blob: 317995134aaa9e80acf3c1458c43bb889c946992 (
plain)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
|
/*
* 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_CRYPTO_OPTIONS_H_
#define API_CRYPTO_CRYPTO_OPTIONS_H_
#include <vector>
#include "rtc_base/system/rtc_export.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 RTC_EXPORT 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
// kSrtpAes128CmSha1_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;
// The most commonly used cipher. Can be disabled, mostly for testing
// purposes.
bool enable_aes128_sha1_80_crypto_cipher = true;
// 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_CRYPTO_OPTIONS_H_
|
