This reverts commit ac2f3d14e45398930bc35ff05ed7a3b9b617d328. Reason for revert: Breaks downstream project Original change's description: > Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h > > Promotes rtc::CryptoOptions to webrtc::CryptoOptions converting it from class > that only handles SRTP configuration to a more generic structure that can be > used and extended for all per peer connection CryptoOptions that can be on a > given PeerConnection. > > Now all SRTP related options are under webrtc::CryptoOptions::Srtp and can be > accessed as crypto_options.srtp.whatever_option_name. This is more inline with > other structures we have in WebRTC such as VideoConfig. As additional features > are added over time this will allow the structure to remain compartmentalized > and concerned components can only request a subset of the overall configuration > structure e.g: > > void MySrtpFunction(const webrtc::CryptoOptions::Srtp& srtp_config); > > In addition to this it made little sense for sslstreamadapter.h to hold all > Srtp related configuration options. The header has become loo large and takes on > too many responsibilities and spilting this up will lead to more maintainable > code going forward. > > This will be used in a future CL to enable configuration options for the newly > supported Frame Crypto. > > Change-Id: I99d1be36740c59548c8e62db52d68d738649707f > Bug: webrtc:9681 > Reviewed-on: https://webrtc-review.googlesource.com/c/105180 > Reviewed-by: Emad Omara <emadomara@webrtc.org> > Reviewed-by: Kári Helgason <kthelgason@webrtc.org> > Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> > Reviewed-by: Qingsi Wang <qingsi@webrtc.org> > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Commit-Queue: Benjamin Wright <benwright@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25130} TBR=steveanton@webrtc.org,sakal@webrtc.org,kthelgason@webrtc.org,emadomara@webrtc.org,qingsi@webrtc.org,benwright@webrtc.org Bug: webrtc:9681 Change-Id: Ib0075c477c951b540d4deecb3b0cf8cf86ba0fff Reviewed-on: https://webrtc-review.googlesource.com/c/105541 Reviewed-by: Oleh Prypin <oprypin@webrtc.org> Commit-Queue: Oleh Prypin <oprypin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25133}
171 lines
5.7 KiB
C++
171 lines
5.7 KiB
C++
/*
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* Copyright 2004 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 "rtc_base/sslstreamadapter.h"
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#include "rtc_base/opensslstreamadapter.h"
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///////////////////////////////////////////////////////////////////////////////
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namespace rtc {
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// TODO(guoweis): Move this to SDP layer and use int form internally.
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// webrtc:5043.
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const char CS_AES_CM_128_HMAC_SHA1_80[] = "AES_CM_128_HMAC_SHA1_80";
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const char CS_AES_CM_128_HMAC_SHA1_32[] = "AES_CM_128_HMAC_SHA1_32";
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const char CS_AEAD_AES_128_GCM[] = "AEAD_AES_128_GCM";
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const char CS_AEAD_AES_256_GCM[] = "AEAD_AES_256_GCM";
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std::string SrtpCryptoSuiteToName(int crypto_suite) {
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switch (crypto_suite) {
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case SRTP_AES128_CM_SHA1_32:
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return CS_AES_CM_128_HMAC_SHA1_32;
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case SRTP_AES128_CM_SHA1_80:
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return CS_AES_CM_128_HMAC_SHA1_80;
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case SRTP_AEAD_AES_128_GCM:
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return CS_AEAD_AES_128_GCM;
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case SRTP_AEAD_AES_256_GCM:
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return CS_AEAD_AES_256_GCM;
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default:
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return std::string();
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}
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}
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int SrtpCryptoSuiteFromName(const std::string& crypto_suite) {
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if (crypto_suite == CS_AES_CM_128_HMAC_SHA1_32)
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return SRTP_AES128_CM_SHA1_32;
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if (crypto_suite == CS_AES_CM_128_HMAC_SHA1_80)
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return SRTP_AES128_CM_SHA1_80;
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if (crypto_suite == CS_AEAD_AES_128_GCM)
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return SRTP_AEAD_AES_128_GCM;
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if (crypto_suite == CS_AEAD_AES_256_GCM)
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return SRTP_AEAD_AES_256_GCM;
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return SRTP_INVALID_CRYPTO_SUITE;
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}
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bool GetSrtpKeyAndSaltLengths(int crypto_suite,
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int* key_length,
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int* salt_length) {
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switch (crypto_suite) {
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case SRTP_AES128_CM_SHA1_32:
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case SRTP_AES128_CM_SHA1_80:
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// SRTP_AES128_CM_HMAC_SHA1_32 and SRTP_AES128_CM_HMAC_SHA1_80 are defined
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// in RFC 5764 to use a 128 bits key and 112 bits salt for the cipher.
