WebRTC音视频通话-实现iOS端调用ossrs视频通话服务

之前搭建ossrs服务，可以查看：https://blog.csdn.net/gloryFlow/article/details/132257196
这里iOS端使用GoogleWebRTC联调ossrs实现视频通话功能。

一、iOS端调用ossrs视频通话效果图

iOS端端效果图

ossrs效果图

一、WebRTC是什么？

WebRTC (Web Real-Time Communications) 是一项实时通讯技术，它允许网络应用或者站点，在不借助中间媒介的情况下，建立浏览器之间点对点（Peer-to-Peer）的连接，实现视频流、音频流或者其他任意数据的传输。

查看https://zhuanlan.zhihu.com/p/421503695

需要了解的关键

• NAT
  Network Address Translation(网络地址转换)
• STUN
  Session Traversal Utilities for NAT(NAT会话穿越应用程序)
• TURN
  Traversal Using Relay NAT(通过Relay方式穿越NAT)
• ICE
  Interactive Connectivity Establishment(交互式连接建立)
• SDP
  Session Description Protocol(会话描述协议)
• WebRTC
  Web Real-Time Communications(web实时通讯技术)

WebRTC offer交换流程如图所示

二、实现iOS端调用ossrs视频通话

创建好实现iOS端调用ossrs视频通话的工程。如果使用P2P点对点的音视频通话，信令服务器，stun/trunP2P穿透和转发服务器这类需要自己搭建了。ossrs中包含stun/trun穿透和转发服务器。我这边实现iOS端调用ossrs服务。

2.1、权限设置

在iOS端调用ossrs视频通话需要相机、语音权限

在info.plist中添加

<key>NSCameraUsageDescription</key>
<string>APP需要获取相机权限</string>
<key>NSMicrophoneUsageDescription</key>
<string>APP需要获取麦克风权限</string>

2.2、工程需要用到GoogleWebRTC

工程需要用到GoogleWebRTC库，在podfile文件中引入库，注意不同版本的GoogleWebRTC代码还是有些差别的。

target 'WebRTCApp' do
    
pod 'GoogleWebRTC'
pod 'ReactiveObjC'
pod 'SocketRocket'
pod 'HGAlertViewController', '~> 1.0.1'

end

之后执行pod install

2.3、GoogleWebRTC主要API

在使用GoogleWebRTC前，先看下主要的类

• RTCPeerConnection

RTCPeerConnection是WebRTC用于构建点对点连接器

• RTCPeerConnectionFactory

RTCPeerConnectionFactory是RTCPeerConnection工厂类

• RTCVideoCapturer

RTCVideoCapturer是摄像头采集器，获取画面与音频，这个之后可以替换掉。可以自定义，方便获取CMSampleBufferRef进行画面的美颜滤镜、虚拟头像等处理。

• RTCVideoTrack

RTCVideoTrack是视频轨Track

• RTCAudioTrack

RTCAudioTrack是音频轨Track

• RTCDataChannel

RTCDataChannel是建立高吞吐量、低延时的信道，可以传输数据。

• RTCMediaStream

RTCMediaStream是媒体流（摄像头的视频、麦克风的音频）的同步流。

• SDP

SDP即Session Description Protocol(会话描述协议)
SDP由一行或多行UTF-8文本组成，每行以一个字符的类型开头，后跟等号(=)，然后是包含值或描述的结构化文本，其格式取决于类型。如下为一个SDP内容示例：

v=0
o=alice 2890844526 2890844526 IN IP4
s=
c=IN IP4
t=0 0
m=audio 49170 RTP/AVP 0
a=rtpmap:0 PCMU/8000
m=video 51372 RTP/AVP 31
a=rtpmap:31 H261/90000
m=video 53000 RTP/AVP 32
a=rtpmap:32 MPV/90000

这是会用到的WebRTC主要的API类。

2.4、使用WebRTC代码实现

使用WebRTC实现P2P音视频流程如图

这里调用ossrs实现步骤如下

关键点设置

初始化RTCPeerConnectionFactory

#pragma mark - Lazy
- (RTCPeerConnectionFactory *)factory {
    if (!_factory) {
        RTCInitializeSSL();
        RTCDefaultVideoEncoderFactory *videoEncoderFactory = [[RTCDefaultVideoEncoderFactory alloc] init];
        
        RTCDefaultVideoDecoderFactory *videoDecoderFactory = [[RTCDefaultVideoDecoderFactory alloc] init];
        
        _factory = [[RTCPeerConnectionFactory alloc] initWithEncoderFactory:videoEncoderFactory decoderFactory:videoDecoderFactory];
    }
    return _factory;
}

通过RTCPeerConnectionFactory生成RTCPeerConnection

self.peerConnection = [self.factory peerConnectionWithConfiguration:newConfig constraints:constraints delegate:nil];

将RTCAudioTrack及RTCVideoTrack添加到peerConnection

NSString *streamId = @"stream";

// Audio
RTCAudioTrack *audioTrack = [self createAudioTrack];
self.localAudioTrack = audioTrack;

RTCRtpTransceiverInit *audioTrackTransceiver = [[RTCRtpTransceiverInit alloc] init];
audioTrackTransceiver.direction = RTCRtpTransceiverDirectionSendOnly;
audioTrackTransceiver.streamIds = @[streamId];

[self.peerConnection addTransceiverWithTrack:audioTrack init:audioTrackTransceiver];

// Video
RTCVideoTrack *videoTrack = [self createVideoTrack];
self.localVideoTrack = videoTrack;
RTCRtpTransceiverInit *videoTrackTransceiver = [[RTCRtpTransceiverInit alloc] init];
videoTrackTransceiver.direction = RTCRtpTransceiverDirectionSendOnly;
videoTrackTransceiver.streamIds = @[streamId];
[self.peerConnection addTransceiverWithTrack:videoTrack init:videoTrackTransceiver];

设置摄像头RTCCameraVideoCapturer及文件视频Capturer

- (RTCVideoTrack *)createVideoTrack {
    RTCVideoSource *videoSource = [self.factory videoSource];
    
    // 经过测试比1920*1080大的尺寸，无法通过srs播放
    [videoSource adaptOutputFormatToWidth:1920 height:1080 fps:20];

    // 如果是模拟器
    if (TARGET_IPHONE_SIMULATOR) {
        self.videoCapturer = [[RTCFileVideoCapturer alloc] initWithDelegate:videoSource];
    } else{
        self.videoCapturer = [[RTCCameraVideoCapturer alloc] initWithDelegate:videoSource];
    }
    
