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This reverts commit23501a2aa6
. Reason for revert: Breaks downstream features Original change's description: > Reland: Remove unsupported configuration value, `allow_codec_switching` > > This reverts commit6b0c5babe0
. > > Reason for revert: Relanding once downstream issues have been addressed > > Original change's description: > > Revert "Remove unsupported configuration value, `allow_codec_switching`" > > > > This reverts commit8f7a17f80f
. > > > > Reason for revert: breaks downstream > > > > Original change's description: > > > Remove unsupported configuration value, `allow_codec_switching` > > > > > > Bug: webrtc:11341 > > > Change-Id: I8ff598848996bd63ccc572e11f8f69c892a4a459 > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324284 > > > Reviewed-by: Philip Eliasson <philipel@webrtc.org> > > > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> > > > Cr-Commit-Position: refs/heads/main@{#40995} > > > > Bug: webrtc:11341 > > Change-Id: I784fd95062fc71f8dcc139b05121985f60709004 > > No-Presubmit: true > > No-Tree-Checks: true > > No-Try: true > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324780 > > Owners-Override: Philip Eliasson <philipel@webrtc.org> > > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > > Commit-Queue: Philip Eliasson <philipel@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#40998} > > Bug: webrtc:11341 > Change-Id: I3cb3e699fd76942c51f0f42a99bcb19ac607632e > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324782 > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#41032} Bug: webrtc:11341 Change-Id: I0eb8e6a464a8a51e6359caf8f43231dc275c4f20 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327382 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41161}
268 lines
9.8 KiB
Objective-C
268 lines
9.8 KiB
Objective-C
/*
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* Copyright 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#import <Foundation/Foundation.h>
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#import "RTCCertificate.h"
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#import "RTCCryptoOptions.h"
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#import "RTCMacros.h"
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@class RTC_OBJC_TYPE(RTCIceServer);
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/**
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* Represents the ice transport policy. This exposes the same states in C++,
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* which include one more state than what exists in the W3C spec.
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*/
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typedef NS_ENUM(NSInteger, RTCIceTransportPolicy) {
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RTCIceTransportPolicyNone,
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RTCIceTransportPolicyRelay,
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RTCIceTransportPolicyNoHost,
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RTCIceTransportPolicyAll
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};
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/** Represents the bundle policy. */
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typedef NS_ENUM(NSInteger, RTCBundlePolicy) {
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RTCBundlePolicyBalanced,
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RTCBundlePolicyMaxCompat,
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RTCBundlePolicyMaxBundle
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};
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/** Represents the rtcp mux policy. */
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typedef NS_ENUM(NSInteger, RTCRtcpMuxPolicy) { RTCRtcpMuxPolicyNegotiate, RTCRtcpMuxPolicyRequire };
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/** Represents the tcp candidate policy. */
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typedef NS_ENUM(NSInteger, RTCTcpCandidatePolicy) {
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RTCTcpCandidatePolicyEnabled,
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RTCTcpCandidatePolicyDisabled
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};
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/** Represents the candidate network policy. */
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typedef NS_ENUM(NSInteger, RTCCandidateNetworkPolicy) {
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RTCCandidateNetworkPolicyAll,
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RTCCandidateNetworkPolicyLowCost
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};
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/** Represents the continual gathering policy. */
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typedef NS_ENUM(NSInteger, RTCContinualGatheringPolicy) {
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RTCContinualGatheringPolicyGatherOnce,
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RTCContinualGatheringPolicyGatherContinually
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};
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/** Represents the encryption key type. */
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typedef NS_ENUM(NSInteger, RTCEncryptionKeyType) {
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RTCEncryptionKeyTypeRSA,
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RTCEncryptionKeyTypeECDSA,
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};
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/** Represents the chosen SDP semantics for the RTCPeerConnection. */
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typedef NS_ENUM(NSInteger, RTCSdpSemantics) {
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// TODO(https://crbug.com/webrtc/13528): Remove support for Plan B.
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RTCSdpSemanticsPlanB,
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RTCSdpSemanticsUnifiedPlan,
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};
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NS_ASSUME_NONNULL_BEGIN
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RTC_OBJC_EXPORT
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@interface RTC_OBJC_TYPE (RTCConfiguration) : NSObject
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/** If true, allows DSCP codes to be set on outgoing packets, configured using
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* networkPriority field of RTCRtpEncodingParameters. Defaults to false.
