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Anders Carlsson7bca8ca2018-08-30 09:30:29 +02001/*
2 * Copyright 2015 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11#import <Foundation/Foundation.h>
12
13#import "RTCCertificate.h"
14#import "RTCMacros.h"
15
16@class RTCIceServer;
17@class RTCIntervalRange;
18
19/**
20 * Represents the ice transport policy. This exposes the same states in C++,
21 * which include one more state than what exists in the W3C spec.
22 */
23typedef NS_ENUM(NSInteger, RTCIceTransportPolicy) {
24 RTCIceTransportPolicyNone,
25 RTCIceTransportPolicyRelay,
26 RTCIceTransportPolicyNoHost,
27 RTCIceTransportPolicyAll
28};
29
30/** Represents the bundle policy. */
31typedef NS_ENUM(NSInteger, RTCBundlePolicy) {
32 RTCBundlePolicyBalanced,
33 RTCBundlePolicyMaxCompat,
34 RTCBundlePolicyMaxBundle
35};
36
37/** Represents the rtcp mux policy. */
38typedef NS_ENUM(NSInteger, RTCRtcpMuxPolicy) { RTCRtcpMuxPolicyNegotiate, RTCRtcpMuxPolicyRequire };
39
40/** Represents the tcp candidate policy. */
41typedef NS_ENUM(NSInteger, RTCTcpCandidatePolicy) {
42 RTCTcpCandidatePolicyEnabled,
43 RTCTcpCandidatePolicyDisabled
44};
45
46/** Represents the candidate network policy. */
47typedef NS_ENUM(NSInteger, RTCCandidateNetworkPolicy) {
48 RTCCandidateNetworkPolicyAll,
49 RTCCandidateNetworkPolicyLowCost
50};
51
52/** Represents the continual gathering policy. */
53typedef NS_ENUM(NSInteger, RTCContinualGatheringPolicy) {
54 RTCContinualGatheringPolicyGatherOnce,
55 RTCContinualGatheringPolicyGatherContinually
56};
57
58/** Represents the encryption key type. */
59typedef NS_ENUM(NSInteger, RTCEncryptionKeyType) {
60 RTCEncryptionKeyTypeRSA,
61 RTCEncryptionKeyTypeECDSA,
62};
63
64/** Represents the chosen SDP semantics for the RTCPeerConnection. */
65typedef NS_ENUM(NSInteger, RTCSdpSemantics) {
66 RTCSdpSemanticsPlanB,
67 RTCSdpSemanticsUnifiedPlan,
68};
69
70NS_ASSUME_NONNULL_BEGIN
71
Mirko Bonadeie8d57242018-09-17 10:22:56 +020072RTC_OBJC_EXPORT
Anders Carlsson7bca8ca2018-08-30 09:30:29 +020073@interface RTCConfiguration : NSObject
74
75/** An array of Ice Servers available to be used by ICE. */
76@property(nonatomic, copy) NSArray<RTCIceServer *> *iceServers;
77
78/** An RTCCertificate for 're' use. */
79@property(nonatomic, nullable) RTCCertificate *certificate;
80
81/** Which candidates the ICE agent is allowed to use. The W3C calls it
82 * |iceTransportPolicy|, while in C++ it is called |type|. */
83@property(nonatomic, assign) RTCIceTransportPolicy iceTransportPolicy;
84
85/** The media-bundling policy to use when gathering ICE candidates. */
86@property(nonatomic, assign) RTCBundlePolicy bundlePolicy;
87
88/** The rtcp-mux policy to use when gathering ICE candidates. */
89@property(nonatomic, assign) RTCRtcpMuxPolicy rtcpMuxPolicy;
90@property(nonatomic, assign) RTCTcpCandidatePolicy tcpCandidatePolicy;
91@property(nonatomic, assign) RTCCandidateNetworkPolicy candidateNetworkPolicy;
92@property(nonatomic, assign) RTCContinualGatheringPolicy continualGatheringPolicy;
93
94/** By default, the PeerConnection will use a limited number of IPv6 network
95 * interfaces, in order to avoid too many ICE candidate pairs being created
96 * and delaying ICE completion.
