Reland "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.
Reland Fix:
- cryptooptions.h - now has enable_aes128_sha1_32_crypto_cipher as an optional
root level configuration.
- peerconnectionfactory - If this optional is set will now overwrite the
underyling value.
This along with the other field will be deprecated once dependent projects
are updated.
TBR=sakal@webrtc.org,kthelgason@webrtc.org,emadomara@webrtc.org,qingsi@webrtc.org
Bug: webrtc:9681
Change-Id: Iaa6b741baafb85d352e42f54226119f19d97151d
Reviewed-on: https://webrtc-review.googlesource.com/c/105560
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Emad Omara <emadomara@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25135}diff --git a/api/BUILD.gn b/api/BUILD.gn
index 60eb751..c420b10 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -52,6 +52,8 @@
"bitrate_constraints.h",
"candidate.cc",
"candidate.h",
+ "crypto/cryptooptions.cc",
+ "crypto/cryptooptions.h",
"crypto/framedecryptorinterface.h",
"crypto/frameencryptorinterface.h",
"cryptoparams.h",
diff --git a/api/crypto/cryptooptions.cc b/api/crypto/cryptooptions.cc
new file mode 100644
index 0000000..ed6db47
--- /dev/null
+++ b/api/crypto/cryptooptions.cc
@@ -0,0 +1,52 @@
+/*
+ * 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
diff --git a/api/crypto/cryptooptions.h b/api/crypto/cryptooptions.h
new file mode 100644
index 0000000..0ac973f
--- /dev/null
+++ b/api/crypto/cryptooptions.h
@@ -0,0 +1,67 @@
+/*
+ * 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.
+ */
+
+#ifndef API_CRYPTO_CRYPTOOPTIONS_H_
+#define API_CRYPTO_CRYPTOOPTIONS_H_
+
+#include <vector>
+#include "absl/types/optional.h"
+
+namespace webrtc {
+
+// CryptoOptions defines advanced cryptographic settings for native WebRTC.
+// These settings must be passed into PeerConnectionFactoryInterface::Options
+// and are only applicable to native use cases of WebRTC.
+struct CryptoOptions {
+ CryptoOptions();
+ CryptoOptions(const CryptoOptions& other);
+ ~CryptoOptions();
+
+ // Helper method to return an instance of the CryptoOptions with GCM crypto
+ // suites disabled. This method should be used instead of depending on current
+ // default values set by the constructor.
+ static CryptoOptions NoGcm();
+
+ // Returns a list of the supported DTLS-SRTP Crypto suites based on this set
+ // of crypto options.
+ std::vector<int> GetSupportedDtlsSrtpCryptoSuites() const;
+
+ // TODO(webrtc:9859) - Remove duplicates once chromium is fixed.
+ // Will be removed once srtp.enable_gcm_crypto_suites is updated in Chrome.
+ absl::optional<bool> enable_gcm_crypto_suites;
+ // TODO(webrtc:9859) - Remove duplicates once chromium is fixed.
+ // Will be removed once srtp.enable_encrypted_rtp_header_extensions is
+ // updated in Chrome.
+ absl::optional<bool> enable_encrypted_rtp_header_extensions;
+ // Will be removed once srtp.enable_encrypted_rtp_header_extensions is
+ // updated in Tacl.
+ absl::optional<bool> enable_aes128_sha1_32_crypto_cipher;
+
+ // SRTP Related Peer Connection options.
+ struct Srtp {
+ // Enable GCM crypto suites from RFC 7714 for SRTP. GCM will only be used
+ // if both sides enable it.
+ bool enable_gcm_crypto_suites = false;
+
+ // If set to true, the (potentially insecure) crypto cipher
+ // SRTP_AES128_CM_SHA1_32 will be included in the list of supported ciphers
+ // during negotiation. It will only be used if both peers support it and no
+ // other ciphers get preferred.
+ bool enable_aes128_sha1_32_crypto_cipher = false;
+
+ // If set to true, encrypted RTP header extensions as defined in RFC 6904
+ // will be negotiated. They will only be used if both peers support them.
+ bool enable_encrypted_rtp_header_extensions = false;
+ } srtp;
+};
+
+} // namespace webrtc
+
+#endif // API_CRYPTO_CRYPTOOPTIONS_H_
diff --git a/api/cryptoparams.h b/api/cryptoparams.h
index 2350528..abe9055 100644
--- a/api/cryptoparams.h
+++ b/api/cryptoparams.h
@@ -16,6 +16,8 @@
namespace cricket {
// Parameters for SRTP negotiation, as described in RFC 4568.
+// TODO(benwright) - Rename to SrtpCryptoParams as these only apply to SRTP and
+// not generic crypto parameters for WebRTC.
struct CryptoParams {
CryptoParams() : tag(0) {}
CryptoParams(int t,
diff --git a/api/peerconnectioninterface.h b/api/peerconnectioninterface.h
index 3d8a9c1..141b2c9 100644
--- a/api/peerconnectioninterface.h
+++ b/api/peerconnectioninterface.h
@@ -77,6 +77,7 @@
#include "api/audio_codecs/audio_encoder_factory.h"
#include "api/audio_options.h"
#include "api/call/callfactoryinterface.h"
+#include "api/crypto/cryptooptions.h"
#include "api/datachannelinterface.h"
#include "api/fec_controller.h"
#include "api/jsep.h"
@@ -1180,7 +1181,7 @@
public:
class Options {
public:
- Options() : crypto_options(rtc::CryptoOptions::NoGcm()) {}
+ Options() {}
// If set to true, created PeerConnections won't enforce any SRTP
// requirement, allowing unsecured media. Should only be used for
@@ -1209,7 +1210,7 @@
rtc::SSLProtocolVersion ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
// Sets crypto related options, e.g. enabled cipher suites.
