| /* |
| * Copyright 2015 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. |
| */ |
| |
| #import <Foundation/Foundation.h> |
| |
| #import "RTCCertificate.h" |
| #import "RTCCryptoOptions.h" |
| #import "RTCMacros.h" |
| |
| @class RTC_OBJC_TYPE(RTCIceServer); |
| |
| /** |
| * Represents the ice transport policy. This exposes the same states in C++, |
| * which include one more state than what exists in the W3C spec. |
| */ |
| typedef NS_ENUM(NSInteger, RTCIceTransportPolicy) { |
| RTCIceTransportPolicyNone, |
| RTCIceTransportPolicyRelay, |
| RTCIceTransportPolicyNoHost, |
| RTCIceTransportPolicyAll |
| }; |
| |
| /** Represents the bundle policy. */ |
| typedef NS_ENUM(NSInteger, RTCBundlePolicy) { |
| RTCBundlePolicyBalanced, |
| RTCBundlePolicyMaxCompat, |
| RTCBundlePolicyMaxBundle |
| }; |
| |
| /** Represents the rtcp mux policy. */ |
| typedef NS_ENUM(NSInteger, RTCRtcpMuxPolicy) { RTCRtcpMuxPolicyNegotiate, RTCRtcpMuxPolicyRequire }; |
| |
| /** Represents the tcp candidate policy. */ |
| typedef NS_ENUM(NSInteger, RTCTcpCandidatePolicy) { |
| RTCTcpCandidatePolicyEnabled, |
| RTCTcpCandidatePolicyDisabled |
| }; |
| |
| /** Represents the candidate network policy. */ |
| typedef NS_ENUM(NSInteger, RTCCandidateNetworkPolicy) { |
| RTCCandidateNetworkPolicyAll, |
| RTCCandidateNetworkPolicyLowCost |
| }; |
| |
| /** Represents the continual gathering policy. */ |
| typedef NS_ENUM(NSInteger, RTCContinualGatheringPolicy) { |
| RTCContinualGatheringPolicyGatherOnce, |
| RTCContinualGatheringPolicyGatherContinually |
| }; |
| |
| /** Represents the encryption key type. */ |
| typedef NS_ENUM(NSInteger, RTCEncryptionKeyType) { |
| RTCEncryptionKeyTypeRSA, |
| RTCEncryptionKeyTypeECDSA, |
| }; |
| |
| /** Represents the chosen SDP semantics for the RTCPeerConnection. */ |
| typedef NS_ENUM(NSInteger, RTCSdpSemantics) { |
| RTCSdpSemanticsPlanB, |
| RTCSdpSemanticsUnifiedPlan, |
| }; |
| |
| NS_ASSUME_NONNULL_BEGIN |
| |
| RTC_OBJC_EXPORT |
| @interface RTC_OBJC_TYPE (RTCConfiguration) : NSObject |
| |
| /** If true, allows DSCP codes to be set on outgoing packets, configured using |
| * networkPriority field of RTCRtpEncodingParameters. Defaults to false. |
| */ |
| @property(nonatomic, assign) BOOL enableDscp; |
| |
| /** An array of Ice Servers available to be used by ICE. */ |
| @property(nonatomic, copy) NSArray<RTC_OBJC_TYPE(RTCIceServer) *> *iceServers; |
| |
| /** An RTCCertificate for 're' use. */ |
| @property(nonatomic, nullable) RTC_OBJC_TYPE(RTCCertificate) * certificate; |
| |
| /** Which candidates the ICE agent is allowed to use. The W3C calls it |
| * |iceTransportPolicy|, while in C++ it is called |type|. */ |
| @property(nonatomic, assign) RTCIceTransportPolicy iceTransportPolicy; |
| |
| /** The media-bundling policy to use when gathering ICE candidates. */ |
| @property(nonatomic, assign) RTCBundlePolicy bundlePolicy; |
| |
| /** The rtcp-mux policy to use when gathering ICE candidates. */ |
| @property(nonatomic, assign) RTCRtcpMuxPolicy rtcpMuxPolicy; |
| @property(nonatomic, assign) RTCTcpCandidatePolicy tcpCandidatePolicy; |
| @property(nonatomic, assign) RTCCandidateNetworkPolicy candidateNetworkPolicy; |
| @property(nonatomic, assign) RTCContinualGatheringPolicy continualGatheringPolicy; |
| |
| /** If set to YES, don't gather IPv6 ICE candidates. |
| * Default is NO. |
| */ |
| @property(nonatomic, assign) BOOL disableIPV6; |
| |
| /** If set to YES, don't gather IPv6 ICE candidates on Wi-Fi. |
| * Only intended to be used on specific devices. Certain phones disable IPv6 |
| * when the screen is turned off and it would be better to just disable the |
| * IPv6 ICE candidates on Wi-Fi in those cases. |
| * Default is NO. |
| */ |
| @property(nonatomic, assign) BOOL disableIPV6OnWiFi; |
| |
| /** By default, the PeerConnection will use a limited number of IPv6 network |
| * interfaces, in order to avoid too many ICE candidate pairs being created |
| * and delaying ICE completion. |
| * |
| * Can be set to INT_MAX to effectively disable the limit. |
| */ |
| @property(nonatomic, assign) int maxIPv6Networks; |
| |
| /** Exclude link-local network interfaces |
| * from considertaion for gathering ICE candidates. |
| * Defaults to NO. |
| */ |
| @property(nonatomic, assign) BOOL disableLinkLocalNetworks; |
| |
| @property(nonatomic, assign) int audioJitterBufferMaxPackets; |
