50 lines
1.6 KiB
C++
50 lines
1.6 KiB
C++
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/*
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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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}
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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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} // namespace webrtc
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