- rtc::CryptoOptions crypto_options;
+ CryptoOptions crypto_options = CryptoOptions::NoGcm();
};
// Set the options to be used for subsequently created PeerConnections.
diff --git a/p2p/base/dtlstransport.cc b/p2p/base/dtlstransport.cc
index 314f63c..f498253 100644
--- a/p2p/base/dtlstransport.cc
+++ b/p2p/base/dtlstransport.cc
@@ -114,12 +114,12 @@
}
DtlsTransport::DtlsTransport(IceTransportInternal* ice_transport,
- const rtc::CryptoOptions& crypto_options)
+ const webrtc::CryptoOptions& crypto_options)
: transport_name_(ice_transport->transport_name()),
component_(ice_transport->component()),
ice_transport_(ice_transport),
downward_(NULL),
- srtp_ciphers_(GetSupportedDtlsSrtpCryptoSuites(crypto_options)),
+ srtp_ciphers_(crypto_options.GetSupportedDtlsSrtpCryptoSuites()),
ssl_max_version_(rtc::SSL_PROTOCOL_DTLS_12),
crypto_options_(crypto_options) {
RTC_DCHECK(ice_transport_);
@@ -128,13 +128,13 @@
DtlsTransport::DtlsTransport(
std::unique_ptr<IceTransportInternal> ice_transport,
- const rtc::CryptoOptions& crypto_options)
+ const webrtc::CryptoOptions& crypto_options)
: transport_name_(ice_transport->transport_name()),
component_(ice_transport->component()),
ice_transport_(ice_transport.get()),
owned_ice_transport_(std::move(ice_transport)),
downward_(NULL),
- srtp_ciphers_(GetSupportedDtlsSrtpCryptoSuites(crypto_options)),
+ srtp_ciphers_(crypto_options.GetSupportedDtlsSrtpCryptoSuites()),
ssl_max_version_(rtc::SSL_PROTOCOL_DTLS_12),
crypto_options_(crypto_options) {
RTC_DCHECK(owned_ice_transport_);
@@ -143,7 +143,7 @@
DtlsTransport::~DtlsTransport() = default;
-const rtc::CryptoOptions& DtlsTransport::crypto_options() const {
+const webrtc::CryptoOptions& DtlsTransport::crypto_options() const {
return crypto_options_;
}
diff --git a/p2p/base/dtlstransport.h b/p2p/base/dtlstransport.h
index 456f375..2eafc9e 100644
--- a/p2p/base/dtlstransport.h
+++ b/p2p/base/dtlstransport.h
@@ -15,6 +15,7 @@
#include <string>
#include <vector>
+#include "api/crypto/cryptooptions.h"
#include "p2p/base/dtlstransportinternal.h"
#include "p2p/base/icetransportinternal.h"
#include "rtc_base/buffer.h"
@@ -96,13 +97,13 @@
// whether GCM crypto suites are negotiated.
// TODO(zhihuang): Remove this once we switch to JsepTransportController.
explicit DtlsTransport(IceTransportInternal* ice_transport,
- const rtc::CryptoOptions& crypto_options);
+ const webrtc::CryptoOptions& crypto_options);
explicit DtlsTransport(std::unique_ptr<IceTransportInternal> ice_transport,
- const rtc::CryptoOptions& crypto_options);
+ const webrtc::CryptoOptions& crypto_options);
~DtlsTransport() override;
- const rtc::CryptoOptions& crypto_options() const override;
+ const webrtc::CryptoOptions& crypto_options() const override;
DtlsTransportState dtls_state() const override;
const std::string& transport_name() const override;
int component() const override;
@@ -231,7 +232,7 @@
rtc::scoped_refptr<rtc::RTCCertificate> local_certificate_;
absl::optional<rtc::SSLRole> dtls_role_;
rtc::SSLProtocolVersion ssl_max_version_;
- rtc::CryptoOptions crypto_options_;
+ webrtc::CryptoOptions crypto_options_;
rtc::Buffer remote_fingerprint_value_;
std::string remote_fingerprint_algorithm_;
diff --git a/p2p/base/dtlstransport_unittest.cc b/p2p/base/dtlstransport_unittest.cc
index 3cf423e..8dfbf61 100644
--- a/p2p/base/dtlstransport_unittest.cc
+++ b/p2p/base/dtlstransport_unittest.cc
@@ -86,8 +86,8 @@
fake_ice_transport_->SignalReadPacket.connect(
this, &DtlsTestClient::OnFakeIceTransportReadPacket);
- dtls_transport_.reset(
- new DtlsTransport(fake_ice_transport_.get(), rtc::CryptoOptions()));
+ dtls_transport_ = absl::make_unique<DtlsTransport>(
+ fake_ice_transport_.get(), webrtc::CryptoOptions());
dtls_transport_->SetSslMaxProtocolVersion(ssl_max_version_);
// Note: Certificate may be null here if testing passthrough.