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*key_length = 16;
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*salt_length = 14;
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break;
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case SRTP_AEAD_AES_128_GCM:
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// SRTP_AEAD_AES_128_GCM is defined in RFC 7714 to use a 128 bits key and
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// a 96 bits salt for the cipher.
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*key_length = 16;
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*salt_length = 12;
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break;
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case SRTP_AEAD_AES_256_GCM:
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// SRTP_AEAD_AES_256_GCM is defined in RFC 7714 to use a 256 bits key and
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// a 96 bits salt for the cipher.
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*key_length = 32;
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*salt_length = 12;
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break;
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default:
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return false;
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}
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return true;
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}
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bool IsGcmCryptoSuite(int crypto_suite) {
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return (crypto_suite == SRTP_AEAD_AES_256_GCM ||
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crypto_suite == SRTP_AEAD_AES_128_GCM);
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}
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bool IsGcmCryptoSuiteName(const std::string& crypto_suite) {
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return (crypto_suite == CS_AEAD_AES_256_GCM ||
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crypto_suite == CS_AEAD_AES_128_GCM);
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}
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// static
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CryptoOptions CryptoOptions::NoGcm() {
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CryptoOptions options;
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options.enable_gcm_crypto_suites = false;
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return options;
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}
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std::vector<int> GetSupportedDtlsSrtpCryptoSuites(
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const rtc::CryptoOptions& crypto_options) {
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std::vector<int> crypto_suites;
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if (crypto_options.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 (crypto_options.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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SSLStreamAdapter* SSLStreamAdapter::Create(StreamInterface* stream) {
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return new OpenSSLStreamAdapter(stream);
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}
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SSLStreamAdapter::SSLStreamAdapter(StreamInterface* stream)
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: StreamAdapterInterface(stream),
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ignore_bad_cert_(false),
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client_auth_enabled_(true) {}
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SSLStreamAdapter::~SSLStreamAdapter() {}
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bool SSLStreamAdapter::GetSslCipherSuite(int* cipher_suite) {
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return false;
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}
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bool SSLStreamAdapter::ExportKeyingMaterial(const std::string& label,
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const uint8_t* context,
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size_t context_len,
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bool use_context,
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uint8_t* result,
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size_t result_len) {
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return false; // Default is unsupported
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}
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bool SSLStreamAdapter::SetDtlsSrtpCryptoSuites(
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const std::vector<int>& crypto_suites) {
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return false;
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}
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bool SSLStreamAdapter::GetDtlsSrtpCryptoSuite(int* crypto_suite) {
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return false;
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}
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bool SSLStreamAdapter::IsBoringSsl() {
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return OpenSSLStreamAdapter::IsBoringSsl();
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}
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bool SSLStreamAdapter::IsAcceptableCipher(int cipher, KeyType key_type) {
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return OpenSSLStreamAdapter::IsAcceptableCipher(cipher, key_type);
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}
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bool SSLStreamAdapter::IsAcceptableCipher(const std::string& cipher,
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KeyType key_type) {
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return OpenSSLStreamAdapter::IsAcceptableCipher(cipher, key_type);
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}
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std::string SSLStreamAdapter::SslCipherSuiteToName(int cipher_suite) {
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return OpenSSLStreamAdapter::SslCipherSuiteToName(cipher_suite);
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}
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void SSLStreamAdapter::enable_time_callback_for_testing() {
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OpenSSLStreamAdapter::enable_time_callback_for_testing();
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}
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///////////////////////////////////////////////////////////////////////////////
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} // namespace rtc
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