зеркало из https://github.com/nextcloud/talk-ios.git
625 строки
21 KiB
Objective-C
625 строки
21 KiB
Objective-C
/**
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* @copyright Copyright (c) 2020 Ivan Sein <ivan@nextcloud.com>
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*
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* @author Ivan Sein <ivan@nextcloud.com>
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*
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* @license GNU GPL version 3 or any later version
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*
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* This program is free software: you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation, either version 3 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program. If not, see <http://www.gnu.org/licenses/>.
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*
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*/
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#import "NCPeerConnection.h"
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#import <WebRTC/RTCConfiguration.h>
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#import <WebRTC/RTCDataChannelConfiguration.h>
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#import <WebRTC/RTCIceServer.h>
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#import <WebRTC/RTCPeerConnectionFactory.h>
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#import <WebRTC/RTCMediaConstraints.h>
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#import <WebRTC/RTCMediaStream.h>
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#import <WebRTC/RTCDefaultVideoEncoderFactory.h>
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#import <WebRTC/RTCDefaultVideoDecoderFactory.h>
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#import "ARDSDPUtils.h"
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#import "NCSignalingMessage.h"
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#import "NextcloudTalk-Swift.h"
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@interface NCPeerConnection () <RTCPeerConnectionDelegate, RTCDataChannelDelegate>
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@property (nonatomic, strong) NSMutableArray *queuedRemoteCandidates;
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@end
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@implementation NCPeerConnection
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- (instancetype)initWithSessionId:(NSString *)sessionId andICEServers:(NSArray *)iceServers forAudioOnlyCall:(BOOL)audioOnly
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{
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self = [super init];
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if (self) {
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[[WebRTCCommon shared] assertQueue];
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RTCMediaConstraints* constraints = [[RTCMediaConstraints alloc]
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initWithMandatoryConstraints:nil
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optionalConstraints:nil];
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RTCConfiguration *config = [[RTCConfiguration alloc] init];
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[config setIceServers:iceServers];
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[config setSdpSemantics:RTCSdpSemanticsUnifiedPlan];
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RTCPeerConnectionFactory *peerConnectionFactory = [WebRTCCommon shared].peerConnectionFactory;
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RTCPeerConnection *peerConnection = [peerConnectionFactory peerConnectionWithConfiguration:config
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constraints:constraints
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delegate:self];
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_peerConnection = peerConnection;
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_peerId = sessionId;
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_isAudioOnly = audioOnly;
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}
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return self;
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}
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- (instancetype)initForPublisherWithSessionId:(NSString *)sessionId andICEServers:(NSArray *)iceServers forAudioOnlyCall:(BOOL)audioOnly
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{
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self = [self initWithSessionId:sessionId andICEServers:iceServers forAudioOnlyCall:audioOnly];
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if (self) {
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_isMCUPublisherPeer = YES;
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}
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return self;
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}
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#pragma mark - NSObject
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- (BOOL)isEqual:(id)object
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{
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if ([object isKindOfClass:[NCPeerConnection class]]) {
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NCPeerConnection *otherConnection = (NCPeerConnection *)object;
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return [otherConnection.peerConnection isEqual:self.peerConnection];
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}
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return NO;
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}
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- (void)dealloc {
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// [self close];
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NSLog(@"NCPeerConnection dealloc");
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}
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#pragma mark - Public
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- (void)addICECandidate:(RTCIceCandidate *)candidate
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{
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[[WebRTCCommon shared] assertQueue];
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if (!_peerConnection.remoteDescription) {
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if (!self.queuedRemoteCandidates) {
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self.queuedRemoteCandidates = [NSMutableArray array];
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}
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NSLog(@"Queued a remote ICE candidate for later.");
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[self.queuedRemoteCandidates addObject:candidate];
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} else {
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NSLog(@"Adding a remote ICE candidate.");
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[self.peerConnection addIceCandidate:candidate completionHandler:^(NSError * _Nullable error) {
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if (error) {
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NSLog(@"Error while adding a remote ICE candidate.");
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}
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}];
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}
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}
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- (void)drainRemoteCandidates
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{
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[[WebRTCCommon shared] assertQueue];
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NSLog(@"Drain %lu remote ICE candidates.", (unsigned long)[self.queuedRemoteCandidates count]);
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for (RTCIceCandidate *candidate in self.queuedRemoteCandidates) {
