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iOS上使用WebRTC推拉流的案例_webrtc添加google公共stun

网友: 技术文档 2025-07-26 20:54:45

一、库集成

首先，确保在你的 Podfile 中添加依赖：

pod \'GoogleWebRTC\'

然后执行 pod install 安装库。

二、代码示例

2.1、权限配置：在 Info.plist 中添加摄像头、麦克风权限

NSCameraUsageDescription需要访问摄像头进行视频通话NSMicrophoneUsageDescription需要访问麦克风进行语音通话NSAppTransportSecurity NSAllowsArbitraryLoads

2.2、代码
核心类oc：WebRTCManager.m

#import \"WebRTCManager.h\"#import <AVFoundation/AVFoundation.h>@interface WebRTCManager ()@property (nonatomic, strong) RTCPeerConnectionFactory *factory;@property (nonatomic, strong) RTCVideoTrack *localVideoTrack;@property (nonatomic, strong) RTCAudioTrack *localAudioTrack;@property (nonatomic, strong) RTCVideoRendererAdapter *localRenderer;@end@implementation WebRTCManager- (instancetype)initWithDelegate:(id<WebRTCManagerDelegate>)delegate { self = [super init]; if (self) { _delegate = delegate; [self setupPeerConnectionFactory]; [self setupPeerConnection]; } return self;}// 初始化 PeerConnection 工厂- (void)setupPeerConnectionFactory { RTCInitializeSSL(); _factory = [[RTCPeerConnectionFactory alloc] init];}// 配置并创建 PeerConnection- (void)setupPeerConnection { RTCConfiguration *config = [[RTCConfiguration alloc] init]; config.iceServers = @[ [[RTCIceServer alloc] initWithURLStrings:@[@\"stun:stun.l.google.com:19302\"]] ]; RTCOfferAnswerOptions *offerAnswerOptions = [[RTCOfferAnswerOptions alloc] init]; offerAnswerOptions.offerToReceiveAudio = YES; offerAnswerOptions.offerToReceiveVideo = YES; RTCMediaConstraints *constraints = [[RTCMediaConstraints alloc] initWithMandatoryConstraints:@[] optionalConstraints:@[]]; _peerConnection = [_factory peerConnectionWithConfiguration:config constraints:constraints delegate:self];}// 初始化本地媒体流- (void)setupLocalStreamWithVideoView:(UIView *)videoView { // 请求音视频权限 [AVCaptureDevice requestAccessForMediaType:AVMediaTypeVideo completionHandler:^(BOOL granted) { dispatch_async(dispatch_get_main_queue(), ^{ if (granted) { [self createLocalMediaStream]; [self setupLocalVideoRender:videoView]; } else { NSLog(@\"需要摄像头权限才能使用视频功能\"); } }); }];}// 创建本地媒体流- (void)createLocalMediaStream { _localStream = [_factory mediaStreamWithStreamId:@\"localStream\"]; // 创建音频轨道 RTCAudioSource *audioSource = [_factory audioSourceWithConstraints:[self defaultMediaConstraints]]; _localAudioTrack = [_factory audioTrackWithSource:audioSource trackId:@\"audio0\"]; [_localStream addAudioTrack:_localAudioTrack]; // 创建视频轨道 RTCVideoSource *videoSource = [_factory videoSourceWithConstraints:[self videoConstraints]]; _localVideoTrack = [_factory videoTrackWithSource:videoSource trackId:@\"video0\"]; [_localStream addVideoTrack:_localVideoTrack]; // 将本地流添加到 PeerConnection [_peerConnection addStream:_localStream];}// 设置本地视频渲染- (void)setupLocalVideoRender:(UIView *)videoView { RTCMTLVideoView *rendererView = [[RTCMTLVideoView alloc] init]; rendererView.frame = videoView.bounds; rendererView.autoresizingMask = UIViewAutoresizingFlexibleWidth | UIViewAutoresizingFlexibleHeight; [videoView addSubview:rendererView]; _localRenderer = [[RTCVideoRendererAdapter alloc] initWithRenderer:rendererView]; [_localVideoTrack addRenderer:_localRenderer];}// 创建 Offer- (void)createOffer { RTCOfferAnswerConstraints *constraints = [[RTCOfferAnswerConstraints alloc] init]; constraints.mandatoryConstraints = @[ [[RTCPair alloc] initWithKey:@\"OfferToReceiveAudio\" value:@\"true\"], [[RTCPair alloc] initWithKey:@\"OfferToReceiveVideo\" value:@\"true\"] ]; [_peerConnection createOfferWithConstraints:constraints completionHandler:^(RTCSessionDescription * _Nullable sdp, NSError * _Nullable error) { if (error) { NSLog(@\"创建 Offer 失败: %@\", error.localizedDescription); return; } [self.peerConnection setLocalDescriptionWithSessionDescription:sdp completionHandler:^(NSError * _Nullable error) { if (error) { NSLog(@\"设置本地描述失败: %@\", error.localizedDescription); return; } // 这里应该将 sdp 发送给信令服务器 NSString *offerString = [self sessionDescriptionToString:sdp]; NSLog(@\"生成 Offer: %@\", offerString); // [self.signalingClient sendOffer:offerString]; }]; }];}// 处理远程 Offer- (void)handleRemoteOffer:(NSString *)offer { RTCSessionDescription *remoteSDP = [self stringToSessionDescription:offer type:RTCSdpTypeOffer]; [_peerConnection setRemoteDescriptionWithSessionDescription:remoteSDP completionHandler:^(NSError * _Nullable error) { if (error) { NSLog(@\"设置远程 Offer 失败: %@\", error.localizedDescription); return; } // 创建 Answer [self createAnswer]; }];}// 创建 Answer- (void)createAnswer { RTCOfferAnswerConstraints *constraints = [[RTCOfferAnswerConstraints alloc] init]; [_peerConnection createAnswerWithConstraints:constraints completionHandler:^(RTCSessionDescription * _Nullable sdp, NSError * _Nullable error) { if (error) { NSLog(@\"创建 Answer 失败: %@\", error.localizedDescription); return; } [self.peerConnection setLocalDescriptionWithSessionDescription:sdp completionHandler:^(NSError * _Nullable error) { if (error) { NSLog(@\"设置本地 Answer 失败: %@\", error.localizedDescription); return; } // 将 Answer 发送给信令服务器 NSString *answerString = [self sessionDescriptionToString:sdp]; NSLog(@\"生成 Answer: %@\", answerString); // [self.signalingClient sendAnswer:answerString]; }]; }];}// 处理远程 Answer- (void)handleRemoteAnswer:(NSString *)answer { RTCSessionDescription *remoteSDP = [self stringToSessionDescription:answer type:RTCSdpTypeAnswer]; [_peerConnection setRemoteDescriptionWithSessionDescription:remoteSDP completionHandler:^(NSError * _Nullable error) { if (error) { NSLog(@\"设置远程 Answer 失败: %@\", error.localizedDescription); } }];}// 处理远程 ICE 候选者- (void)handleRemoteICECandidate:(NSString *)candidate { NSDictionary *candidateDict = [NSJSONSerialization JSONObjectWithData:[candidate dataUsingEncoding:NSUTF8StringEncoding] options:0 error:nil]; if (!candidateDict) return; RTCIceCandidate *iceCandidate = [[RTCIceCandidate alloc] initWithSdpMLineIndex:[candidateDict[@\"sdpMLineIndex\"] integerValue] sdpMid:candidateDict[@\"sdpMid\"] sdp:candidateDict[@\"sdp\"]]; [_peerConnection addIceCandidate:iceCandidate completionHandler:^(NSError * _Nullable error) { if (error) { NSLog(@\"添加远程 ICE 候选者失败: %@\", error.localizedDescription); } }];}// 断开连接- (void)disconnect { [_peerConnection close]; _peerConnection = nil; [_localStream removeAllAudioTracks]; [_localStream removeAllVideoTracks]; _localStream = nil;}#pragma mark - RTCPeerConnectionDelegate// 收到远程媒体流- (void)peerConnection:(RTCPeerConnection *)peerConnection didAddStream:(RTCMediaStream *)stream { NSLog(@\"收到远程媒体流\"); if ([self.delegate respondsToSelector:@selector(didReceiveRemoteStream:)] && stream) { [self.delegate didReceiveRemoteStream:stream]; }}// 生成 ICE 候选者- (void)peerConnection:(RTCPeerConnection *)peerConnection didGenerateIceCandidate:(RTCIceCandidate *)candidate { NSDictionary *candidateDict = @{ @\"sdpMLineIndex\": @(candidate.sdpMLineIndex), @\"sdpMid\": candidate.sdpMid ?: @\"\", @\"sdp\": candidate.sdp }; NSData *jsonData = [NSJSONSerialization dataWithJSONObject:candidateDict options:0 error:nil]; NSString *candidateString = [[NSString alloc] initWithData:jsonData encoding:NSUTF8StringEncoding]; // 将 ICE 候选者发送给信令服务器 NSLog(@\"生成 ICE 候选者: %@\", candidateString); // [self.signalingClient sendICECandidate:candidateString];}#pragma mark - 工具方法// 媒体约束配置- (RTCMediaConstraints *)defaultMediaConstraints { return [[RTCMediaConstraints alloc] initWithMandatoryConstraints:@[] optionalConstraints:@[]];}// 视频约束配置- (RTCMediaConstraints *)videoConstraints { return [[RTCMediaConstraints alloc] initWithMandatoryConstraints:@[ [[RTCPair alloc] initWithKey:@\"maxWidth\" value:@\"1280\"], [[RTCPair alloc] initWithKey:@\"maxHeight\" value:@\"720\"], [[RTCPair alloc] initWithKey:@\"maxFrameRate\" value:@\"30\"] ] optionalConstraints:@[]];}// 将 SessionDescription 转换为字符串- (NSString *)sessionDescriptionToString:(RTCSessionDescription *)sdp { NSDictionary *dict = @{ @\"type\": [self sdpTypeToString:sdp.type], @\"sdp\": sdp.sdp }; NSData *jsonData = [NSJSONSerialization dataWithJSONObject:dict options:0 error:nil]; return [[NSString alloc] initWithData:jsonData encoding:NSUTF8StringEncoding];}// 将字符串转换为 SessionDescription- (RTCSessionDescription *)stringToSessionDescription:(NSString *)string type:(RTCSdpType)type { NSData *data = [string dataUsingEncoding:NSUTF8StringEncoding]; NSDictionary *dict = [NSJSONSerialization JSONObjectWithData:data options:0 error:nil]; return [[RTCSessionDescription alloc] initWithType:type sdp:dict[@\"sdp\"]];}// SDP 类型转换- (NSString *)sdpTypeToString:(RTCSdpType)type { switch (type) { case RTCSdpTypeOffer: return @\"offer\"; case RTCSdpTypeAnswer: return @\"answer\"; case RTCSdpTypePrAnswer: return @\"pranswer\"; case RTCSdpTypeRollback: return @\"rollback\"; default: return @\"\"; }}@end