    RTCVideoTrack *videoTrack = [self.factory videoTrackWithSource:videoSource trackId:@"video0"];
    
    return videoTrack;
}

摄像头本地采集的画面本地显示
startCaptureLocalVideo的renderer为RTCEAGLVideoView

- (void)startCaptureLocalVideo:(id<RTCVideoRenderer>)renderer {
    if (!self.isPublish) {
        return;
    }
    
    if (!renderer) {
        return;
    }
    
    if (!self.videoCapturer) {
        return;
    }
    
    
    RTCVideoCapturer *capturer = self.videoCapturer;
    if ([capturer isKindOfClass:[RTCCameraVideoCapturer class]]) {
        if (!([RTCCameraVideoCapturer captureDevices].count > 0)) {
            return;
        }
        AVCaptureDevice *frontCamera = RTCCameraVideoCapturer.captureDevices.firstObject;
//        if (frontCamera.position != AVCaptureDevicePositionFront) {
//            return;
//        }
        
        RTCCameraVideoCapturer *cameraVideoCapturer = (RTCCameraVideoCapturer *)capturer;
        
        AVCaptureDeviceFormat *formatNilable;
        
        NSArray *supportDeviceFormats = [RTCCameraVideoCapturer supportedFormatsForDevice:frontCamera];
        NSLog(@"supportDeviceFormats:%@",supportDeviceFormats);
        formatNilable = supportDeviceFormats[4];
//        if (supportDeviceFormats && supportDeviceFormats.count > 0) {
//            NSMutableArray *formats = [NSMutableArray arrayWithCapacity:0];
//            for (AVCaptureDeviceFormat *format in supportDeviceFormats) {
//                CMVideoDimensions videoVideoDimensions = CMVideoFormatDescriptionGetDimensions(format.formatDescription);
//                float width = videoVideoDimensions.width;
//                float height = videoVideoDimensions.height;
//                // only use 16:9 format.
//                if ((width / height) >= (16.0/9.0)) {
//                    [formats addObject:format];
//                }
//            }
//
//            if (formats.count > 0) {
//                NSArray *sortedFormats = [formats sortedArrayUsingComparator:^NSComparisonResult(AVCaptureDeviceFormat *obj1, AVCaptureDeviceFormat *obj2) {
//                    CMVideoDimensions f1VD = CMVideoFormatDescriptionGetDimensions(obj1.formatDescription);
//                    CMVideoDimensions f2VD = CMVideoFormatDescriptionGetDimensions(obj2.formatDescription);
//                    float width1 = f1VD.width;
//                    float width2 = f2VD.width;
//                    float height2 = f2VD.height;
//                    // only use 16:9 format.
//                    if ((width2 / height2) >= (1.7)) {
//                        return NSOrderedAscending;
//                    } else {
//                        return NSOrderedDescending;
//                    }
//                }];
//
//                if (sortedFormats && sortedFormats.count > 0) {
//                    formatNilable = sortedFormats.lastObject;
//                }
//            }
//        }
        
        if (!formatNilable) {
            return;
        }
        
        NSArray *formatArr = [RTCCameraVideoCapturer supportedFormatsForDevice:frontCamera];
        for (AVCaptureDeviceFormat *format in formatArr) {
            NSLog(@"AVCaptureDeviceFormat format:%@", format);
        }
        
        [cameraVideoCapturer startCaptureWithDevice:frontCamera format:formatNilable fps:20 completionHandler:^(NSError *error) {
            NSLog(@"startCaptureWithDevice error:%@", error);
        }];
    }
    
    if ([capturer isKindOfClass:[RTCFileVideoCapturer class]]) {
        RTCFileVideoCapturer *fileVideoCapturer = (RTCFileVideoCapturer *)capturer;
        [fileVideoCapturer startCapturingFromFileNamed:@"beautyPicture.mp4" onError:^(NSError * _Nonnull error) {
            NSLog(@"startCaptureLocalVideo startCapturingFromFileNamed error:%@", error);
        }];
    }
    
    [self.localVideoTrack addRenderer:renderer];
}

创建的createOffer

- (void)offer:(void (^)(RTCSessionDescription *sdp))completion {
    if (self.isPublish) {
        self.mediaConstrains = self.publishMediaConstrains;
    } else {
        self.mediaConstrains = self.playMediaConstrains;
    }
    
    RTCMediaConstraints *constrains = [[RTCMediaConstraints alloc] initWithMandatoryConstraints:self.mediaConstrains optionalConstraints:self.optionalConstraints];
    NSLog(@"peerConnection:%@",self.peerConnection);
    __weak typeof(self) weakSelf = self;
    [weakSelf.peerConnection offerForConstraints:constrains completionHandler:^(RTCSessionDescription * _Nullable sdp, NSError * _Nullable error) {
        if (error) {
            NSLog(@"offer offerForConstraints error:%@", error);
        }
        
        if (sdp) {
            [weakSelf.peerConnection setLocalDescription:sdp completionHandler:^(NSError * _Nullable error) {
                if (error) {
                    NSLog(@"offer setLocalDescription error:%@", error);
                }
                if (completion) {
                    completion(sdp);
                }
            }];
        }
    }];
}

设置setRemoteDescription

- (void)setRemoteSdp:(RTCSessionDescription *)remoteSdp completion:(void (^)(NSError * _Nullable error))completion {
    [self.peerConnection setRemoteDescription:remoteSdp completionHandler:completion];
}

整体代码如下

WebRTCClient.h

#import <Foundation/Foundation.h>
#import <WebRTC/WebRTC.h>
#import <UIKit/UIKit.h>

@protocol WebRTCClientDelegate;
@interface WebRTCClient : NSObject

@property (nonatomic, weak) id<WebRTCClientDelegate> delegate;

/**
 connect工厂
 */
@property (nonatomic, strong) RTCPeerConnectionFactory *factory;

/**
 是否push
 */
@property (nonatomic, assign) BOOL isPublish;

/**
 connect
 */
@property (nonatomic, strong) RTCPeerConnection *peerConnection;

/**
 RTCAudioSession
 */
@property (nonatomic, strong) RTCAudioSession *rtcAudioSession;

/**
 DispatchQueue
 */
@property (nonatomic) dispatch_queue_t audioQueue;

/**
 mediaConstrains
 */
@property (nonatomic, strong) NSDictionary *mediaConstrains;

/**
 publishMediaConstrains
 */
@property (nonatomic, strong) NSDictionary *publishMediaConstrains;

/**
 playMediaConstrains
 */
@property (nonatomic, strong) NSDictionary *playMediaConstrains;

/**
 optionalConstraints
 */
@property (nonatomic, strong) NSDictionary *optionalConstraints;

/**
 RTCVideoCapturer摄像头采集器
 */
@property (nonatomic, strong) RTCVideoCapturer *videoCapturer;

/**
 local语音localAudioTrack
 */
@property (nonatomic, strong) RTCAudioTrack *localAudioTrack;

/**
 localVideoTrack
 */
@property (nonatomic, strong) RTCVideoTrack *localVideoTrack;

/**
 remoteVideoTrack
 */
@property (nonatomic, strong) RTCVideoTrack *remoteVideoTrack;