dtls_transport_->SetLocalCertificate(certificate_);
diff --git a/p2p/base/dtlstransportinternal.h b/p2p/base/dtlstransportinternal.h
index 238057e..a713744 100644
--- a/p2p/base/dtlstransportinternal.h
+++ b/p2p/base/dtlstransportinternal.h
@@ -15,6 +15,7 @@
#include <string>
#include <vector>
+#include "api/crypto/cryptooptions.h"
#include "p2p/base/icetransportinternal.h"
#include "p2p/base/packettransportinternal.h"
#include "rtc_base/sslstreamadapter.h"
@@ -50,7 +51,7 @@
public:
~DtlsTransportInternal() override;
- virtual const rtc::CryptoOptions& crypto_options() const = 0;
+ virtual const webrtc::CryptoOptions& crypto_options() const = 0;
virtual DtlsTransportState dtls_state() const = 0;
diff --git a/p2p/base/fakedtlstransport.h b/p2p/base/fakedtlstransport.h
index cb944d3..c4f9d2c 100644
--- a/p2p/base/fakedtlstransport.h
+++ b/p2p/base/fakedtlstransport.h
@@ -17,6 +17,7 @@
#include <vector>
#include "absl/memory/memory.h"
+#include "api/crypto/cryptooptions.h"
#include "p2p/base/dtlstransportinternal.h"
#include "p2p/base/fakeicetransport.h"
#include "rtc_base/fakesslidentity.h"
@@ -149,10 +150,10 @@
*role = *dtls_role_;
return true;
}
- const rtc::CryptoOptions& crypto_options() const override {
+ const webrtc::CryptoOptions& crypto_options() const override {
return crypto_options_;
}
- void SetCryptoOptions(const rtc::CryptoOptions& crypto_options) {
+ void SetCryptoOptions(const webrtc::CryptoOptions& crypto_options) {
crypto_options_ = crypto_options;
}
bool SetLocalCertificate(
@@ -272,7 +273,7 @@
rtc::SSLFingerprint dtls_fingerprint_;
absl::optional<rtc::SSLRole> dtls_role_;
int crypto_suite_ = rtc::SRTP_AES128_CM_SHA1_80;
- rtc::CryptoOptions crypto_options_;
+ webrtc::CryptoOptions crypto_options_;
DtlsTransportState dtls_state_ = DTLS_TRANSPORT_NEW;
diff --git a/p2p/base/transportfactoryinterface.h b/p2p/base/transportfactoryinterface.h
index ce32ee8..9805db0 100644
--- a/p2p/base/transportfactoryinterface.h
+++ b/p2p/base/transportfactoryinterface.h
@@ -34,7 +34,7 @@
virtual std::unique_ptr<DtlsTransportInternal> CreateDtlsTransport(
std::unique_ptr<IceTransportInternal> ice,
- const rtc::CryptoOptions& crypto_options) = 0;
+ const webrtc::CryptoOptions& crypto_options) = 0;
};
} // namespace cricket
diff --git a/pc/channel.cc b/pc/channel.cc
index b63cea7..cd1534f 100644
--- a/pc/channel.cc
+++ b/pc/channel.cc
@@ -101,7 +101,7 @@
std::unique_ptr<MediaChannel> media_channel,
const std::string& content_name,
bool srtp_required,
- rtc::CryptoOptions crypto_options)
+ webrtc::CryptoOptions crypto_options)
: worker_thread_(worker_thread),
network_thread_(network_thread),
signaling_thread_(signaling_thread),
@@ -673,7 +673,7 @@
RtpHeaderExtensions BaseChannel::GetFilteredRtpHeaderExtensions(
const RtpHeaderExtensions& extensions) {
RTC_DCHECK(rtp_transport_);
- if (crypto_options_.enable_encrypted_rtp_header_extensions) {
+ if (crypto_options_.srtp.enable_encrypted_rtp_header_extensions) {
RtpHeaderExtensions filtered;
auto pred = [](const webrtc::RtpExtension& extension) {
return !extension.encrypt;
@@ -742,7 +742,7 @@
std::unique_ptr<VoiceMediaChannel> media_channel,
const std::string& content_name,
bool srtp_required,
- rtc::CryptoOptions crypto_options)
+ webrtc::CryptoOptions crypto_options)
: BaseChannel(worker_thread,
network_thread,
signaling_thread,
@@ -881,7 +881,7 @@
std::unique_ptr<VideoMediaChannel> media_channel,
const std::string& content_name,
bool srtp_required,
- rtc::CryptoOptions crypto_options)
+ webrtc::CryptoOptions crypto_options)
: BaseChannel(worker_thread,
network_thread,
signaling_thread,
@@ -1019,7 +1019,7 @@
std::unique_ptr<DataMediaChannel> media_channel,
const std::string& content_name,
bool srtp_required,
- rtc::CryptoOptions crypto_options)
+ webrtc::CryptoOptions crypto_options)
: BaseChannel(worker_thread,
network_thread,
signaling_thread,
diff --git a/pc/channel.h b/pc/channel.h
index ba58469..535f64e 100644
--- a/pc/channel.h
+++ b/pc/channel.h
@@ -82,7 +82,7 @@
std::unique_ptr<MediaChannel> media_channel,
const std::string& content_name,
bool srtp_required,
- rtc::CryptoOptions crypto_options);
+ webrtc::CryptoOptions crypto_options);
virtual ~BaseChannel();
void Init_w(webrtc::RtpTransportInternal* rtp_transport);
@@ -313,7 +313,7 @@
bool was_ever_writable_ = false;
bool has_received_packet_ = false;
const bool srtp_required_ = true;
- rtc::CryptoOptions crypto_options_;
+ webrtc::CryptoOptions crypto_options_;
// MediaChannel related members that should be accessed from the worker
// thread.
@@ -343,7 +343,7 @@
std::unique_ptr<VoiceMediaChannel> channel,
const std::string& content_name,
bool srtp_required,
- rtc::CryptoOptions crypto_options);
+ webrtc::CryptoOptions crypto_options);
~VoiceChannel();
// downcasts a MediaChannel
@@ -383,7 +383,7 @@
std::unique_ptr<VideoMediaChannel> media_channel,
const std::string& content_name,
bool srtp_required,
- rtc::CryptoOptions crypto_options);
+ webrtc::CryptoOptions crypto_options);
~VideoChannel();
// downcasts a MediaChannel
@@ -422,7 +422,7 @@
std::unique_ptr<DataMediaChannel> channel,
const std::string& content_name,
bool srtp_required,
- rtc::CryptoOptions crypto_options);
+ webrtc::CryptoOptions crypto_options);
~RtpDataChannel();
// TODO(zhihuang): Remove this once the RtpTransport can be shared between
// BaseChannels.
