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Reland "Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h"

Browse Source 
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}
This commit is contained in:
Benjamin Wright 2018-10-11 15:33:17 -07:00 committed by Commit Bot
parent edd204ed4d
commit a54daf162f
33 changed files with 252 additions and 153 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

2
api/BUILD.gn
View File

@ -52,6 +52,8 @@ rtc_static_library("libjingle_peerconnection_api") {
"bitrate_constraints.h",
"candidate.cc",
"candidate.h",
"crypto/cryptooptions.cc",
"crypto/cryptooptions.h",
"crypto/framedecryptorinterface.h",
"crypto/frameencryptorinterface.h",
"cryptoparams.h",

52
api/crypto/cryptooptions.cc Normal file
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@ -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

67
api/crypto/cryptooptions.h Normal file
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@ -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_

2
api/cryptoparams.h
View File

@ -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,

5
api/peerconnectioninterface.h
View File

@ -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 @@ class PeerConnectionFactoryInterface : public rtc::RefCountInterface {
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 @@ class PeerConnectionFactoryInterface : public rtc::RefCountInterface {
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.

10
p2p/base/dtlstransport.cc
View File

@ -114,12 +114,12 @@ void StreamInterfaceChannel::Close() {
}
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(IceTransportInternal* ice_transport,
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(
DtlsTransport::~DtlsTransport() = default;
const rtc::CryptoOptions& DtlsTransport::crypto_options() const {
const webrtc::CryptoOptions& DtlsTransport::crypto_options() const {
return crypto_options_;
}

9
p2p/base/dtlstransport.h
View File

@ -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 @@ class DtlsTransport : public DtlsTransportInternal {
// 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 @@ class DtlsTransport : public DtlsTransportInternal {
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_;

4
p2p/base/dtlstransport_unittest.cc
View File

@ -86,8 +86,8 @@ class DtlsTestClient : public sigslot::has_slots<> {
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_);

3
p2p/base/dtlstransportinternal.h
View File

@ -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 @@ class DtlsTransportInternal : public rtc::PacketTransportInternal {
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;

7
p2p/base/fakedtlstransport.h
View File

@ -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 @@ class FakeDtlsTransport : public DtlsTransportInternal {
*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 @@ class FakeDtlsTransport : public DtlsTransportInternal {
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;

2
p2p/base/transportfactoryinterface.h
View File

@ -34,7 +34,7 @@ class TransportFactoryInterface {
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

10
pc/channel.cc
View File

@ -101,7 +101,7 @@ BaseChannel::BaseChannel(rtc::Thread* worker_thread,
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 @@ bool BaseChannel::UpdateRemoteStreams_w(
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 @@ VoiceChannel::VoiceChannel(rtc::Thread* worker_thread,
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 @@ VideoChannel::VideoChannel(rtc::Thread* worker_thread,
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 @@ RtpDataChannel::RtpDataChannel(rtc::Thread* worker_thread,
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,

10
pc/channel.h
View File

@ -82,7 +82,7 @@ class BaseChannel : public rtc::MessageHandler,
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 @@ class BaseChannel : public rtc::MessageHandler,
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 @@ class VoiceChannel : public BaseChannel {
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 @@ class VideoChannel : public BaseChannel {
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 @@ class RtpDataChannel : public BaseChannel {
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.

6
pc/channel_unittest.cc
View File

@ -250,7 +250,7 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
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 @@ std::unique_ptr<cricket::VideoChannel> ChannelTest<VideoTraits>::CreateChannel(
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 @@ std::unique_ptr<cricket::RtpDataChannel> ChannelTest<DataTraits>::CreateChannel(
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;
}

6
pc/channelmanager.cc
View File

@ -159,7 +159,7 @@ VoiceChannel* ChannelManager::CreateVoiceChannel(
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 @@ VideoChannel* ChannelManager::CreateVideoChannel(
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 @@ RtpDataChannel* ChannelManager::CreateRtpDataChannel(
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,

6
pc/channelmanager.h
View File

@ -86,7 +86,7 @@ class ChannelManager final {
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 @@ class ChannelManager final {
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 @@ class ChannelManager final {
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);

6
pc/channelmanager_unittest.cc
View File

@ -65,16 +65,16 @@ class ChannelManagerTest : public testing::Test {
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);

2
pc/jseptransportcontroller.cc
View File

@ -814,7 +814,7 @@ std::vector<int> JsepTransportController::GetEncryptedHeaderExtensionIds(
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>();
}

4
pc/jseptransportcontroller.h
View File

@ -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 @@ class JsepTransportController : public sigslot::has_slots<> {
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 =

2
pc/jseptransportcontroller_unittest.cc
View File

@ -52,7 +52,7 @@ class FakeTransportFactory : public cricket::TransportFactoryInterface {
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));