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*/
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@property(nonatomic, assign) BOOL enableDscp;
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/** An array of Ice Servers available to be used by ICE. */
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@property(nonatomic, copy) NSArray<RTC_OBJC_TYPE(RTCIceServer) *> *iceServers;
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/** An RTCCertificate for 're' use. */
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@property(nonatomic, nullable) RTC_OBJC_TYPE(RTCCertificate) * certificate;
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/** Which candidates the ICE agent is allowed to use. The W3C calls it
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* `iceTransportPolicy`, while in C++ it is called `type`. */
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@property(nonatomic, assign) RTCIceTransportPolicy iceTransportPolicy;
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/** The media-bundling policy to use when gathering ICE candidates. */
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@property(nonatomic, assign) RTCBundlePolicy bundlePolicy;
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/** The rtcp-mux policy to use when gathering ICE candidates. */
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@property(nonatomic, assign) RTCRtcpMuxPolicy rtcpMuxPolicy;
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@property(nonatomic, assign) RTCTcpCandidatePolicy tcpCandidatePolicy;
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@property(nonatomic, assign) RTCCandidateNetworkPolicy candidateNetworkPolicy;
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@property(nonatomic, assign) RTCContinualGatheringPolicy continualGatheringPolicy;
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/** If set to YES, don't gather IPv6 ICE candidates on Wi-Fi.
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* Only intended to be used on specific devices. Certain phones disable IPv6
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* when the screen is turned off and it would be better to just disable the
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* IPv6 ICE candidates on Wi-Fi in those cases.
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* Default is NO.
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*/
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@property(nonatomic, assign) BOOL disableIPV6OnWiFi;
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/** By default, the PeerConnection will use a limited number of IPv6 network
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* interfaces, in order to avoid too many ICE candidate pairs being created
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* and delaying ICE completion.
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*
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* Can be set to INT_MAX to effectively disable the limit.
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*/
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@property(nonatomic, assign) int maxIPv6Networks;
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/** Exclude link-local network interfaces
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* from considertaion for gathering ICE candidates.
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* Defaults to NO.
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*/
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@property(nonatomic, assign) BOOL disableLinkLocalNetworks;
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@property(nonatomic, assign) int audioJitterBufferMaxPackets;
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@property(nonatomic, assign) BOOL audioJitterBufferFastAccelerate;
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@property(nonatomic, assign) int iceConnectionReceivingTimeout;
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@property(nonatomic, assign) int iceBackupCandidatePairPingInterval;
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/** Key type used to generate SSL identity. Default is ECDSA. */
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@property(nonatomic, assign) RTCEncryptionKeyType keyType;
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/** ICE candidate pool size as defined in JSEP. Default is 0. */
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@property(nonatomic, assign) int iceCandidatePoolSize;
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/** Prune turn ports on the same network to the same turn server.
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* Default is NO.
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*/
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@property(nonatomic, assign) BOOL shouldPruneTurnPorts;
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/** If set to YES, this means the ICE transport should presume TURN-to-TURN
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* candidate pairs will succeed, even before a binding response is received.
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*/
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@property(nonatomic, assign) BOOL shouldPresumeWritableWhenFullyRelayed;
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/* This flag is only effective when `continualGatheringPolicy` is
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* RTCContinualGatheringPolicyGatherContinually.
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*
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* If YES, after the ICE transport type is changed such that new types of
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* ICE candidates are allowed by the new transport type, e.g. from
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* RTCIceTransportPolicyRelay to RTCIceTransportPolicyAll, candidates that
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* have been gathered by the ICE transport but not matching the previous
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* transport type and as a result not observed by PeerConnectionDelegateAdapter,
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* will be surfaced to the delegate.
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*/
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@property(nonatomic, assign) BOOL shouldSurfaceIceCandidatesOnIceTransportTypeChanged;
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/** If set to non-nil, controls the minimal interval between consecutive ICE
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* check packets.
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*/
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@property(nonatomic, copy, nullable) NSNumber *iceCheckMinInterval;
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/**
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* Configure the SDP semantics used by this PeerConnection. By default, this
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* is RTCSdpSemanticsUnifiedPlan which is compliant to the WebRTC 1.0
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* specification. It is possible to overrwite this to the deprecated
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* RTCSdpSemanticsPlanB SDP format, but note that RTCSdpSemanticsPlanB will be
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* deleted at some future date, see https://crbug.com/webrtc/13528.