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[self.peerConnection addIceCandidate:candidate completionHandler:^(NSError * _Nullable error) {
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if (error) {
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NSLog(@"Error while adding a remote ICE candidate.");
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}
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}];
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}
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self.queuedRemoteCandidates = nil;
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}
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- (void)setRemoteDescription:(RTCSessionDescription *)sessionDescription
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{
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[[WebRTCCommon shared] assertQueue];
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__weak NCPeerConnection *weakSelf = self;
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RTCSessionDescription *sdpPreferringCodec = [ARDSDPUtils descriptionForDescription:sessionDescription preferredVideoCodec:@"H264"];
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[_peerConnection setRemoteDescription:sdpPreferringCodec completionHandler:^(NSError *error) {
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[[WebRTCCommon shared] dispatch:^{
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NCPeerConnection *strongSelf = weakSelf;
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if (strongSelf) {
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[strongSelf peerConnectionDidSetRemoteSessionDescription:sdpPreferringCodec error:error];
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}
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}];
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}];
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}
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- (void)sendOffer
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{
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[self sendOfferWithConstraints:[self defaultOfferConstraints]];
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}
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- (void)sendPublisherOffer
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{
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[self sendOfferWithConstraints:[self publisherOfferConstraints]];
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}
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- (void)sendOfferWithConstraints:(RTCMediaConstraints *)constraints
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{
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[[WebRTCCommon shared] assertQueue];
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//Create data channel before creating the offer to enable data channels
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RTCDataChannelConfiguration* config = [[RTCDataChannelConfiguration alloc] init];
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config.isNegotiated = NO;
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_localDataChannel = [_peerConnection dataChannelForLabel:@"status" configuration:config];
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_localDataChannel.delegate = self;
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// Create offer
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__weak NCPeerConnection *weakSelf = self;
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[_peerConnection offerForConstraints:constraints completionHandler:^(RTCSessionDescription *sdp, NSError *error) {
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[[WebRTCCommon shared] dispatch:^{
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NCPeerConnection *strongSelf = weakSelf;
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if (strongSelf) {
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[strongSelf peerConnectionDidCreateLocalSessionDescription:sdp error:error];
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}
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}];
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}];
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}
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- (void)setStatusForDataChannelMessageType:(NSString *)type withPayload:(id)payload
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{
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[[WebRTCCommon shared] assertQueue];
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if ([type isEqualToString:@"nickChanged"]) {
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NSString *nick = @"";
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if ([payload isKindOfClass:[NSString class]]) {
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nick = payload;
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} else if ([payload isKindOfClass:[NSDictionary class]]) {
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nick = [payload objectForKey:@"name"];
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}
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_peerName = nick;
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[self.delegate peerConnection:self didReceivePeerNick:nick];
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} else {
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// Check remote audio/video status
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if ([type isEqualToString:@"audioOn"]) {
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_isRemoteAudioDisabled = NO;
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} else if ([type isEqualToString:@"audioOff"]) {
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_isRemoteAudioDisabled = YES;
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} else if ([type isEqualToString:@"videoOn"]) {
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_isRemoteVideoDisabled = NO;
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} else if ([type isEqualToString:@"videoOff"]) {
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_isRemoteVideoDisabled = YES;
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} else if ([type isEqualToString:@"speaking"]) {
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_isPeerSpeaking = YES;
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} else if ([type isEqualToString:@"stoppedSpeaking"]) {
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_isPeerSpeaking = NO;
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} else if ([type isEqualToString:@"raiseHand"]) {
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_isHandRaised = [payload boolValue];
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}
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[self.delegate peerConnection:self didReceiveStatusDataChannelMessage:type];
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}
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}
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- (void)close
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{
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[[WebRTCCommon shared] assertQueue];
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RTCMediaStream *localStream = [self.peerConnection.localStreams firstObject];
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if (localStream) {
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[self.peerConnection removeStream:localStream];
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}
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[self.peerConnection close];
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self.remoteStream = nil;
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self.localDataChannel = nil;
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self.remoteDataChannel = nil;
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self.peerConnection = nil;
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}
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#pragma mark - RTCPeerConnectionDelegate