diff --git a/pc/channel_unittest.cc b/pc/channel_unittest.cc
index 58cb17b..94f38c9 100644
--- a/pc/channel_unittest.cc
+++ b/pc/channel_unittest.cc
@@ -250,7 +250,7 @@
rtc::Thread* signaling_thread = rtc::Thread::Current();
auto channel = absl::make_unique<typename T::Channel>(
worker_thread, network_thread, signaling_thread, engine, std::move(ch),
- cricket::CN_AUDIO, (flags & DTLS) != 0, rtc::CryptoOptions());
+ cricket::CN_AUDIO, (flags & DTLS) != 0, webrtc::CryptoOptions());
channel->Init_w(rtp_transport);
return channel;
}
@@ -1545,7 +1545,7 @@
rtc::Thread* signaling_thread = rtc::Thread::Current();
auto channel = absl::make_unique<cricket::VideoChannel>(
worker_thread, network_thread, signaling_thread, std::move(ch),
- cricket::CN_VIDEO, (flags & DTLS) != 0, rtc::CryptoOptions());
+ cricket::CN_VIDEO, (flags & DTLS) != 0, webrtc::CryptoOptions());
channel->Init_w(rtp_transport);
return channel;
}
@@ -2164,7 +2164,7 @@
rtc::Thread* signaling_thread = rtc::Thread::Current();
auto channel = absl::make_unique<cricket::RtpDataChannel>(
worker_thread, network_thread, signaling_thread, std::move(ch),
- cricket::CN_DATA, (flags & DTLS) != 0, rtc::CryptoOptions());
+ cricket::CN_DATA, (flags & DTLS) != 0, webrtc::CryptoOptions());
channel->Init_w(rtp_transport);
return channel;
}
diff --git a/pc/channelmanager.cc b/pc/channelmanager.cc
index dab622a..1c80719 100644
--- a/pc/channelmanager.cc
+++ b/pc/channelmanager.cc
@@ -159,7 +159,7 @@
rtc::Thread* signaling_thread,
const std::string& content_name,
bool srtp_required,
- const rtc::CryptoOptions& crypto_options,
+ const webrtc::CryptoOptions& crypto_options,
const AudioOptions& options) {
if (!worker_thread_->IsCurrent()) {
return worker_thread_->Invoke<VoiceChannel*>(RTC_FROM_HERE, [&] {
@@ -226,7 +226,7 @@
rtc::Thread* signaling_thread,
const std::string& content_name,
bool srtp_required,
- const rtc::CryptoOptions& crypto_options,
+ const webrtc::CryptoOptions& crypto_options,
const VideoOptions& options) {
if (!worker_thread_->IsCurrent()) {
return worker_thread_->Invoke<VideoChannel*>(RTC_FROM_HERE, [&] {
@@ -291,7 +291,7 @@
rtc::Thread* signaling_thread,
const std::string& content_name,
bool srtp_required,
- const rtc::CryptoOptions& crypto_options) {
+ const webrtc::CryptoOptions& crypto_options) {
if (!worker_thread_->IsCurrent()) {
return worker_thread_->Invoke<RtpDataChannel*>(RTC_FROM_HERE, [&] {
return CreateRtpDataChannel(media_config, rtp_transport, signaling_thread,
diff --git a/pc/channelmanager.h b/pc/channelmanager.h
index c6b601e..6430f8e 100644
--- a/pc/channelmanager.h
+++ b/pc/channelmanager.h
@@ -86,7 +86,7 @@
rtc::Thread* signaling_thread,
const std::string& content_name,
bool srtp_required,
- const rtc::CryptoOptions& crypto_options,
+ const webrtc::CryptoOptions& crypto_options,
const AudioOptions& options);
// Destroys a voice channel created by CreateVoiceChannel.
void DestroyVoiceChannel(VoiceChannel* voice_channel);
@@ -100,7 +100,7 @@
rtc::Thread* signaling_thread,
const std::string& content_name,
bool srtp_required,
- const rtc::CryptoOptions& crypto_options,
+ const webrtc::CryptoOptions& crypto_options,
const VideoOptions& options);
// Destroys a video channel created by CreateVideoChannel.
void DestroyVideoChannel(VideoChannel* video_channel);
@@ -111,7 +111,7 @@
rtc::Thread* signaling_thread,
const std::string& content_name,
bool srtp_required,
- const rtc::CryptoOptions& crypto_options);
+ const webrtc::CryptoOptions& crypto_options);
// Destroys a data channel created by CreateRtpDataChannel.
void DestroyRtpDataChannel(RtpDataChannel* data_channel);
diff --git a/pc/channelmanager_unittest.cc b/pc/channelmanager_unittest.cc
index 47c9530..053166b 100644
--- a/pc/channelmanager_unittest.cc
+++ b/pc/channelmanager_unittest.cc
@@ -65,16 +65,16 @@
cricket::VoiceChannel* voice_channel = cm_->CreateVoiceChannel(
&fake_call_, cricket::MediaConfig(), rtp_transport,
rtc::Thread::Current(), cricket::CN_AUDIO, kDefaultSrtpRequired,
- rtc::CryptoOptions(), AudioOptions());
+ webrtc::CryptoOptions(), AudioOptions());
EXPECT_TRUE(voice_channel != nullptr);
cricket::VideoChannel* video_channel = cm_->CreateVideoChannel(
&fake_call_, cricket::MediaConfig(), rtp_transport,
rtc::Thread::Current(), cricket::CN_VIDEO, kDefaultSrtpRequired,
- rtc::CryptoOptions(), VideoOptions());
+ webrtc::CryptoOptions(), VideoOptions());
EXPECT_TRUE(video_channel != nullptr);
cricket::RtpDataChannel* rtp_data_channel = cm_->CreateRtpDataChannel(
cricket::MediaConfig(), rtp_transport, rtc::Thread::Current(),
- cricket::CN_DATA, kDefaultSrtpRequired, rtc::CryptoOptions());
+ cricket::CN_DATA, kDefaultSrtpRequired, webrtc::CryptoOptions());
EXPECT_TRUE(rtp_data_channel != nullptr);
cm_->DestroyVideoChannel(video_channel);
cm_->DestroyVoiceChannel(voice_channel);
diff --git a/pc/jseptransportcontroller.cc b/pc/jseptransportcontroller.cc
index 1847d95..19a2d40 100644
--- a/pc/jseptransportcontroller.cc
+++ b/pc/jseptransportcontroller.cc
@@ -814,7 +814,7 @@
static_cast<const cricket::MediaContentDescription*>(
content_info.description);
- if (!config_.crypto_options.enable_encrypted_rtp_header_extensions) {
+ if (!config_.crypto_options.srtp.enable_encrypted_rtp_header_extensions) {
return std::vector<int>();
}
diff --git a/pc/jseptransportcontroller.h b/pc/jseptransportcontroller.h
index bca4481..518d310 100644
--- a/pc/jseptransportcontroller.h
+++ b/pc/jseptransportcontroller.h
@@ -18,6 +18,7 @@
#include <vector>
#include "api/candidate.h"
+#include "api/crypto/cryptooptions.h"
#include "api/media_transport_interface.h"
#include "api/peerconnectioninterface.h"
#include "logging/rtc_event_log/rtc_event_log.h"
@@ -33,7 +34,6 @@
#include "rtc_base/asyncinvoker.h"
#include "rtc_base/constructormagic.h"
#include "rtc_base/refcountedobject.h"
-#include "rtc_base/sslstreamadapter.h"
#include "rtc_base/third_party/sigslot/sigslot.h"
namespace rtc {
@@ -68,7 +68,7 @@
rtc::SSLProtocolVersion ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
// |crypto_options| is used to determine if created DTLS transports
// negotiate GCM crypto suites or not.
