| /* |
| * 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/cryptooptions.h" |
| #include "rtc_base/sslstreamadapter.h" |
| |
| namespace webrtc { |
| |
| CryptoOptions::CryptoOptions() {} |
| |
| CryptoOptions::CryptoOptions(const CryptoOptions& other) { |
| enable_gcm_crypto_suites = other.enable_gcm_crypto_suites; |
| enable_encrypted_rtp_header_extensions = |
| other.enable_encrypted_rtp_header_extensions; |
| srtp = other.srtp; |
| } |
| |
| 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; |
| if (srtp.enable_gcm_crypto_suites) { |
| crypto_suites.push_back(rtc::SRTP_AEAD_AES_256_GCM); |
| crypto_suites.push_back(rtc::SRTP_AEAD_AES_128_GCM); |
| } |
| // Note: SRTP_AES128_CM_SHA1_80 is what is required to be supported (by |
| // draft-ietf-rtcweb-security-arch), but SRTP_AES128_CM_SHA1_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::SRTP_AES128_CM_SHA1_32); |
| } |
| crypto_suites.push_back(rtc::SRTP_AES128_CM_SHA1_80); |
| return crypto_suites; |
| } |
| |
| } // namespace webrtc |