webrtc_m130/api/crypto/cryptooptions.h
Benjamin Wright bfb444ce2c Adds new CryptoOption crypto_options.frame.require_frame_encryption.
This change adds a new subcategory to the public native webrtc::CryptoOptions
structure: webrtc::CryptoOptions::Frame.

This new structure has a single off by default property:
crypto_options.frame.require_frame_encryption.

This new flag if set prevents RtpSenders from sending outgoing payloads unless
a frame_encryptor_ is attached and prevents RtpReceivers from receiving
incoming payloads unless a frame_decryptor_ is attached.

This option is important to enforce no unencrypted data can ever leave the
device or be received.

I have also attached bindings for Java and Objective-C.

I have implemented this functionality for E2EE audio but not E2EE video
since the changes are still in review.

Bug: webrtc:9681
Change-Id: Ie184711190e0cdf5ac781f69e9489ceec904736f
Reviewed-on: https://webrtc-review.googlesource.com/c/105540
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25238}
2018-10-17 17:44:19 +00:00

68 lines
2.5 KiB
C++

/*
* 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;
bool operator==(const CryptoOptions& other) const;
bool operator!=(const CryptoOptions& other) const;
// 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;
// Options to be used when the FrameEncryptor / FrameDecryptor APIs are used.
struct SFrame {
// If set all RtpSenders must have an FrameEncryptor attached to them before
// they are allowed to send packets. All RtpReceivers must have a
// FrameDecryptor attached to them before they are able to receive packets.
bool require_frame_encryption = false;
} sframe;
};
} // namespace webrtc
#endif // API_CRYPTO_CRYPTOOPTIONS_H_