- rtc::CryptoOptions crypto_options;
+ webrtc::CryptoOptions crypto_options;
PeerConnectionInterface::BundlePolicy bundle_policy =
PeerConnectionInterface::kBundlePolicyBalanced;
PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy =
diff --git a/pc/jseptransportcontroller_unittest.cc b/pc/jseptransportcontroller_unittest.cc
index ce431c3..26813e1 100644
--- a/pc/jseptransportcontroller_unittest.cc
+++ b/pc/jseptransportcontroller_unittest.cc
@@ -52,7 +52,7 @@
std::unique_ptr<cricket::DtlsTransportInternal> CreateDtlsTransport(
std::unique_ptr<cricket::IceTransportInternal> ice,
- const rtc::CryptoOptions& crypto_options) override {
+ const webrtc::CryptoOptions& crypto_options) override {
std::unique_ptr<cricket::FakeIceTransport> fake_ice(
static_cast<cricket::FakeIceTransport*>(ice.release()));
return absl::make_unique<FakeDtlsTransport>(std::move(fake_ice));
diff --git a/pc/mediasession.cc b/pc/mediasession.cc
index d186513..889576c 100644
--- a/pc/mediasession.cc
+++ b/pc/mediasession.cc
@@ -39,10 +39,10 @@
const char kInline[] = "inline:";
-void GetSupportedSdesCryptoSuiteNames(void (*func)(const rtc::CryptoOptions&,
- std::vector<int>*),
- const rtc::CryptoOptions& crypto_options,
- std::vector<std::string>* names) {
+void GetSupportedSdesCryptoSuiteNames(
+ void (*func)(const webrtc::CryptoOptions&, std::vector<int>*),
+ const webrtc::CryptoOptions& crypto_options,
+ std::vector<std::string>* names) {
std::vector<int> crypto_suites;
func(crypto_options, &crypto_suites);
for (const auto crypto : crypto_suites) {
@@ -195,28 +195,30 @@
// For audio, HMAC 32 (if enabled) is prefered over HMAC 80 because of the
// low overhead.
-void GetSupportedAudioSdesCryptoSuites(const rtc::CryptoOptions& crypto_options,
- std::vector<int>* crypto_suites) {
- if (crypto_options.enable_gcm_crypto_suites) {
+void GetSupportedAudioSdesCryptoSuites(
+ const webrtc::CryptoOptions& crypto_options,
+ std::vector<int>* crypto_suites) {
+ if (crypto_options.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);
}
- if (crypto_options.enable_aes128_sha1_32_crypto_cipher) {
+ if (crypto_options.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);
}
void GetSupportedAudioSdesCryptoSuiteNames(
- const rtc::CryptoOptions& crypto_options,
+ const webrtc::CryptoOptions& crypto_options,
std::vector<std::string>* crypto_suite_names) {
GetSupportedSdesCryptoSuiteNames(GetSupportedAudioSdesCryptoSuites,
crypto_options, crypto_suite_names);
}
-void GetSupportedVideoSdesCryptoSuites(const rtc::CryptoOptions& crypto_options,
- std::vector<int>* crypto_suites) {
- if (crypto_options.enable_gcm_crypto_suites) {
+void GetSupportedVideoSdesCryptoSuites(
+ const webrtc::CryptoOptions& crypto_options,
+ std::vector<int>* crypto_suites) {
+ if (crypto_options.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);
}
@@ -224,15 +226,16 @@
}
void GetSupportedVideoSdesCryptoSuiteNames(
- const rtc::CryptoOptions& crypto_options,
+ const webrtc::CryptoOptions& crypto_options,
std::vector<std::string>* crypto_suite_names) {
GetSupportedSdesCryptoSuiteNames(GetSupportedVideoSdesCryptoSuites,
crypto_options, crypto_suite_names);
}
-void GetSupportedDataSdesCryptoSuites(const rtc::CryptoOptions& crypto_options,
- std::vector<int>* crypto_suites) {
- if (crypto_options.enable_gcm_crypto_suites) {
+void GetSupportedDataSdesCryptoSuites(
+ const webrtc::CryptoOptions& crypto_options,
+ std::vector<int>* crypto_suites) {
+ if (crypto_options.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);
}
@@ -240,7 +243,7 @@
}
void GetSupportedDataSdesCryptoSuiteNames(
- const rtc::CryptoOptions& crypto_options,
+ const webrtc::CryptoOptions& crypto_options,
std::vector<std::string>* crypto_suite_names) {
GetSupportedSdesCryptoSuiteNames(GetSupportedDataSdesCryptoSuites,
crypto_options, crypto_suite_names);
@@ -252,17 +255,17 @@
// Pick the crypto in the list that is supported.
static bool SelectCrypto(const MediaContentDescription* offer,
bool bundle,
- const rtc::CryptoOptions& crypto_options,
+ const webrtc::CryptoOptions& crypto_options,
CryptoParams* crypto_out) {
bool audio = offer->type() == MEDIA_TYPE_AUDIO;
const CryptoParamsVec& cryptos = offer->cryptos();
for (const CryptoParams& crypto : cryptos) {
- if ((crypto_options.enable_gcm_crypto_suites &&
+ if ((crypto_options.srtp.enable_gcm_crypto_suites &&
rtc::IsGcmCryptoSuiteName(crypto.cipher_suite)) ||
rtc::CS_AES_CM_128_HMAC_SHA1_80 == crypto.cipher_suite ||
(rtc::CS_AES_CM_128_HMAC_SHA1_32 == crypto.cipher_suite && audio &&
- !bundle && crypto_options.enable_aes128_sha1_32_crypto_cipher)) {
+ !bundle && crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher)) {
return CreateCryptoParams(crypto.tag, crypto.cipher_suite, crypto_out);
}
}
diff --git a/pc/mediasession.h b/pc/mediasession.h
index 81b88e1..b1df1db 100644
--- a/pc/mediasession.h
+++ b/pc/mediasession.h
@@ -95,7 +95,7 @@
bool bundle_enabled = false;
bool is_unified_plan = false;
std::string rtcp_cname = kDefaultRtcpCname;
- rtc::CryptoOptions crypto_options;
+ webrtc::CryptoOptions crypto_options;
// List of media description options in the same order that the media
// descriptions will be generated.
std::vector<MediaDescriptionOptions> media_description_options;
@@ -337,20 +337,23 @@
SessionDescription* sdesc);
// Helper functions to return crypto suites used for SDES.