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// Delegates from RTCPeerConnection are called on the "signaling_thread" of WebRTC
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- (void)peerConnection:(RTCPeerConnection *)peerConnection didChangeSignalingState:(RTCSignalingState)stateChanged
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{
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NSLog(@"Signaling state with '%@' changed to: %@", self.peerId, [self stringForSignalingState:stateChanged]);
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}
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- (void)peerConnection:(RTCPeerConnection *)peerConnection didAddStream:(RTCMediaStream *)stream
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{
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[[WebRTCCommon shared] dispatch:^{
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NSLog(@"Received %lu video tracks and %lu audio tracks from %@",
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(unsigned long)stream.videoTracks.count,
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(unsigned long)stream.audioTracks.count,
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self.peerId);
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self.remoteStream = stream;
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if ([stream.videoTracks count] == 0) {
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self.isRemoteVideoDisabled = YES;
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}
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[self.delegate peerConnection:self didAddStream:stream];
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}];
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}
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- (void)peerConnection:(RTCPeerConnection *)peerConnection didRemoveStream:(RTCMediaStream *)stream
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{
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[[WebRTCCommon shared] dispatch:^{
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NSLog(@"Stream was removed from %@", self.peerId);
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self.remoteStream = nil;
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[self.delegate peerConnection:self didRemoveStream:stream];
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}];
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}
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- (void)peerConnection:(RTCPeerConnection *)peerConnection didAddReceiver:(RTCRtpReceiver *)rtpReceiver streams:(NSArray<RTCMediaStream *> *)mediaStreams
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{
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[[WebRTCCommon shared] dispatch:^{
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RTCMediaStream *stream = mediaStreams[0];
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if (!stream) {
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return;
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}
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NSLog(@"Received %lu video tracks and %lu audio tracks from %@",
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(unsigned long)stream.videoTracks.count,
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(unsigned long)stream.audioTracks.count,
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self.peerId);
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self.remoteStream = stream;
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[self.delegate peerConnection:self didAddStream:stream];
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}];
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}
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- (void)peerConnection:(RTCPeerConnection *)peerConnection didRemoveReceiver:(RTCRtpReceiver *)rtpReceiver
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{
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[[WebRTCCommon shared] dispatch:^{
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NSLog(@"Receiver was removed from %@", self.peerId);
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self.remoteStream = nil;
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[self.delegate peerConnection:self didRemoveStream:nil];
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}];
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}
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- (void)peerConnectionShouldNegotiate:(RTCPeerConnection *)peerConnection
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{
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NSLog(@"WARNING: Renegotiation needed but unimplemented.");
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}
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- (void)peerConnection:(RTCPeerConnection *)peerConnection didChangeIceConnectionState:(RTCIceConnectionState)newState
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{
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[[WebRTCCommon shared] dispatch:^{
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NSLog(@"ICE state with '%@' changed to: %@", self.peerId, [self stringForConnectionState:newState]);
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[self.delegate peerConnection:self didChangeIceConnectionState:newState];
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}];
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}
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- (void)peerConnection:(RTCPeerConnection *)peerConnection didChangeIceGatheringState:(RTCIceGatheringState)newState
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{
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NSLog(@"ICE gathering state with '%@' changed to : %@", self.peerId, [self stringForGatheringState:newState]);
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}
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- (void)peerConnection:(RTCPeerConnection *)peerConnection didGenerateIceCandidate:(RTCIceCandidate *)candidate
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{
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[[WebRTCCommon shared] dispatch:^{
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NSLog(@"Peer '%@' did generate Ice Candidate: %@", self.peerId, candidate);
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[self.delegate peerConnection:self didGenerateIceCandidate:candidate];
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}];
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}
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- (void)peerConnection:(RTCPeerConnection *)peerConnection didRemoveIceCandidates:(NSArray<RTCIceCandidate *> *)candidates
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{
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NSLog(@"PeerConnection didRemoveIceCandidates delegate has been called.");
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}
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- (void)peerConnection:(RTCPeerConnection *)peerConnection didOpenDataChannel:(RTCDataChannel *)dataChannel
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{
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[[WebRTCCommon shared] dispatch:^{
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if ([dataChannel.label isEqualToString:@"status"] || [dataChannel.label isEqualToString:@"JanusDataChannel"]) {
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self->_remoteDataChannel = dataChannel;
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self->_remoteDataChannel.delegate = self;
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NSLog(@"Remote data channel '%@' was opened.", dataChannel.label);
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} else {
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NSLog(@"Data channel '%@' was opened.", dataChannel.label);
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}
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}];
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}
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#pragma mark - RTCDataChannelDelegate
