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/*
* 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
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