/**
 RTCVideoRenderer
 */
@property (nonatomic, weak) id<RTCVideoRenderer> remoteRenderView;

/**
 localDataChannel
 */
@property (nonatomic, strong) RTCDataChannel *localDataChannel;

/**
 localDataChannel
 */
@property (nonatomic, strong) RTCDataChannel *remoteDataChannel;


- (instancetype)initWithPublish:(BOOL)isPublish;

- (void)startCaptureLocalVideo:(id<RTCVideoRenderer>)renderer;

- (void)answer:(void (^)(RTCSessionDescription *sdp))completionHandler;

- (void)offer:(void (^)(RTCSessionDescription *sdp))completionHandler;

#pragma mark - Hiden or show Video
- (void)hidenVideo;

- (void)showVideo;

#pragma mark - Hiden or show Audio
- (void)muteAudio;

- (void)unmuteAudio;

- (void)speakOff;

- (void)speakOn;

- (void)changeSDP2Server:(RTCSessionDescription *)sdp
                  urlStr:(NSString *)urlStr
               streamUrl:(NSString *)streamUrl
                 closure:(void (^)(BOOL isServerRetSuc))closure;

@end


@protocol WebRTCClientDelegate <NSObject>

- (void)webRTCClient:(WebRTCClient *)client didDiscoverLocalCandidate:(RTCIceCandidate *)candidate;
- (void)webRTCClient:(WebRTCClient *)client didChangeConnectionState:(RTCIceConnectionState)state;
- (void)webRTCClient:(WebRTCClient *)client didReceiveData:(NSData *)data;

@end

WebRTCClient.m

#import "WebRTCClient.h"
#import "HttpClient.h"

@interface WebRTCClient ()<RTCPeerConnectionDelegate, RTCDataChannelDelegate>

@property (nonatomic, strong) HttpClient *httpClient;

@end

@implementation WebRTCClient

- (instancetype)initWithPublish:(BOOL)isPublish {
    self = [super init];
    if (self) {
        self.isPublish = isPublish;
        
        self.httpClient = [[HttpClient alloc] init];
                        
        RTCMediaConstraints *constraints = [[RTCMediaConstraints alloc] initWithMandatoryConstraints:nil optionalConstraints:self.optionalConstraints];
        
        RTCConfiguration *newConfig = [[RTCConfiguration alloc] init];
        newConfig.sdpSemantics = RTCSdpSemanticsUnifiedPlan;
        
        self.peerConnection = [self.factory peerConnectionWithConfiguration:newConfig constraints:constraints delegate:nil];
        
        [self createMediaSenders];
        
        [self createMediaReceivers];
        
        // srs not support data channel.
        // self.createDataChannel()
        [self configureAudioSession];
        
        self.peerConnection.delegate = self;
    }
    return self;
}


- (void)createMediaSenders {
    if (!self.isPublish) {
        return;
    }
    
    NSString *streamId = @"stream";
    
    // Audio
    RTCAudioTrack *audioTrack = [self createAudioTrack];
    self.localAudioTrack = audioTrack;
    
    RTCRtpTransceiverInit *audioTrackTransceiver = [[RTCRtpTransceiverInit alloc] init];
    audioTrackTransceiver.direction = RTCRtpTransceiverDirectionSendOnly;
    audioTrackTransceiver.streamIds = @[streamId];
    
    [self.peerConnection addTransceiverWithTrack:audioTrack init:audioTrackTransceiver];
    
    // Video
    RTCVideoTrack *videoTrack = [self createVideoTrack];
    self.localVideoTrack = videoTrack;
    RTCRtpTransceiverInit *videoTrackTransceiver = [[RTCRtpTransceiverInit alloc] init];
    videoTrackTransceiver.direction = RTCRtpTransceiverDirectionSendOnly;
    videoTrackTransceiver.streamIds = @[streamId];
    [self.peerConnection addTransceiverWithTrack:videoTrack init:videoTrackTransceiver];
}

- (void)createMediaReceivers {
    if (!self.isPublish) {
        return;
    }
    
    if (self.peerConnection.transceivers.count > 0) {
        RTCRtpTransceiver *transceiver = self.peerConnection.transceivers.firstObject;
        if (transceiver.mediaType == RTCRtpMediaTypeVideo) {
            RTCVideoTrack *track = (RTCVideoTrack *)transceiver.receiver.track;
            self.remoteVideoTrack = track;
        }
    }
}

- (void)configureAudioSession {
    [self.rtcAudioSession lockForConfiguration];
    @try {
        NSError *error;
        [self.rtcAudioSession setCategory:AVAudioSessionCategoryPlayAndRecord withOptions:AVAudioSessionCategoryOptionDefaultToSpeaker error:&error];
        
        NSError *modeError;
        [self.rtcAudioSession setMode:AVAudioSessionModeVoiceChat error:&modeError];
        NSLog(@"configureAudioSession error:%@, modeError:%@", error, modeError);
    } @catch (NSException *exception) {
        NSLog(@"configureAudioSession exception:%@", exception);
    }
    
    [self.rtcAudioSession unlockForConfiguration];
}

- (RTCAudioTrack *)createAudioTrack {
    /// enable google 3A algorithm.
    NSDictionary *mandatory = @{
        @"googEchoCancellation": kRTCMediaConstraintsValueTrue,
        @"googAutoGainControl": kRTCMediaConstraintsValueTrue,
        @"googNoiseSuppression": kRTCMediaConstraintsValueTrue,
    };
    
    RTCMediaConstraints *audioConstrains = [[RTCMediaConstraints alloc] initWithMandatoryConstraints:mandatory optionalConstraints:self.optionalConstraints];
    RTCAudioSource *audioSource = [self.factory audioSourceWithConstraints:audioConstrains];
    
    RTCAudioTrack *audioTrack = [self.factory audioTrackWithSource:audioSource trackId:@"audio0"];
    
    return audioTrack;
}

- (RTCVideoTrack *)createVideoTrack {
    RTCVideoSource *videoSource = [self.factory videoSource];
    
    // 经过测试比1920*1080大的尺寸，无法通过srs播放
    [videoSource adaptOutputFormatToWidth:1920 height:1080 fps:20];

    // 如果是模拟器
    if (TARGET_IPHONE_SIMULATOR) {
        self.videoCapturer = [[RTCFileVideoCapturer alloc] initWithDelegate:videoSource];
    } else{
        self.videoCapturer = [[RTCCameraVideoCapturer alloc] initWithDelegate:videoSource];
    }
    
    RTCVideoTrack *videoTrack = [self.factory videoTrackWithSource:videoSource trackId:@"video0"];
    
    return videoTrack;
}

- (void)offer:(void (^)(RTCSessionDescription *sdp))completion {
    if (self.isPublish) {
        self.mediaConstrains = self.publishMediaConstrains;
    } else {
        self.mediaConstrains = self.playMediaConstrains;
    }
    