97 *
98 * Can be set to INT_MAX to effectively disable the limit.
99 */
100@property(nonatomic, assign) int maxIPv6Networks;
101
102/** Exclude link-local network interfaces
103 * from considertaion for gathering ICE candidates.
104 * Defaults to NO.
105 */
106@property(nonatomic, assign) BOOL disableLinkLocalNetworks;
107
108@property(nonatomic, assign) int audioJitterBufferMaxPackets;
109@property(nonatomic, assign) BOOL audioJitterBufferFastAccelerate;
110@property(nonatomic, assign) int iceConnectionReceivingTimeout;
111@property(nonatomic, assign) int iceBackupCandidatePairPingInterval;
112
113/** Key type used to generate SSL identity. Default is ECDSA. */
114@property(nonatomic, assign) RTCEncryptionKeyType keyType;
115
116/** ICE candidate pool size as defined in JSEP. Default is 0. */
117@property(nonatomic, assign) int iceCandidatePoolSize;
118
119/** Prune turn ports on the same network to the same turn server.
120 * Default is NO.
121 */
122@property(nonatomic, assign) BOOL shouldPruneTurnPorts;
123
124/** If set to YES, this means the ICE transport should presume TURN-to-TURN
125 * candidate pairs will succeed, even before a binding response is received.
126 */
127@property(nonatomic, assign) BOOL shouldPresumeWritableWhenFullyRelayed;
128
129/** If set to non-nil, controls the minimal interval between consecutive ICE
130 * check packets.
131 */
132@property(nonatomic, copy, nullable) NSNumber *iceCheckMinInterval;
133
134/** ICE Periodic Regathering
135 * If set, WebRTC will periodically create and propose candidates without
136 * starting a new ICE generation. The regathering happens continuously with
137 * interval specified in milliseconds by the uniform distribution [a, b].
138 */
139@property(nonatomic, strong, nullable) RTCIntervalRange *iceRegatherIntervalRange;
140
141/** Configure the SDP semantics used by this PeerConnection. Note that the
142 * WebRTC 1.0 specification requires UnifiedPlan semantics. The
143 * RTCRtpTransceiver API is only available with UnifiedPlan semantics.
144 *
145 * PlanB will cause RTCPeerConnection to create offers and answers with at
146 * most one audio and one video m= section with multiple RTCRtpSenders and
147 * RTCRtpReceivers specified as multiple a=ssrc lines within the section. This
148 * will also cause RTCPeerConnection to ignore all but the first m= section of
149 * the same media type.
150 *
151 * UnifiedPlan will cause RTCPeerConnection to create offers and answers with
152 * multiple m= sections where each m= section maps to one RTCRtpSender and one
153 * RTCRtpReceiver (an RTCRtpTransceiver), either both audio or both video. This
154 * will also cause RTCPeerConnection to ignore all but the first a=ssrc lines
155 * that form a Plan B stream.
156 *
157 * For users who wish to send multiple audio/video streams and need to stay
158 * interoperable with legacy WebRTC implementations or use legacy APIs,
159 * specify PlanB.
160 *
161 * For all other users, specify UnifiedPlan.
162 */
163@property(nonatomic, assign) RTCSdpSemantics sdpSemantics;
164
165/** Actively reset the SRTP parameters when the DTLS transports underneath are
166 * changed after offer/answer negotiation. This is only intended to be a
167 * workaround for crbug.com/835958
168 */
169@property(nonatomic, assign) BOOL activeResetSrtpParams;
170
Piotr (Peter) Slatalae0c2e972018-10-08 09:43:21 -0700171/**
172 * If MediaTransportFactory is provided in PeerConnectionFactory, this flag informs PeerConnection
173 * that it should use the MediaTransportInterface.
174 */
175@property(nonatomic, assign) BOOL useMediaTransport;
176
Anders Carlsson7bca8ca2018-08-30 09:30:29 +0200177- (instancetype)init;
178
179@end
180
181NS_ASSUME_NONNULL_END