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5e23a41ef4
CryptoOptions provided top level optional fields to support Chromium and internal use cases. These locations have been updated to use the new API and this CL removes these legacy compatability options. This CL will be checked in after the chromium CL lands: https://chromium-review.googlesource.com/c/chromium/src/+/1275025 Bug: webrtc:9860 Change-Id: I2790b42c91c49b83e5380a5271df2ceda556c53f Reviewed-on: https://webrtc-review.googlesource.com/c/105644 Commit-Queue: Benjamin Wright <benwright@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25148}
49 lines
1.6 KiB
C++
49 lines
1.6 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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}
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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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