    RTCMediaConstraints *constrains = [[RTCMediaConstraints alloc] initWithMandatoryConstraints:self.mediaConstrains optionalConstraints:self.optionalConstraints];
    NSLog(@"peerConnection:%@",self.peerConnection);
    __weak typeof(self) weakSelf = self;
    [weakSelf.peerConnection offerForConstraints:constrains completionHandler:^(RTCSessionDescription * _Nullable sdp, NSError * _Nullable error) {
        if (error) {
            NSLog(@"offer offerForConstraints error:%@", error);
        }
        
        if (sdp) {
            [weakSelf.peerConnection setLocalDescription:sdp completionHandler:^(NSError * _Nullable error) {
                if (error) {
                    NSLog(@"offer setLocalDescription error:%@", error);
                }
                if (completion) {
                    completion(sdp);
                }
            }];
        }
    }];
}

- (void)answer:(void (^)(RTCSessionDescription *sdp))completion {
    RTCMediaConstraints *constrains = [[RTCMediaConstraints alloc] initWithMandatoryConstraints:self.mediaConstrains optionalConstraints:self.optionalConstraints];
    __weak typeof(self) weakSelf = self;
    [weakSelf.peerConnection answerForConstraints:constrains completionHandler:^(RTCSessionDescription * _Nullable sdp, NSError * _Nullable error) {
        if (error) {
            NSLog(@"answer answerForConstraints error:%@", error);
        }
        
        if (sdp) {
            [weakSelf.peerConnection setLocalDescription:sdp completionHandler:^(NSError * _Nullable error) {
                if (error) {
                    NSLog(@"answer setLocalDescription error:%@", error);
                }
                if (completion) {
                    completion(sdp);
                }
            }];
        }
    }];
}

- (void)setRemoteSdp:(RTCSessionDescription *)remoteSdp completion:(void (^)(NSError * _Nullable error))completion {
    [self.peerConnection setRemoteDescription:remoteSdp completionHandler:completion];
}

- (void)setRemoteCandidate:(RTCIceCandidate *)remoteCandidate {
    [self.peerConnection addIceCandidate:remoteCandidate];
}

- (void)setMaxBitrate:(int)maxBitrate {
    NSMutableArray *videoSenders = [NSMutableArray arrayWithCapacity:0];
    for (RTCRtpSender *sender in self.peerConnection.senders) {
        if (sender.track && [kRTCMediaStreamTrackKindVideo isEqualToString:sender.track.kind]) {
            [videoSenders addObject:sender];
        }
    }
    
    if (videoSenders.count > 0) {
        RTCRtpSender *firstSender = [videoSenders firstObject];
        RTCRtpParameters *parameters = firstSender.parameters;
        NSNumber *maxBitrateBps = [NSNumber numberWithInt:maxBitrate];
        parameters.encodings.firstObject.maxBitrateBps = maxBitrateBps;
    }
}

- (void)setMaxFramerate:(int)maxFramerate {
    NSMutableArray *videoSenders = [NSMutableArray arrayWithCapacity:0];
    for (RTCRtpSender *sender in self.peerConnection.senders) {
        if (sender.track && [kRTCMediaStreamTrackKindVideo isEqualToString:sender.track.kind]) {
            [videoSenders addObject:sender];
        }
    }
    
    if (videoSenders.count > 0) {
        RTCRtpSender *firstSender = [videoSenders firstObject];
        RTCRtpParameters *parameters = firstSender.parameters;
        NSNumber *maxFramerateNum = [NSNumber numberWithInt:maxFramerate];
        
        // 该版本暂时没有maxFramerate，需要更新到最新版本
        parameters.encodings.firstObject.maxFramerate = maxFramerateNum;
    }
}

- (void)startCaptureLocalVideo:(id<RTCVideoRenderer>)renderer {
    if (!self.isPublish) {
        return;
    }
    
    if (!renderer) {
        return;
    }
    
    if (!self.videoCapturer) {
        return;
    }
    
    
    RTCVideoCapturer *capturer = self.videoCapturer;
    if ([capturer isKindOfClass:[RTCCameraVideoCapturer class]]) {
        if (!([RTCCameraVideoCapturer captureDevices].count > 0)) {
            return;
        }
        AVCaptureDevice *frontCamera = RTCCameraVideoCapturer.captureDevices.firstObject;
//        if (frontCamera.position != AVCaptureDevicePositionFront) {
//            return;
//        }
        
        RTCCameraVideoCapturer *cameraVideoCapturer = (RTCCameraVideoCapturer *)capturer;
        
        AVCaptureDeviceFormat *formatNilable;
        
        NSArray *supportDeviceFormats = [RTCCameraVideoCapturer supportedFormatsForDevice:frontCamera];
        NSLog(@"supportDeviceFormats:%@",supportDeviceFormats);
        formatNilable = supportDeviceFormats[4];
//        if (supportDeviceFormats && supportDeviceFormats.count > 0) {
//            NSMutableArray *formats = [NSMutableArray arrayWithCapacity:0];
//            for (AVCaptureDeviceFormat *format in supportDeviceFormats) {
//                CMVideoDimensions videoVideoDimensions = CMVideoFormatDescriptionGetDimensions(format.formatDescription);
//                float width = videoVideoDimensions.width;
//                float height = videoVideoDimensions.height;
//                // only use 16:9 format.
//                if ((width / height) >= (16.0/9.0)) {
//                    [formats addObject:format];
//                }
//            }
//
//            if (formats.count > 0) {
//                NSArray *sortedFormats = [formats sortedArrayUsingComparator:^NSComparisonResult(AVCaptureDeviceFormat *obj1, AVCaptureDeviceFormat *obj2) {
//                    CMVideoDimensions f1VD = CMVideoFormatDescriptionGetDimensions(obj1.formatDescription);
//                    CMVideoDimensions f2VD = CMVideoFormatDescriptionGetDimensions(obj2.formatDescription);
//                    float width1 = f1VD.width;
//                    float width2 = f2VD.width;
//                    float height2 = f2VD.height;
//                    // only use 16:9 format.
//                    if ((width2 / height2) >= (1.7)) {
//                        return NSOrderedAscending;
//                    } else {
//                        return NSOrderedDescending;
//                    }
//                }];
//
//                if (sortedFormats && sortedFormats.count > 0) {
//                    formatNilable = sortedFormats.lastObject;
//                }
//            }
//        }
        
        if (!formatNilable) {
            return;
        }
        
        NSArray *formatArr = [RTCCameraVideoCapturer supportedFormatsForDevice:frontCamera];
        for (AVCaptureDeviceFormat *format in formatArr) {
            NSLog(@"AVCaptureDeviceFormat format:%@", format);
        }
        
