_suites; + // Note: kSrtpAes128CmSha1_80 is what is required to be supported (by + // draft-ietf-rtcweb-security-arch), but kSrtpAes128CmSha1_32 is allowed as + // well, and saves a few bytes per packet if it ends up selected. + // As the cipher suite is potentially insecure, it will only be used if + // enabled by both peers. + if (srtp.enable_aes128_sha1_32_crypto_cipher) { + crypto_suites.push_back(rtc::kSrtpAes128CmSha1_32); + } + if (srtp.enable_aes128_sha1_80_crypto_cipher) { + crypto_suites.push_back(rtc::kSrtpAes128CmSha1_80); + } + + // Note: GCM cipher suites are not the top choice since they increase the + // packet size. In order to negotiate them the other side must not support + // kSrtpAes128CmSha1_80. + if (srtp.enable_gcm_crypto_suites) { + crypto_suites.push_back(rtc::kSrtpAeadAes256Gcm); + crypto_suites.push_back(rtc::kSrtpAeadAes128Gcm); + } + RTC_CHECK(!crypto_suites.empty()); + return crypto_suites; +} + +bool CryptoOptions::operator==(const CryptoOptions& other) const { + struct data_being_tested_for_equality { + struct Srtp { + bool enable_gcm_crypto_suites; + bool enable_aes128_sha1_32_crypto_cipher; + bool enable_aes128_sha1_80_crypto_cipher; + bool enable_encrypted_rtp_header_extensions; + } srtp; + struct SFrame { + bool require_frame_encryption; + } sframe; + }; + static_assert(sizeof(data_being_tested_for_equality) == sizeof(*this), + "Did you add something to CryptoOptions and forget to " + "update operator==?"); + + return srtp.enable_gcm_crypto_suites == other.srtp.enable_gcm_crypto_suites && + srtp.enable_aes128_sha1_32_crypto_cipher == + other.srtp.enable_aes128_sha1_32_crypto_cipher && + srtp.enable_aes128_sha1_80_crypto_cipher == + other.srtp.enable_aes128_sha1_80_crypto_cipher && + srtp.enable_encrypted_rtp_header_extensions == + other.srtp.enable_encrypted_rtp_header_extensions && + sframe.require_frame_encryption == + other.sframe.require_frame_encryption; +} + +bool CryptoOptions::operator!=(const CryptoOptions& other) const { + return !(*this == other); +} + +} // namespace webrtc |