-void GetSupportedAudioSdesCryptoSuites(const rtc::CryptoOptions& crypto_options,
- std::vector<int>* crypto_suites);
-void GetSupportedVideoSdesCryptoSuites(const rtc::CryptoOptions& crypto_options,
- std::vector<int>* crypto_suites);
-void GetSupportedDataSdesCryptoSuites(const rtc::CryptoOptions& crypto_options,
- std::vector<int>* crypto_suites);
+void GetSupportedAudioSdesCryptoSuites(
+ const webrtc::CryptoOptions& crypto_options,
+ std::vector<int>* crypto_suites);
+void GetSupportedVideoSdesCryptoSuites(
+ const webrtc::CryptoOptions& crypto_options,
+ std::vector<int>* crypto_suites);
+void GetSupportedDataSdesCryptoSuites(
+ const webrtc::CryptoOptions& crypto_options,
+ std::vector<int>* crypto_suites);
void GetSupportedAudioSdesCryptoSuiteNames(
- const rtc::CryptoOptions& crypto_options,
+ const webrtc::CryptoOptions& crypto_options,
std::vector<std::string>* crypto_suite_names);
void GetSupportedVideoSdesCryptoSuiteNames(
- const rtc::CryptoOptions& crypto_options,
+ const webrtc::CryptoOptions& crypto_options,
std::vector<std::string>* crypto_suite_names);
void GetSupportedDataSdesCryptoSuiteNames(
- const rtc::CryptoOptions& crypto_options,
+ const webrtc::CryptoOptions& crypto_options,
std::vector<std::string>* crypto_suite_names);
// Returns true if the given media section protocol indicates use of RTP.
diff --git a/pc/mediasession_unittest.cc b/pc/mediasession_unittest.cc
index 7162a37..3bc2766 100644
--- a/pc/mediasession_unittest.cc
+++ b/pc/mediasession_unittest.cc
@@ -609,11 +609,11 @@
void TestVideoGcmCipher(bool gcm_offer, bool gcm_answer) {
MediaSessionOptions offer_opts;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &offer_opts);
- offer_opts.crypto_options.enable_gcm_crypto_suites = gcm_offer;
+ offer_opts.crypto_options.srtp.enable_gcm_crypto_suites = gcm_offer;
MediaSessionOptions answer_opts;
AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &answer_opts);
- answer_opts.crypto_options.enable_gcm_crypto_suites = gcm_answer;
+ answer_opts.crypto_options.srtp.enable_gcm_crypto_suites = gcm_answer;
f1_.set_secure(SEC_ENABLED);
f2_.set_secure(SEC_ENABLED);
@@ -953,7 +953,7 @@
f1_.set_secure(SEC_ENABLED);
f2_.set_secure(SEC_ENABLED);
MediaSessionOptions opts = CreatePlanBMediaSessionOptions();
- opts.crypto_options.enable_gcm_crypto_suites = true;
+ opts.crypto_options.srtp.enable_gcm_crypto_suites = true;
std::unique_ptr<SessionDescription> offer(f1_.CreateOffer(opts, NULL));
ASSERT_TRUE(offer.get() != NULL);
std::unique_ptr<SessionDescription> answer(
@@ -1057,7 +1057,7 @@
TEST_F(MediaSessionDescriptionFactoryTest, TestCreateDataAnswerGcm) {
MediaSessionOptions opts = CreatePlanBMediaSessionOptions();
AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly, &opts);
- opts.crypto_options.enable_gcm_crypto_suites = true;
+ opts.crypto_options.srtp.enable_gcm_crypto_suites = true;
f1_.set_secure(SEC_ENABLED);
f2_.set_secure(SEC_ENABLED);
std::unique_ptr<SessionDescription> offer(f1_.CreateOffer(opts, NULL));
diff --git a/pc/peerconnection.cc b/pc/peerconnection.cc
index 5cca4a8..0861246 100644
--- a/pc/peerconnection.cc
+++ b/pc/peerconnection.cc
@@ -1027,7 +1027,7 @@
}
webrtc_session_desc_factory_->set_enable_encrypted_rtp_header_extensions(
- options.crypto_options.enable_encrypted_rtp_header_extensions);
+ options.crypto_options.srtp.enable_encrypted_rtp_header_extensions);
// Add default audio/video transceivers for Plan B SDP.
if (!IsUnifiedPlan()) {
diff --git a/pc/peerconnection_crypto_unittest.cc b/pc/peerconnection_crypto_unittest.cc
index 1ccfe55..71e8763 100644
--- a/pc/peerconnection_crypto_unittest.cc
+++ b/pc/peerconnection_crypto_unittest.cc
@@ -284,7 +284,7 @@
// in the answer.