        [cameraVideoCapturer startCaptureWithDevice:frontCamera format:formatNilable fps:20 completionHandler:^(NSError *error) {
            NSLog(@"startCaptureWithDevice error:%@", error);
        }];
    }
    
    if ([capturer isKindOfClass:[RTCFileVideoCapturer class]]) {
        RTCFileVideoCapturer *fileVideoCapturer = (RTCFileVideoCapturer *)capturer;
        [fileVideoCapturer startCapturingFromFileNamed:@"beautyPicture.mp4" onError:^(NSError * _Nonnull error) {
            NSLog(@"startCaptureLocalVideo startCapturingFromFileNamed error:%@", error);
        }];
    }
    
    [self.localVideoTrack addRenderer:renderer];
}

- (void)renderRemoteVideo:(id<RTCVideoRenderer>)renderer {
    if (!self.isPublish) {
        return;
    }
    
    self.remoteRenderView = renderer;
}

- (RTCDataChannel *)createDataChannel {
    RTCDataChannelConfiguration *config = [[RTCDataChannelConfiguration alloc] init];
    RTCDataChannel *dataChannel = [self.peerConnection dataChannelForLabel:@"WebRTCData" configuration:config];
    if (!dataChannel) {
        return nil;
    }
    
    dataChannel.delegate = self;
    self.localDataChannel = dataChannel;
    return dataChannel;
}

- (void)sendData:(NSData *)data {
    RTCDataBuffer *buffer = [[RTCDataBuffer alloc] initWithData:data isBinary:YES];
    [self.remoteDataChannel sendData:buffer];
}

- (void)changeSDP2Server:(RTCSessionDescription *)sdp
                  urlStr:(NSString *)urlStr
               streamUrl:(NSString *)streamUrl
                 closure:(void (^)(BOOL isServerRetSuc))closure {
    
    __weak typeof(self) weakSelf = self;
    [self.httpClient changeSDP2Server:sdp urlStr:urlStr streamUrl:streamUrl closure:^(NSDictionary *result) {
        if (result && [result isKindOfClass:[NSDictionary class]]) {
            NSString *sdp = [result objectForKey:@"sdp"];
            if (sdp && [sdp isKindOfClass:[NSString class]] && sdp.length > 0) {
                RTCSessionDescription *remoteSDP = [[RTCSessionDescription alloc] initWithType:RTCSdpTypeAnswer sdp:sdp];
                [weakSelf setRemoteSdp:remoteSDP completion:^(NSError * _Nullable error) {
                    NSLog(@"changeSDP2Server setRemoteDescription error:%@", error);
                }];
            }
        }
    }];
}

#pragma mark - Hiden or show Video
- (void)hidenVideo {
    [self setVideoEnabled:NO];
}

- (void)showVideo {
    [self setVideoEnabled:YES];
}

- (void)setVideoEnabled:(BOOL)isEnabled {
    [self setTrackEnabled:[RTCVideoTrack class] isEnabled:isEnabled];
}

- (void)setTrackEnabled:(Class)track isEnabled:(BOOL)isEnabled {
    for (RTCRtpTransceiver *transceiver in self.peerConnection.transceivers) {
        if (transceiver && [transceiver isKindOfClass:track]) {
            transceiver.sender.track.isEnabled = isEnabled;
        }
    }
}


#pragma mark - Hiden or show Audio
- (void)muteAudio {
    [self setAudioEnabled:NO];
}

- (void)unmuteAudio {
    [self setAudioEnabled:YES];
}

- (void)speakOff {
    __weak typeof(self) weakSelf = self;
    dispatch_async(self.audioQueue, ^{
        [weakSelf.rtcAudioSession lockForConfiguration];
        @try {
            
            NSError *error;
            [self.rtcAudioSession setCategory:AVAudioSessionCategoryPlayAndRecord withOptions:AVAudioSessionCategoryOptionDefaultToSpeaker error:&error];
            
            NSError *ooapError;
            [self.rtcAudioSession overrideOutputAudioPort:AVAudioSessionPortOverrideNone error:&ooapError];
            NSLog(@"speakOff error:%@, ooapError:%@", error, ooapError);
        } @catch (NSException *exception) {
            NSLog(@"speakOff exception:%@", exception);
        }
        [weakSelf.rtcAudioSession unlockForConfiguration];
    });
}

- (void)speakOn {
    __weak typeof(self) weakSelf = self;
    dispatch_async(self.audioQueue, ^{
        [weakSelf.rtcAudioSession lockForConfiguration];
        @try {
            
            NSError *error;
            [self.rtcAudioSession setCategory:AVAudioSessionCategoryPlayAndRecord withOptions:AVAudioSessionCategoryOptionDefaultToSpeaker error:&error];
            
            NSError *ooapError;
            [self.rtcAudioSession overrideOutputAudioPort:AVAudioSessionPortOverrideSpeaker error:&ooapError];
            
            NSError *activeError;
            [self.rtcAudioSession setActive:YES error:&activeError];
            
            NSLog(@"speakOn error:%@, ooapError:%@, activeError:%@", error, ooapError, activeError);
        } @catch (NSException *exception) {
            NSLog(@"speakOn exception:%@", exception);
        }
        [weakSelf.rtcAudioSession unlockForConfiguration];
    });
}

- (void)setAudioEnabled:(BOOL)isEnabled {
    [self setTrackEnabled:[RTCAudioTrack class] isEnabled:isEnabled];
}


#pragma mark - RTCPeerConnectionDelegate
/** Called when the SignalingState changed. */
- (void)peerConnection:(RTCPeerConnection *)peerConnection
didChangeSignalingState:(RTCSignalingState)stateChanged {
    NSLog(@"peerConnection didChangeSignalingState:%ld", (long)stateChanged);
}

/** Called when media is received on a new stream from remote peer. */
- (void)peerConnection:(RTCPeerConnection *)peerConnection didAddStream:(RTCMediaStream *)stream {
    NSLog(@"peerConnection didAddStream");
    if (self.isPublish) {
        return;
    }
    
    NSArray *videoTracks = stream.videoTracks;
    if (videoTracks && videoTracks.count > 0) {
        RTCVideoTrack *track = videoTracks.firstObject;
        self.remoteVideoTrack = track;
    }
    
    if (self.remoteVideoTrack && self.remoteRenderView) {
        id<RTCVideoRenderer> remoteRenderView = self.remoteRenderView;
        RTCVideoTrack *remoteVideoTrack = self.remoteVideoTrack;
        [remoteVideoTrack addRenderer:remoteRenderView];
    }
    
    /**
     if let audioTrack = stream.audioTracks.first{
     print("audio track faund")
     audioTrack.source.volume = 8
     }
     */
}

/** Called when a remote peer closes a stream.
 *  This is not called when RTCSdpSemanticsUnifiedPlan is specified.
 */
- (void)peerConnection:(RTCPeerConnection *)peerConnection didRemoveStream:(RTCMediaStream *)stream {
    NSLog(@"peerConnection didRemoveStream");
}