调用OC：ViewController.m

#import \"ViewController.h\"#import <WebRTC/WebRTC.h>@interface ViewController ()@property (nonatomic, strong) WebRTCManager *rtcManager;@property (nonatomic, strong) UIView *localVideoView;@property (nonatomic, strong) UIView *remoteVideoView;@property (nonatomic, strong) UIButton *connectButton;@end@implementation ViewController- (void)viewDidLoad { [super viewDidLoad]; self.view.backgroundColor = UIColor.whiteColor; [self setupUI]; [self setupWebRTC];}- (void)setupUI { // 本地视频视图 _localVideoView = [[UIView alloc] init]; _localVideoView.backgroundColor = UIColor.lightGrayColor; _localVideoView.translatesAutoresizingMaskIntoConstraints = NO; [self.view addSubview:_localVideoView]; // 远程视频视图 _remoteVideoView = [[UIView alloc] init]; _remoteVideoView.backgroundColor = UIColor.darkGrayColor; _remoteVideoView.translatesAutoresizingMaskIntoConstraints = NO; [self.view addSubview:_remoteVideoView]; // 连接按钮 _connectButton = [UIButton buttonWithType:UIButtonTypeSystem]; [_connectButton setTitle:@\"建立连接\" forState:UIControlStateNormal]; [_connectButton addTarget:self action:@selector(connectButtonTapped) forControlEvents:UIControlEventTouchUpInside]; _connectButton.translatesAutoresizingMaskIntoConstraints = NO; [self.view addSubview:_connectButton]; // 布局 [NSLayoutConstraint activateConstraints:@[ // 本地视频（右上角小窗口） [_localVideoView topAnchor constraintEqualToAnchor:self.view.safeAreaLayoutGuide.topAnchor constant:20], [_localVideoView.trailingAnchor constraintEqualToAnchor:self.view.trailingAnchor constant:-20], [_localVideoView.widthAnchor constraintEqualToConstant:120], [_localVideoView.heightAnchor constraintEqualToConstant:180], // 远程视频（全屏） [_remoteVideoView.topAnchor constraintEqualToAnchor:self.view.safeAreaLayoutGuide.topAnchor], [_remoteVideoView.leadingAnchor constraintEqualToAnchor:self.view.leadingAnchor], [_remoteVideoView.trailingAnchor constraintEqualToAnchor:self.view.trailingAnchor], [_remoteVideoView.bottomAnchor constraintEqualToAnchor:self.connectButton.topAnchor constant:-20], // 连接按钮 [_connectButton.bottomAnchor constraintEqualToAnchor:self.view.safeAreaLayoutGuide.bottomAnchor constant:-20], [_connectButton.centerXAnchor constraintEqualToAnchor:self.view.centerXAnchor], [_connectButton.widthAnchor constraintEqualToConstant:120], [_connectButton.heightAnchor constraintEqualToConstant:44] ]];}- (void)setupWebRTC { _rtcManager = [[WebRTCManager alloc] initWithDelegate:self]; [_rtcManager setupLocalStreamWithVideoView:self.localVideoView];}- (void)connectButtonTapped { [_rtcManager createOffer];}#pragma mark - WebRTCManagerDelegate// 处理收到的远程流- (void)didReceiveRemoteStream:(RTCMediaStream *)stream { dispatch_async(dispatch_get_main_queue(), ^{ // 渲染远程视频 if (stream.videoTracks.count > 0) { RTCVideoTrack *remoteVideoTrack = stream.videoTracks[0]; RTCMTLVideoView *rendererView = [[RTCMTLVideoView alloc] init]; rendererView.frame = self.remoteVideoView.bounds; rendererView.autoresizingMask = UIViewAutoresizingFlexibleWidth | UIViewAutoresizingFlexibleHeight; [self.remoteVideoView addSubview:rendererView]; [remoteVideoTrack addRenderer:[[RTCVideoRendererAdapter alloc] initWithRenderer:rendererView]]; } });}- (void)dealloc { [_rtcManager disconnect];}@end

代码说明
这个示例实现了 iOS 平台上基于 WebRTC 的基本音视频推拉流功能，主要包含以下部分：
WebRTCManager：核心管理类，负责：

  • 初始化 WebRTC 相关组件
  • 处理本地音视频流的捕获和预览
  • 管理 PeerConnection 连接
  • 处理 SDP 交换和 ICE 候选者

ViewController：界面控制器，负责：

  • 创建本地和远程视频的预览视图
  • 处理用户交互（如建立连接）
  • 渲染远程视频流

三、使用说明

这个示例缺少信令服务器的实现，你需要自己搭建一个信令服务器
在实际使用中，需要将代码中注释掉的信令发送部分替换为实际的网络请求
代码中的 STUN 服务器使用了 Google 的公共服务器，生产环境中建议使用自己的 STUN/TURN 服务器

四、扩展建议

  • 添加错误处理和重连机制
  • 实现多人通话功能
  • 添加视频质量控制
  • 实现屏幕共享功能
  • 添加音频 /video 开关控制

这个示例仅仅提供了基础的推拉流框架，你可以根据实际需求进行扩展和优化。

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