TEST_P(PeerConnectionCryptoTest, CorrectCryptoInOfferWithSdesAndGcm) {
PeerConnectionFactoryInterface::Options options;
- options.crypto_options.enable_gcm_crypto_suites = true;
+ options.crypto_options.srtp.enable_gcm_crypto_suites = true;
pc_factory_->SetOptions(options);
RTCConfiguration config;
@@ -299,7 +299,7 @@
}
TEST_P(PeerConnectionCryptoTest, CorrectCryptoInAnswerWithSdesAndGcm) {
PeerConnectionFactoryInterface::Options options;
- options.crypto_options.enable_gcm_crypto_suites = true;
+ options.crypto_options.srtp.enable_gcm_crypto_suites = true;
pc_factory_->SetOptions(options);
RTCConfiguration config;
@@ -317,7 +317,7 @@
TEST_P(PeerConnectionCryptoTest, CanSetSdesGcmRemoteOfferAndLocalAnswer) {
PeerConnectionFactoryInterface::Options options;
- options.crypto_options.enable_gcm_crypto_suites = true;
+ options.crypto_options.srtp.enable_gcm_crypto_suites = true;
pc_factory_->SetOptions(options);
RTCConfiguration config;
diff --git a/pc/peerconnection_integrationtest.cc b/pc/peerconnection_integrationtest.cc
index 5825898..36a832c 100644
--- a/pc/peerconnection_integrationtest.cc
+++ b/pc/peerconnection_integrationtest.cc
@@ -1558,9 +1558,11 @@
bool remote_gcm_enabled,
int expected_cipher_suite) {
PeerConnectionFactory::Options caller_options;
- caller_options.crypto_options.enable_gcm_crypto_suites = local_gcm_enabled;
+ caller_options.crypto_options.srtp.enable_gcm_crypto_suites =
+ local_gcm_enabled;
PeerConnectionFactory::Options callee_options;
- callee_options.crypto_options.enable_gcm_crypto_suites = remote_gcm_enabled;
+ callee_options.crypto_options.srtp.enable_gcm_crypto_suites =
+ remote_gcm_enabled;
TestNegotiatedCipherSuite(caller_options, callee_options,
expected_cipher_suite);
}
@@ -2843,9 +2845,10 @@
TEST_P(PeerConnectionIntegrationTest,
Aes128Sha1_32_CipherNotUsedWhenOnlyCallerSupported) {
PeerConnectionFactory::Options caller_options;
- caller_options.crypto_options.enable_aes128_sha1_32_crypto_cipher = true;
+ caller_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher = true;
PeerConnectionFactory::Options callee_options;
- callee_options.crypto_options.enable_aes128_sha1_32_crypto_cipher = false;
+ callee_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher =
+ false;
int expected_cipher_suite = rtc::SRTP_AES128_CM_SHA1_80;
TestNegotiatedCipherSuite(caller_options, callee_options,
expected_cipher_suite);
@@ -2854,9 +2857,10 @@
TEST_P(PeerConnectionIntegrationTest,
Aes128Sha1_32_CipherNotUsedWhenOnlyCalleeSupported) {
PeerConnectionFactory::Options caller_options;
- caller_options.crypto_options.enable_aes128_sha1_32_crypto_cipher = false;
+ caller_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher =
+ false;
PeerConnectionFactory::Options callee_options;
- callee_options.crypto_options.enable_aes128_sha1_32_crypto_cipher = true;
+ callee_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher = true;
int expected_cipher_suite = rtc::SRTP_AES128_CM_SHA1_80;
TestNegotiatedCipherSuite(caller_options, callee_options,
expected_cipher_suite);
@@ -2864,9 +2868,9 @@
TEST_P(PeerConnectionIntegrationTest, Aes128Sha1_32_CipherUsedWhenSupported) {
PeerConnectionFactory::Options caller_options;
- caller_options.crypto_options.enable_aes128_sha1_32_crypto_cipher = true;
+ caller_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher = true;
PeerConnectionFactory::Options callee_options;
- callee_options.crypto_options.enable_aes128_sha1_32_crypto_cipher = true;
+ callee_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher = true;
int expected_cipher_suite = rtc::SRTP_AES128_CM_SHA1_32;
TestNegotiatedCipherSuite(caller_options, callee_options,
expected_cipher_suite);
@@ -2916,7 +2920,7 @@
// works with it.
TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithGcmCipher) {
PeerConnectionFactory::Options gcm_options;
- gcm_options.crypto_options.enable_gcm_crypto_suites = true;
+ gcm_options.crypto_options.srtp.enable_gcm_crypto_suites = true;
ASSERT_TRUE(
CreatePeerConnectionWrappersWithOptions(gcm_options, gcm_options));
ConnectFakeSignaling();
diff --git a/pc/peerconnectionfactory.cc b/pc/peerconnectionfactory.cc
index edc13034..fd4ec0b 100644
--- a/pc/peerconnectionfactory.cc
+++ b/pc/peerconnectionfactory.cc
@@ -234,6 +234,21 @@
void PeerConnectionFactory::SetOptions(const Options& options) {
options_ = options;
+ // TODO(webrtc:9859) - Remove Chromium Compatibility once fix lands in
+ // Chromium
+ if (options.crypto_options.enable_gcm_crypto_suites.has_value()) {
+ options_.crypto_options.srtp.enable_gcm_crypto_suites =
+ *options.crypto_options.enable_gcm_crypto_suites;
+ }
+ if (options.crypto_options.enable_encrypted_rtp_header_extensions
+ .has_value()) {
+ options_.crypto_options.srtp.enable_encrypted_rtp_header_extensions =
+ *options.crypto_options.enable_encrypted_rtp_header_extensions;
+ }
+ if (options.crypto_options.enable_aes128_sha1_32_crypto_cipher.has_value()) {
+ options_.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher =
+ *options.crypto_options.enable_aes128_sha1_32_crypto_cipher;
+ }
}
RtpCapabilities PeerConnectionFactory::GetRtpSenderCapabilities(
diff --git a/pc/rtpsenderreceiver_unittest.cc b/pc/rtpsenderreceiver_unittest.cc
index 07ff6a3..fad839c 100644
--- a/pc/rtpsenderreceiver_unittest.cc
+++ b/pc/rtpsenderreceiver_unittest.cc
@@ -81,11 +81,11 @@
voice_channel_ = channel_manager_.CreateVoiceChannel(
&fake_call_, cricket::MediaConfig(), rtp_transport_.get(),
rtc::Thread::Current(), cricket::CN_AUDIO, srtp_required,
- rtc::CryptoOptions(), cricket::AudioOptions());