/** Called when negotiation is needed, for example ICE has restarted. */
- (void)peerConnectionShouldNegotiate:(RTCPeerConnection *)peerConnection {
    NSLog(@"peerConnection peerConnectionShouldNegotiate");
}

/** Called any time the IceConnectionState changes. */
- (void)peerConnection:(RTCPeerConnection *)peerConnection
    didChangeIceConnectionState:(RTCIceConnectionState)newState {
    NSLog(@"peerConnection didChangeIceConnectionState:%ld", newState);
    if (self.delegate && [self.delegate respondsToSelector:@selector(webRTCClient:didChangeConnectionState:)]) {
        [self.delegate webRTCClient:self didChangeConnectionState:newState];
    }
}

/** Called any time the IceGatheringState changes. */
- (void)peerConnection:(RTCPeerConnection *)peerConnection
    didChangeIceGatheringState:(RTCIceGatheringState)newState {
    NSLog(@"peerConnection didChangeIceGatheringState:%ld", newState);
}

/** New ice candidate has been found. */
- (void)peerConnection:(RTCPeerConnection *)peerConnection
    didGenerateIceCandidate:(RTCIceCandidate *)candidate {
    NSLog(@"peerConnection didGenerateIceCandidate:%@", candidate);
    if (self.delegate && [self.delegate respondsToSelector:@selector(webRTCClient:didDiscoverLocalCandidate:)]) {
        [self.delegate webRTCClient:self didDiscoverLocalCandidate:candidate];
    }
}

/** Called when a group of local Ice candidates have been removed. */
- (void)peerConnection:(RTCPeerConnection *)peerConnection
    didRemoveIceCandidates:(NSArray<RTCIceCandidate *> *)candidates {
    NSLog(@"peerConnection didRemoveIceCandidates:%@", candidates);
}

/** New data channel has been opened. */
- (void)peerConnection:(RTCPeerConnection *)peerConnection
    didOpenDataChannel:(RTCDataChannel *)dataChannel {
    NSLog(@"peerConnection didOpenDataChannel:%@", dataChannel);
    self.remoteDataChannel = dataChannel;
}

/** Called when signaling indicates a transceiver will be receiving media from
 *  the remote endpoint.
 *  This is only called with RTCSdpSemanticsUnifiedPlan specified.
 */
- (void)peerConnection:(RTCPeerConnection *)peerConnection
    didStartReceivingOnTransceiver:(RTCRtpTransceiver *)transceiver {
    NSLog(@"peerConnection didStartReceivingOnTransceiver:%@", transceiver);
}

/** Called when a receiver and its track are created. */
- (void)peerConnection:(RTCPeerConnection *)peerConnection
        didAddReceiver:(RTCRtpReceiver *)rtpReceiver
               streams:(NSArray<RTCMediaStream *> *)mediaStreams {
    NSLog(@"peerConnection didAddReceiver");
}

/** Called when the receiver and its track are removed. */
- (void)peerConnection:(RTCPeerConnection *)peerConnection
     didRemoveReceiver:(RTCRtpReceiver *)rtpReceiver {
    NSLog(@"peerConnection didRemoveReceiver");
}

#pragma mark - RTCDataChannelDelegate
/** The data channel state changed. */
- (void)dataChannelDidChangeState:(RTCDataChannel *)dataChannel {
    NSLog(@"dataChannelDidChangeState:%@", dataChannel);
}

/** The data channel successfully received a data buffer. */
- (void)dataChannel:(RTCDataChannel *)dataChannel
didReceiveMessageWithBuffer:(RTCDataBuffer *)buffer {
    if (self.delegate && [self.delegate respondsToSelector:@selector(webRTCClient:didReceiveData:)]) {
        [self.delegate webRTCClient:self didReceiveData:buffer.data];
    }
}


#pragma mark - Lazy
- (RTCPeerConnectionFactory *)factory {
    if (!_factory) {
        RTCInitializeSSL();
        RTCDefaultVideoEncoderFactory *videoEncoderFactory = [[RTCDefaultVideoEncoderFactory alloc] init];
        
        RTCDefaultVideoDecoderFactory *videoDecoderFactory = [[RTCDefaultVideoDecoderFactory alloc] init];
        
        for (RTCVideoCodecInfo *codec in videoEncoderFactory.supportedCodecs) {
            if (codec.parameters) {
                NSString *profile_level_id = codec.parameters[@"profile-level-id"];
                if (profile_level_id && [profile_level_id isEqualToString:@"42e01f"]) {
                    videoEncoderFactory.preferredCodec = codec;
                    break;
                }
            }
        }
        _factory = [[RTCPeerConnectionFactory alloc] initWithEncoderFactory:videoEncoderFactory decoderFactory:videoDecoderFactory];
    }
    return _factory;
}

- (dispatch_queue_t)audioQueue {
    if (!_audioQueue) {
        _audioQueue = dispatch_queue_create("cn.ifour.webrtc", NULL);
    }
    return _audioQueue;
}

- (RTCAudioSession *)rtcAudioSession {
    if (!_rtcAudioSession) {
        _rtcAudioSession = [RTCAudioSession sharedInstance];
    }
    
    return _rtcAudioSession;
}

- (NSDictionary *)mediaConstrains {
    if (!_mediaConstrains) {
        _mediaConstrains = [[NSDictionary alloc] initWithObjectsAndKeys:
                            kRTCMediaConstraintsValueFalse, kRTCMediaConstraintsOfferToReceiveAudio,
                            kRTCMediaConstraintsValueFalse, kRTCMediaConstraintsOfferToReceiveVideo,
                            kRTCMediaConstraintsValueTrue, @"IceRestart",
                            nil];
    }
    return _mediaConstrains;
}

- (NSDictionary *)publishMediaConstrains {
    if (!_publishMediaConstrains) {
        _publishMediaConstrains = [[NSDictionary alloc] initWithObjectsAndKeys:
                                   kRTCMediaConstraintsValueFalse, kRTCMediaConstraintsOfferToReceiveAudio,
                                   kRTCMediaConstraintsValueFalse, kRTCMediaConstraintsOfferToReceiveVideo,
                                   kRTCMediaConstraintsValueTrue, @"IceRestart",
                                   nil];
    }
    return _publishMediaConstrains;
}

- (NSDictionary *)playMediaConstrains {
    if (!_playMediaConstrains) {
        _playMediaConstrains = [[NSDictionary alloc] initWithObjectsAndKeys:
                                kRTCMediaConstraintsValueTrue, kRTCMediaConstraintsOfferToReceiveAudio,
                                kRTCMediaConstraintsValueTrue, kRTCMediaConstraintsOfferToReceiveVideo,
                                kRTCMediaConstraintsValueTrue, @"IceRestart",
                                nil];
    }
    return _playMediaConstrains;
}