35
pc/mediasession.cc
View File

@ -39,9 +39,9 @@ using webrtc::RtpTransceiverDirection;
const char kInline[] = "inline:";
void GetSupportedSdesCryptoSuiteNames(void (*func)(const rtc::CryptoOptions&,
std::vector<int>*),
const rtc::CryptoOptions& crypto_options,
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);
@ -195,28 +195,30 @@ bool FindMatchingCrypto(const CryptoParamsVec& cryptos,
// For audio, HMAC 32 (if enabled) is prefered over HMAC 80 because of the
// low overhead.
void GetSupportedAudioSdesCryptoSuites(const rtc::CryptoOptions& crypto_options,
void GetSupportedAudioSdesCryptoSuites(
const webrtc::CryptoOptions& crypto_options,
std::vector<int>* crypto_suites) {
if (crypto_options.enable_gcm_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,
void GetSupportedVideoSdesCryptoSuites(
const webrtc::CryptoOptions& crypto_options,
std::vector<int>* crypto_suites) {
if (crypto_options.enable_gcm_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 GetSupportedVideoSdesCryptoSuites(const rtc::CryptoOptions& crypto_options,
}
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,
void GetSupportedDataSdesCryptoSuites(
const webrtc::CryptoOptions& crypto_options,
std::vector<int>* crypto_suites) {
if (crypto_options.enable_gcm_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 GetSupportedDataSdesCryptoSuites(const rtc::CryptoOptions& crypto_options,
}
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 @@ void GetSupportedDataSdesCryptoSuiteNames(
// 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);
}
}

17
pc/mediasession.h
View File

@ -95,7 +95,7 @@ struct MediaSessionOptions {
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 @@ DataContentDescription* GetFirstDataContentDescription(
SessionDescription* sdesc);
// Helper functions to return crypto suites used for SDES.
void GetSupportedAudioSdesCryptoSuites(const rtc::CryptoOptions& crypto_options,
void GetSupportedAudioSdesCryptoSuites(
const webrtc::CryptoOptions& crypto_options,
std::vector<int>* crypto_suites);
void GetSupportedVideoSdesCryptoSuites(const rtc::CryptoOptions& crypto_options,
void GetSupportedVideoSdesCryptoSuites(
const webrtc::CryptoOptions& crypto_options,
std::vector<int>* crypto_suites);
void GetSupportedDataSdesCryptoSuites(const rtc::CryptoOptions& crypto_options,
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.

8
pc/mediasession_unittest.cc
View File

@ -609,11 +609,11 @@ class MediaSessionDescriptionFactoryTest : public testing::Test {
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 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateAudioAnswerGcm) {
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, TestCreateDataAnswer) {
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));

2
pc/peerconnection.cc
View File

@ -1027,7 +1027,7 @@ bool PeerConnection::Initialize(
}
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()) {

6
pc/peerconnection_crypto_unittest.cc
View File

@ -284,7 +284,7 @@ TEST_P(PeerConnectionCryptoTest, CorrectCryptoInAnswerWhenEncryptionDisabled) {
// 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, CorrectCryptoInOfferWithSdesAndGcm) {
}
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, CorrectCryptoInAnswerWithSdesAndGcm) {
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;

22
pc/peerconnection_integrationtest.cc
View File

@ -1558,9 +1558,11 @@ class PeerConnectionIntegrationBaseTest : public testing::Test {
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, CallerDtls10ToCalleeDtls12) {
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,
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,
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 @@ TEST_P(PeerConnectionIntegrationTest,
// 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();

15
pc/peerconnectionfactory.cc
View File

@ -234,6 +234,21 @@ bool PeerConnectionFactory::Initialize() {
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(

4
pc/rtpsenderreceiver_unittest.cc
View File

@ -81,11 +81,11 @@ class RtpSenderReceiverTest : public testing::Test,
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);

4
pc/test/fakepeerconnectionforstats.h
View File

@ -126,7 +126,7 @@ class FakePeerConnectionForStats : public FakePeerConnectionBase {
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 @@ class FakePeerConnectionForStats : public FakePeerConnectionBase {
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()

26
rtc_base/sslstreamadapter.cc
View File

@ -89,32 +89,6 @@ bool IsGcmCryptoSuiteName(const std::string& crypto_suite) {
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);
}

28
rtc_base/sslstreamadapter.h
View File

@ -70,34 +70,6 @@ bool IsGcmCryptoSuite(int crypto_suite);
// 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

4
sdk/android/src/jni/pc/peerconnectionfactory.cc
View File

@ -63,9 +63,9 @@ JavaToNativePeerConnectionFactoryOptions(JNIEnv* jni,
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;
}

5
sdk/objc/api/peerconnection/RTCPeerConnectionFactoryOptions.mm
View File

@ -52,8 +52,9 @@ void setNetworkBit(webrtc::PeerConnectionFactoryInterface::Options* options,
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;
}