| @property(nonatomic, assign) BOOL audioJitterBufferFastAccelerate; |
| @property(nonatomic, assign) int iceConnectionReceivingTimeout; |
| @property(nonatomic, assign) int iceBackupCandidatePairPingInterval; |
| |
| /** Key type used to generate SSL identity. Default is ECDSA. */ |
| @property(nonatomic, assign) RTCEncryptionKeyType keyType; |
| |
| /** ICE candidate pool size as defined in JSEP. Default is 0. */ |
| @property(nonatomic, assign) int iceCandidatePoolSize; |
| |
| /** Prune turn ports on the same network to the same turn server. |
| * Default is NO. |
| */ |
| @property(nonatomic, assign) BOOL shouldPruneTurnPorts; |
| |
| /** If set to YES, this means the ICE transport should presume TURN-to-TURN |
| * candidate pairs will succeed, even before a binding response is received. |
| */ |
| @property(nonatomic, assign) BOOL shouldPresumeWritableWhenFullyRelayed; |
| |
| /* This flag is only effective when |continualGatheringPolicy| is |
| * RTCContinualGatheringPolicyGatherContinually. |
| * |
| * If YES, after the ICE transport type is changed such that new types of |
| * ICE candidates are allowed by the new transport type, e.g. from |
| * RTCIceTransportPolicyRelay to RTCIceTransportPolicyAll, candidates that |
| * have been gathered by the ICE transport but not matching the previous |
| * transport type and as a result not observed by PeerConnectionDelegateAdapter, |
| * will be surfaced to the delegate. |
| */ |
| @property(nonatomic, assign) BOOL shouldSurfaceIceCandidatesOnIceTransportTypeChanged; |
| |
| /** If set to non-nil, controls the minimal interval between consecutive ICE |
| * check packets. |
| */ |
| @property(nonatomic, copy, nullable) NSNumber *iceCheckMinInterval; |
| |
| /** Configure the SDP semantics used by this PeerConnection. Note that the |
| * WebRTC 1.0 specification requires UnifiedPlan semantics. The |
| * RTCRtpTransceiver API is only available with UnifiedPlan semantics. |
| * |
| * PlanB will cause RTCPeerConnection to create offers and answers with at |
| * most one audio and one video m= section with multiple RTCRtpSenders and |
| * RTCRtpReceivers specified as multiple a=ssrc lines within the section. This |
| * will also cause RTCPeerConnection to ignore all but the first m= section of |
| * the same media type. |
| * |
| * UnifiedPlan will cause RTCPeerConnection to create offers and answers with |
| * multiple m= sections where each m= section maps to one RTCRtpSender and one |
| * RTCRtpReceiver (an RTCRtpTransceiver), either both audio or both |
| * video. This will also cause RTCPeerConnection) to ignore all but the first a=ssrc |
| * lines that form a Plan B stream. |
| * |
| * For users who wish to send multiple audio/video streams and need to stay |
| * interoperable with legacy WebRTC implementations or use legacy APIs, |
| * specify PlanB. |
| * |
| * For all other users, specify UnifiedPlan. |
| */ |
| @property(nonatomic, assign) RTCSdpSemantics sdpSemantics; |
| |
| /** Actively reset the SRTP parameters when the DTLS transports underneath are |
| * changed after offer/answer negotiation. This is only intended to be a |
| * workaround for crbug.com/835958 |
| */ |
| @property(nonatomic, assign) BOOL activeResetSrtpParams; |
| |
| /** If the remote side support mid-stream codec switches then allow encoder |
| * switching to be performed. |
| */ |
| |
| @property(nonatomic, assign) BOOL allowCodecSwitching; |
| |
| /** |
| * Defines advanced optional cryptographic settings related to SRTP and |
| * frame encryption for native WebRTC. Setting this will overwrite any |
| * options set through the PeerConnectionFactory (which is deprecated). |
| */ |
| @property(nonatomic, nullable) RTC_OBJC_TYPE(RTCCryptoOptions) * cryptoOptions; |
| |
| /** |
| * An optional string that will be attached to the TURN_ALLOCATE_REQUEST which |
| * which can be used to correlate client logs with backend logs. |
| */ |
| @property(nonatomic, nullable, copy) NSString *turnLoggingId; |
| |
| /** |
| * Time interval between audio RTCP reports. |
| */ |
| @property(nonatomic, assign) int rtcpAudioReportIntervalMs; |
| |
| /** |
| * Time interval between video RTCP reports. |
| */ |
| @property(nonatomic, assign) int rtcpVideoReportIntervalMs; |
| |
| - (instancetype)init; |
| |
| @end |
| |
| NS_ASSUME_NONNULL_END |