- (NSDictionary *)optionalConstraints {
    if (!_optionalConstraints) {
        _optionalConstraints = [[NSDictionary alloc] initWithObjectsAndKeys:
                                kRTCMediaConstraintsValueTrue, @"DtlsSrtpKeyAgreement",
                                nil];
    }
    return _optionalConstraints;
}

@end

三、本地视频画面显示

使用RTCEAGLVideoView本地摄像头视频画面

self.localRenderer = [[RTCEAGLVideoView alloc] initWithFrame:CGRectZero];
//        self.localRenderer.videoContentMode = UIViewContentModeScaleAspectFill;
        [self addSubview:self.localRenderer];

        [self.webRTCClient startCaptureLocalVideo:self.localRenderer];

代码如下

PublishView.h

#import <UIKit/UIKit.h>
#import "WebRTCClient.h"

@interface PublishView : UIView

- (instancetype)initWithFrame:(CGRect)frame webRTCClient:(WebRTCClient *)webRTCClient;

@end

PublishView.m

#import "PublishView.h"

@interface PublishView ()

@property (nonatomic, strong) WebRTCClient *webRTCClient;
@property (nonatomic, strong) RTCEAGLVideoView *localRenderer;

@end

@implementation PublishView

- (instancetype)initWithFrame:(CGRect)frame webRTCClient:(WebRTCClient *)webRTCClient {
    self = [super initWithFrame:frame];
    if (self) {
        self.webRTCClient = webRTCClient;
        
        self.localRenderer = [[RTCEAGLVideoView alloc] initWithFrame:CGRectZero];
//        self.localRenderer.videoContentMode = UIViewContentModeScaleAspectFill;
        [self addSubview:self.localRenderer];

        [self.webRTCClient startCaptureLocalVideo:self.localRenderer];
    }
    return self;
}

- (void)layoutSubviews {
    [super layoutSubviews];
    self.localRenderer.frame = self.bounds;
    NSLog(@"self.localRenderer frame:%@", NSStringFromCGRect(self.localRenderer.frame));
}

@end

四、ossrs推流rtc服务

我这里通过调用rtc/v1/publish/从ossrs获得remotesdp，这里请求的地址如下：https://192.168.10.100:1990/rtc/v1/publish/

使用NSURLSessionDataTask实现http请求，请求代码如下

HttpClient.h

#import <Foundation/Foundation.h>
#import <WebRTC/WebRTC.h>

@interface HttpClient : NSObject<NSURLSessionDelegate>

- (void)changeSDP2Server:(RTCSessionDescription *)sdp
                  urlStr:(NSString *)urlStr
               streamUrl:(NSString *)streamUrl
                 closure:(void (^)(NSDictionary *result))closure;

@end

WebRTCClient.m

#import "HttpClient.h"
#import "IPUtil.h"

@interface HttpClient ()

@property (nonatomic, strong) NSURLSession *session;

@end

@implementation HttpClient

- (instancetype)init
{
    self = [super init];
    if (self) {
        self.session = [NSURLSession sessionWithConfiguration:[NSURLSessionConfiguration defaultSessionConfiguration] delegate:self delegateQueue:[NSOperationQueue mainQueue]];
    }
    return self;
}

- (void)changeSDP2Server:(RTCSessionDescription *)sdp
                  urlStr:(NSString *)urlStr
               streamUrl:(NSString *)streamUrl
                 closure:(void (^)(NSDictionary *result))closure {
    
    //设置URL
    NSURL *urlString = [NSURL URLWithString:urlStr];

    //创建可变请求对象
    NSMutableURLRequest* mutableRequest = [[NSMutableURLRequest alloc] initWithURL:urlString];

    //设置请求类型
    [mutableRequest setHTTPMethod:@"POST"];
    //创建字典，存放要上传的数据
    NSMutableDictionary *dict = [[NSMutableDictionary alloc] init];
    [dict setValue:urlStr forKey:@"api"];
    [dict setValue:[self createTid] forKey:@"tid"];
    [dict setValue:streamUrl forKey:@"streamurl"];
    [dict setValue:sdp.sdp forKey:@"sdp"];
    [dict setValue:[IPUtil localWiFiIPAddress] forKey:@"clientip"];

    //将字典转化NSData类型
    NSData *dictPhoneData = [NSJSONSerialization dataWithJSONObject:dict options:0 error:nil];
    //设置请求体
    [mutableRequest setHTTPBody:dictPhoneData];
    //设置请求头
    [mutableRequest addValue:@"application/json" forHTTPHeaderField:@"Content-Type"];
    [mutableRequest addValue:@"application/json" forHTTPHeaderField:@"Accept"];
    //创建任务
    NSURLSessionDataTask *dataTask = [self.session dataTaskWithRequest:mutableRequest completionHandler:^(NSData * _Nullable data, NSURLResponse * _Nullable response, NSError * _Nullable error) {
        if (error == nil) {
            NSLog(@"请求成功:%@",data);
            NSString *dataString = [[NSString alloc] initWithData:data encoding:kCFStringEncodingUTF8];
            NSLog(@"请求成功 dataString:%@",dataString);
            NSDictionary *result = [NSJSONSerialization JSONObjectWithData:data options:NSJSONReadingAllowFragments error:nil];
            NSLog(@"NSURLSessionDataTask result:%@", result);
            if (closure) {
                closure(result);
            }
        } else {
            NSLog(@"网络请求失败！");
        }
    }];

    //启动任务
    [dataTask resume];
}

- (NSString *)createTid {
    NSDate *date = [[NSDate alloc] init];
    int timeInterval = (int)([date timeIntervalSince1970]);
    int random = (int)(arc4random());
    NSString *str = [NSString stringWithFormat:@"%d*%d", timeInterval, random];
    if (str.length > 7) {
        NSString *tid = [str substringToIndex:7];
        return tid;
    }
    return @"";
}

#pragma mark -session delegate
-(void)URLSession:(NSURLSession *)session didReceiveChallenge:(NSURLAuthenticationChallenge *)challenge completionHandler:(void (^)(NSURLSessionAuthChallengeDisposition, NSURLCredential * _Nullable))completionHandler {

    NSURLSessionAuthChallengeDisposition disposition = NSURLSessionAuthChallengePerformDefaultHandling;
    __block NSURLCredential *credential = nil;

    if ([challenge.protectionSpace.authenticationMethod isEqualToString:NSURLAuthenticationMethodServerTrust]) {
        credential = [NSURLCredential credentialForTrust:challenge.protectionSpace.serverTrust];
        if (credential) {
            disposition = NSURLSessionAuthChallengeUseCredential;
        } else {
            disposition = NSURLSessionAuthChallengePerformDefaultHandling;
        }
    } else {
        disposition = NSURLSessionAuthChallengePerformDefaultHandling;
    }

    if (completionHandler) {
        completionHandler(disposition, credential);
    }
}