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// Delegates from RTCDataChannel are called on the "signaling_thread" of WebRTC
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- (void)dataChannelDidChangeState:(RTCDataChannel *)dataChannel
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{
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[[WebRTCCommon shared] dispatch:^{
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NSLog(@"Data cahnnel '%@' did change state: %ld", dataChannel.label, (long)dataChannel.readyState);
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if (dataChannel.readyState == RTCDataChannelStateOpen && [dataChannel.label isEqualToString:@"status"]) {
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[self.delegate peerConnectionDidOpenStatusDataChannel:self];
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}
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}];
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}
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- (void)dataChannel:(RTCDataChannel *)dataChannel didReceiveMessageWithBuffer:(RTCDataBuffer *)buffer
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{
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[[WebRTCCommon shared] dispatch:^{
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NSDictionary *message = [self getDataChannelMessageFromJSONData:buffer.data];
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NSString *messageType = [message objectForKey:@"type"];
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id messagePayload = [message objectForKey:@"payload"];
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[self setStatusForDataChannelMessageType:messageType withPayload:messagePayload];
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}];
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}
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- (NSDictionary *)getDataChannelMessageFromJSONData:(NSData *)jsonData
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{
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NSError *error;
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NSDictionary* messageDict = [NSJSONSerialization JSONObjectWithData:jsonData
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options:kNilOptions
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error:&error];
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if (!messageDict) {
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NSLog(@"Error parsing data channel message: %@", error);
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}
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return messageDict;
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}
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- (NSData *)createDataChannelMessage:(NSDictionary *)message
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{
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NSError *error;
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NSData *jsonData = [NSJSONSerialization dataWithJSONObject:message
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options:0
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error:&error];
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if (!jsonData) {
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NSLog(@"Error creating data channel message: %@", error);
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}
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return jsonData;
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}
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- (void)sendDataChannelMessageOfType:(NSString *)type withPayload:(id)payload
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{
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[[WebRTCCommon shared] assertQueue];
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NSDictionary *message = @{@"type": type};
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if (payload) {
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message = @{@"type": type,
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@"payload": payload};
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}
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NSData *jsonMessage = [self createDataChannelMessage:message];
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RTCDataBuffer *dataBuffer = [[RTCDataBuffer alloc] initWithData:jsonMessage isBinary:NO];
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if (_localDataChannel) {
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[_localDataChannel sendData:dataBuffer];
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} else if (_remoteDataChannel) {
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[_remoteDataChannel sendData:dataBuffer];
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} else {
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NSLog(@"No data channel opened");
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}
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}
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#pragma mark - RTCSessionDescriptionDelegate
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// Delegates from RTCSessionDescription are already dispatched to the webrtc client thread
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- (void)peerConnectionDidCreateLocalSessionDescription:(RTCSessionDescription *)sdp error:(NSError *)error
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{
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if (error) {
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NSLog(@"Failed to create local session description for peer %@. Error: %@", _peerId, error);
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return;
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}
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[[WebRTCCommon shared] assertQueue];
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// Set H264 as preferred codec.
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RTCSessionDescription *sdpPreferringCodec = [ARDSDPUtils descriptionForDescription:sdp preferredVideoCodec:@"H264"];
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__weak NCPeerConnection *weakSelf = self;
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[self->_peerConnection setLocalDescription:sdpPreferringCodec completionHandler:^(NSError *error) {
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if (error) {
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NSLog(@"Failed to set local session description: %@", error);
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return;
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}
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[[WebRTCCommon shared] dispatch:^{
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NCPeerConnection *strongSelf = weakSelf;
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if (strongSelf) {
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[strongSelf.delegate peerConnection:strongSelf needsToSendSessionDescription:sdpPreferringCodec];
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}
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}];
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}];
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}
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- (void)peerConnectionDidSetRemoteSessionDescription:(RTCSessionDescription *)sessionDescription error:(NSError *)error
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{
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if (error) {
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NSLog(@"Failed to set remote session description for peer %@. Error: %@", _peerId, error);
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return;
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}
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[[WebRTCCommon shared] assertQueue];
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// If we just set a remote offer we need to create an answer and set it as local description.