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*
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* RTCSdpSemanticsUnifiedPlan will cause RTCPeerConnection to create offers and
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* answers with multiple m= sections where each m= section maps to one
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* RTCRtpSender and one RTCRtpReceiver (an RTCRtpTransceiver), either both audio
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* or both video. This will also cause RTCPeerConnection to ignore all but the
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* first a=ssrc lines that form a Plan B stream.
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*
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* RTCSdpSemanticsPlanB will cause RTCPeerConnection to create offers and
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* answers with at most one audio and one video m= section with multiple
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* RTCRtpSenders and RTCRtpReceivers specified as multiple a=ssrc lines within
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* the section. This will also cause RTCPeerConnection to ignore all but the
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* first m= section of the same media type.
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*/
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@property(nonatomic, assign) RTCSdpSemantics sdpSemantics;
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/** Actively reset the SRTP parameters when the DTLS transports underneath are
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* changed after offer/answer negotiation. This is only intended to be a
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* workaround for crbug.com/835958
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*/
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@property(nonatomic, assign) BOOL activeResetSrtpParams;
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/** If the remote side support mid-stream codec switches then allow encoder
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* switching to be performed.
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*/
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@property(nonatomic, assign) BOOL allowCodecSwitching;
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/**
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* Defines advanced optional cryptographic settings related to SRTP and
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* frame encryption for native WebRTC. Setting this will overwrite any
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* options set through the PeerConnectionFactory (which is deprecated).
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*/
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@property(nonatomic, nullable) RTC_OBJC_TYPE(RTCCryptoOptions) * cryptoOptions;
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/**
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* An optional string that will be attached to the TURN_ALLOCATE_REQUEST which
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* which can be used to correlate client logs with backend logs.
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*/
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@property(nonatomic, nullable, copy) NSString *turnLoggingId;
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/**
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* Time interval between audio RTCP reports.
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*/
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@property(nonatomic, assign) int rtcpAudioReportIntervalMs;
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/**
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* Time interval between video RTCP reports.
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*/
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@property(nonatomic, assign) int rtcpVideoReportIntervalMs;
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/**
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* Allow implicit rollback of local description when remote description
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* conflicts with local description.
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* See: https://w3c.github.io/webrtc-pc/#dom-peerconnection-setremotedescription
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*/
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@property(nonatomic, assign) BOOL enableImplicitRollback;
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/**
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* Control if "a=extmap-allow-mixed" is included in the offer.
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* See: https://www.chromestatus.com/feature/6269234631933952
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*/
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@property(nonatomic, assign) BOOL offerExtmapAllowMixed;
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/**
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* Defines the interval applied to ALL candidate pairs
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* when ICE is strongly connected, and it overrides the
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* default value of this interval in the ICE implementation;
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*/
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@property(nonatomic, copy, nullable) NSNumber *iceCheckIntervalStrongConnectivity;
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/**
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* Defines the counterpart for ALL pairs when ICE is
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* weakly connected, and it overrides the default value of
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* this interval in the ICE implementation
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*/
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@property(nonatomic, copy, nullable) NSNumber *iceCheckIntervalWeakConnectivity;
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/**
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* The min time period for which a candidate pair must wait for response to
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* connectivity checks before it becomes unwritable. This parameter
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* overrides the default value in the ICE implementation if set.
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*/
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@property(nonatomic, copy, nullable) NSNumber *iceUnwritableTimeout;
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/**
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* The min number of connectivity checks that a candidate pair must sent
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* without receiving response before it becomes unwritable. This parameter
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* overrides the default value in the ICE implementation if set.
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*/
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@property(nonatomic, copy, nullable) NSNumber *iceUnwritableMinChecks;
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/**
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* The min time period for which a candidate pair must wait for response to
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* connectivity checks it becomes inactive. This parameter overrides the
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* default value in the ICE implementation if set.
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*/
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@property(nonatomic, copy, nullable) NSNumber *iceInactiveTimeout;
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- (instancetype)init;
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@end
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NS_ASSUME_NONNULL_END
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