+ webrtc::CryptoOptions(), cricket::AudioOptions());
video_channel_ = channel_manager_.CreateVideoChannel(
&fake_call_, cricket::MediaConfig(), rtp_transport_.get(),
rtc::Thread::Current(), cricket::CN_VIDEO, srtp_required,
- rtc::CryptoOptions(), cricket::VideoOptions());
+ webrtc::CryptoOptions(), cricket::VideoOptions());
voice_channel_->Enable(true);
video_channel_->Enable(true);
voice_media_channel_ = media_engine_->GetVoiceChannel(0);
diff --git a/pc/test/fakepeerconnectionforstats.h b/pc/test/fakepeerconnectionforstats.h
index 642fa01..ae329e4 100644
--- a/pc/test/fakepeerconnectionforstats.h
+++ b/pc/test/fakepeerconnectionforstats.h
@@ -126,7 +126,7 @@
voice_channel_ = absl::make_unique<cricket::VoiceChannel>(
worker_thread_, network_thread_, signaling_thread_, nullptr,
std::move(voice_media_channel), mid, kDefaultSrtpRequired,
- rtc::CryptoOptions());
+ webrtc::CryptoOptions());
voice_channel_->set_transport_name_for_testing(transport_name);
GetOrCreateFirstTransceiverOfType(cricket::MEDIA_TYPE_AUDIO)
->internal()
@@ -144,7 +144,7 @@
video_channel_ = absl::make_unique<cricket::VideoChannel>(
worker_thread_, network_thread_, signaling_thread_,
std::move(video_media_channel), mid, kDefaultSrtpRequired,
- rtc::CryptoOptions());
+ webrtc::CryptoOptions());
video_channel_->set_transport_name_for_testing(transport_name);
GetOrCreateFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)
->internal()
diff --git a/rtc_base/sslstreamadapter.cc b/rtc_base/sslstreamadapter.cc
index 746ebd5..fe42bf4 100644
--- a/rtc_base/sslstreamadapter.cc
+++ b/rtc_base/sslstreamadapter.cc
@@ -89,32 +89,6 @@
crypto_suite == CS_AEAD_AES_128_GCM);
}
-// static
-CryptoOptions CryptoOptions::NoGcm() {
- CryptoOptions options;
- options.enable_gcm_crypto_suites = false;
- return options;
-}
-
-std::vector<int> GetSupportedDtlsSrtpCryptoSuites(
- const rtc::CryptoOptions& crypto_options) {
- std::vector<int> crypto_suites;
- if (crypto_options.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 (crypto_options.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;
-}
-
SSLStreamAdapter* SSLStreamAdapter::Create(StreamInterface* stream) {
return new OpenSSLStreamAdapter(stream);
}
diff --git a/rtc_base/sslstreamadapter.h b/rtc_base/sslstreamadapter.h
index 2d4e19f..dad8a4e 100644
--- a/rtc_base/sslstreamadapter.h
+++ b/rtc_base/sslstreamadapter.h
@@ -70,34 +70,6 @@
// Returns true if the given crypto suite name uses a GCM cipher.
bool IsGcmCryptoSuiteName(const std::string& crypto_suite);
-struct CryptoOptions {
- CryptoOptions() {}
-
- // Helper method to return an instance of the CryptoOptions with GCM crypto
- // suites disabled. This method should be used instead of depending on current
- // default values set by the constructor.
- static CryptoOptions NoGcm();
-
- // Enable GCM crypto suites from RFC 7714 for SRTP. GCM will only be used
- // if both sides enable it.
- bool enable_gcm_crypto_suites = false;
-
- // If set to true, the (potentially insecure) crypto cipher
- // SRTP_AES128_CM_SHA1_32 will be included in the list of supported ciphers
- // during negotiation. It will only be used if both peers support it and no
- // other ciphers get preferred.
- bool enable_aes128_sha1_32_crypto_cipher = false;
-
- // If set to true, encrypted RTP header extensions as defined in RFC 6904
- // will be negotiated. They will only be used if both peers support them.
- bool enable_encrypted_rtp_header_extensions = false;
-};
-
-// Returns supported crypto suites, given |crypto_options|.
-// CS_AES_CM_128_HMAC_SHA1_32 will be preferred by default.
-std::vector<int> GetSupportedDtlsSrtpCryptoSuites(
- const rtc::CryptoOptions& crypto_options);
-
// SSLStreamAdapter : A StreamInterfaceAdapter that does SSL/TLS.
// After SSL has been started, the stream will only open on successful
// SSL verification of certificates, and the communication is
diff --git a/sdk/android/src/jni/pc/peerconnectionfactory.cc b/sdk/android/src/jni/pc/peerconnectionfactory.cc
index 01c1acc..049858a 100644
--- a/sdk/android/src/jni/pc/peerconnectionfactory.cc
+++ b/sdk/android/src/jni/pc/peerconnectionfactory.cc
@@ -63,9 +63,9 @@
native_options.disable_encryption = disable_encryption;
native_options.disable_network_monitor = disable_network_monitor;
- native_options.crypto_options.enable_aes128_sha1_32_crypto_cipher =
+ native_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher =
enable_aes128_sha1_32_crypto_cipher;
- native_options.crypto_options.enable_gcm_crypto_suites =
+ native_options.crypto_options.srtp.enable_gcm_crypto_suites =
enable_gcm_crypto_suites;
return native_options;
}
diff --git a/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryOptions.mm b/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryOptions.mm
index 103a130..1690385 100644
--- a/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryOptions.mm
+++ b/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryOptions.mm
@@ -52,8 +52,9 @@
setNetworkBit(&options, rtc::ADAPTER_TYPE_WIFI, self.ignoreWiFiNetworkAdapter);
setNetworkBit(&options, rtc::ADAPTER_TYPE_ETHERNET, self.ignoreEthernetNetworkAdapter);
- options.crypto_options.enable_aes128_sha1_32_crypto_cipher = self.enableAes128Sha1_32CryptoCipher;
- options.crypto_options.enable_gcm_crypto_suites = self.enableGcmCryptoSuites;
+ options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher =
+ self.enableAes128Sha1_32CryptoCipher;
+ options.crypto_options.srtp.enable_gcm_crypto_suites = self.enableGcmCryptoSuites;
return options;
}