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if (self->_peerConnection.signalingState == RTCSignalingStateHaveRemoteOffer) {
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// Create data channel before sending answer
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RTCDataChannelConfiguration* config = [[RTCDataChannelConfiguration alloc] init];
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config.isNegotiated = NO;
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self->_localDataChannel = [self->_peerConnection dataChannelForLabel:@"status" configuration:config];
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self->_localDataChannel.delegate = self;
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// Stop video transceiver in audio only peer connections
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// Constraints are no longer supported when creating answers (with Unified Plan semantics)
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if (_isAudioOnly) {
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for (RTCRtpTransceiver *transceiver in self->_peerConnection.transceivers) {
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if (transceiver.mediaType == RTCRtpMediaTypeVideo) {
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[transceiver stopInternal];
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NSLog(@"Stop video transceiver in audio only peer connections.");
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}
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}
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}
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// Create answer
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RTCMediaConstraints *constraints = [self defaultAnswerConstraints];
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__weak NCPeerConnection *weakSelf = self;
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[self->_peerConnection answerForConstraints:constraints completionHandler:^(RTCSessionDescription *sdp, NSError *error) {
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[[WebRTCCommon shared] dispatch:^{
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NCPeerConnection *strongSelf = weakSelf;
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if (strongSelf) {
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[strongSelf peerConnectionDidCreateLocalSessionDescription:sdp error:error];
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}
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}];
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}];
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}
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if (self->_peerConnection.remoteDescription) {
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[self drainRemoteCandidates];
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}
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}
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#pragma mark - Utils
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- (RTCMediaConstraints *)defaultAnswerConstraints
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{
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return [self defaultOfferConstraints];
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}
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- (RTCMediaConstraints *)defaultOfferConstraints
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{
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NSDictionary *mandatoryConstraints = @{
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@"OfferToReceiveAudio" : @"true",
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@"OfferToReceiveVideo" : @"true"
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};
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if (_isAudioOnly) {
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mandatoryConstraints = @{
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@"OfferToReceiveAudio" : @"true",
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@"OfferToReceiveVideo" : @"false"
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};
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}
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NSDictionary *optionalConstraints = @{
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@"internalSctpDataChannels": @"true",
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@"DtlsSrtpKeyAgreement": @"true"
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};
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RTCMediaConstraints* constraints = [[RTCMediaConstraints alloc]
|
|
initWithMandatoryConstraints:mandatoryConstraints
|
|
optionalConstraints:optionalConstraints];
|
|
return constraints;
|
|
}
|
|
|
|
- (RTCMediaConstraints *)publisherOfferConstraints
|
|
{
|
|
NSDictionary *mandatoryConstraints = @{
|
|
@"OfferToReceiveAudio" : @"false",
|
|
@"OfferToReceiveVideo" : @"false"
|
|
};
|
|
|
|
NSDictionary *optionalConstraints = @{
|
|
@"internalSctpDataChannels": @"true",
|
|
@"DtlsSrtpKeyAgreement": @"true"
|
|
};
|
|
|
|
RTCMediaConstraints* constraints = [[RTCMediaConstraints alloc]
|
|
initWithMandatoryConstraints:mandatoryConstraints
|
|
optionalConstraints:optionalConstraints];
|
|
return constraints;
|
|
}
|
|
|
|
- (NSString *)stringForSignalingState:(RTCSignalingState)state
|
|
{
|
|
switch (state) {
|
|
case RTCSignalingStateStable:
|
|
return @"Stable";
|
|
break;
|
|
case RTCSignalingStateHaveLocalOffer:
|
|
return @"Have Local Offer";
|
|
break;
|
|
case RTCSignalingStateHaveRemoteOffer:
|
|
return @"Have Remote Offer";
|
|
break;
|
|
case RTCSignalingStateClosed:
|
|
return @"Closed";
|
|
break;
|
|
default:
|
|
return @"Other state";
|
|
break;
|
|
}
|
|
}
|
|
|
|
- (NSString *)stringForConnectionState:(RTCIceConnectionState)state
|
|
{
|
|
switch (state) {
|
|
case RTCIceConnectionStateNew:
|
|
return @"New";
|
|
break;
|
|
case RTCIceConnectionStateChecking:
|
|
return @"Checking";
|
|
break;
|
|
case RTCIceConnectionStateConnected:
|
|
return @"Connected";
|
|
break;
|
|
case RTCIceConnectionStateCompleted:
|
|
return @"Completed";
|
|
break;
|
|
case RTCIceConnectionStateFailed:
|
|
return @"Failed";
|
|
break;
|
|
case RTCIceConnectionStateDisconnected:
|
|
return @"Disconnected";
|
|
break;
|
|
case RTCIceConnectionStateClosed:
|
|
return @"Closed";
|
|
break;
|
|
default:
|
|
return @"Other state";
|
|
break;
|
|
}
|
|
}
|
|
|
|
- (NSString *)stringForGatheringState:(RTCIceGatheringState)state
|
|
{
|
|
switch (state) {
|
|
case RTCIceGatheringStateNew:
|
|
return @"New";
|
|
break;
|
|
case RTCIceGatheringStateGathering:
|
|
return @"Gathering";
|
|
break;
|
|
case RTCIceGatheringStateComplete:
|
|
return @"Complete";
|
|
break;
|
|
default:
|
|
return @"Other state";
|
|
break;
|
|
}
|
|
}
|
|
|
|
@end
|