@end

这里用到了获取ip的类，代码如下

IPUtil.h

#import <Foundation/Foundation.h>

@interface IPUtil : NSObject

+ (NSString *)localWiFiIPAddress;

@end

IPUtil.m

#import "IPUtil.h"

#include <arpa/inet.h>
#include <netdb.h>

#include <net/if.h>

#include <ifaddrs.h>
#import <dlfcn.h>

#import <SystemConfiguration/SystemConfiguration.h>


@implementation IPUtil

+ (NSString *)localWiFiIPAddress
{
    BOOL success;
    struct ifaddrs * addrs;
    const struct ifaddrs * cursor;
   
    success = getifaddrs(&addrs) == 0;
    if (success) {
        cursor = addrs;
        while (cursor != NULL) {
            // the second test keeps from picking up the loopback address
            if (cursor->ifa_addr->sa_family == AF_INET && (cursor->ifa_flags & IFF_LOOPBACK) == 0)
            {
                NSString *name = [NSString stringWithUTF8String:cursor->ifa_name];
                if ([name isEqualToString:@"en0"])  // Wi-Fi adapter
                    return [NSString stringWithUTF8String:inet_ntoa(((struct sockaddr_in *)cursor->ifa_addr)->sin_addr)];
            }
            cursor = cursor->ifa_next;
        }
        freeifaddrs(addrs);
    }
    return nil;
}

@end

五、调用ossrs推流rtc服务

通过ossrs推流rtc服务，实现本地createOffer之后设置setLocalDescription，再调用rtc/v1/publish/

代码如下

- (void)publishBtnClick {
    __weak typeof(self) weakSelf = self;
    [self.webRTCClient offer:^(RTCSessionDescription *sdp) {
        [weakSelf.webRTCClient changeSDP2Server:sdp urlStr:@"https://192.168.10.100:1990/rtc/v1/publish/" streamUrl:@"webrtc://192.168.10.100:1990/live/livestream" closure:^(BOOL isServerRetSuc) {
            NSLog(@"isServerRetSuc:%@",(isServerRetSuc?@"YES":@"NO"));
        }];
    }];
}

在ViewController上的界面及推流操作

PublishViewController.h

#import <UIKit/UIKit.h>
#import "PublishView.h"

@interface PublishViewController : UIViewController

@end

PublishViewController.m

#import "PublishViewController.h"

@interface PublishViewController ()<WebRTCClientDelegate>

@property (nonatomic, strong) WebRTCClient *webRTCClient;

@property (nonatomic, strong) PublishView *publishView;

@property (nonatomic, strong) UIButton *publishBtn;

@end

@implementation PublishViewController

- (void)viewDidLoad {
    [super viewDidLoad];
    // Do any additional setup after loading the view.
    
    self.view.backgroundColor = [UIColor whiteColor];
    
    self.publishView = [[PublishView alloc] initWithFrame:CGRectZero webRTCClient:self.webRTCClient];
    [self.view addSubview:self.publishView];
    self.publishView.backgroundColor = [UIColor lightGrayColor];
    self.publishView.frame = self.view.bounds;
    
    CGFloat screenWidth = CGRectGetWidth(self.view.bounds);
    CGFloat screenHeight = CGRectGetHeight(self.view.bounds);
    self.publishBtn = [UIButton buttonWithType:UIButtonTypeCustom];
    self.publishBtn.frame = CGRectMake(50, screenHeight - 160, screenWidth - 2*50, 46);
    self.publishBtn.layer.cornerRadius = 4;
    self.publishBtn.backgroundColor = [UIColor grayColor];
    [self.publishBtn setTitle:@"publish" forState:UIControlStateNormal];
    [self.publishBtn addTarget:self action:@selector(publishBtnClick) forControlEvents:UIControlEventTouchUpInside];
    [self.view addSubview:self.publishBtn];
    
    self.webRTCClient.delegate = self;
}

- (void)publishBtnClick {
    __weak typeof(self) weakSelf = self;
    [self.webRTCClient offer:^(RTCSessionDescription *sdp) {
        [weakSelf.webRTCClient changeSDP2Server:sdp urlStr:@"https://192.168.10.100:1990/rtc/v1/publish/" streamUrl:@"webrtc://192.168.10.100:1990/live/livestream" closure:^(BOOL isServerRetSuc) {
            NSLog(@"isServerRetSuc:%@",(isServerRetSuc?@"YES":@"NO"));
        }];
    }];
}

#pragma mark - WebRTCClientDelegate
- (void)webRTCClient:(WebRTCClient *)client didDiscoverLocalCandidate:(RTCIceCandidate *)candidate {
    NSLog(@"webRTCClient didDiscoverLocalCandidate");
}

- (void)webRTCClient:(WebRTCClient *)client didChangeConnectionState:(RTCIceConnectionState)state {
    NSLog(@"webRTCClient didChangeConnectionState");
    /**
     RTCIceConnectionStateNew,
     RTCIceConnectionStateChecking,
     RTCIceConnectionStateConnected,
     RTCIceConnectionStateCompleted,
     RTCIceConnectionStateFailed,
     RTCIceConnectionStateDisconnected,
     RTCIceConnectionStateClosed,
     RTCIceConnectionStateCount,
     */
    UIColor *textColor = [UIColor blackColor];
    BOOL openSpeak = NO;
    switch (state) {
        case RTCIceConnectionStateCompleted:
        case RTCIceConnectionStateConnected:
            textColor = [UIColor greenColor];
            openSpeak = YES;
            break;
            
        case RTCIceConnectionStateDisconnected:
            textColor = [UIColor orangeColor];
            break;
            
        case RTCIceConnectionStateFailed:
        case RTCIceConnectionStateClosed:
            textColor = [UIColor redColor];
            break;
            
        case RTCIceConnectionStateNew:
        case RTCIceConnectionStateChecking:
        case RTCIceConnectionStateCount:
            textColor = [UIColor blackColor];
            break;
            
        default:
            break;
    }
    
    dispatch_async(dispatch_get_main_queue(), ^{
        NSString *text = [NSString stringWithFormat:@"%ld", state];
        [self.publishBtn setTitle:text forState:UIControlStateNormal];
        [self.publishBtn setTitleColor:textColor forState:UIControlStateNormal];
        
        if (openSpeak) {
            [self.webRTCClient speakOn];
        }
//        if textColor == .green {
//            self?.webRTCClient.speakerOn()
//        }
    });
}

- (void)webRTCClient:(WebRTCClient *)client didReceiveData:(NSData *)data {
    NSLog(@"webRTCClient didReceiveData");
}


#pragma mark - Lazy
- (WebRTCClient *)webRTCClient {
    if (!_webRTCClient) {
        _webRTCClient = [[WebRTCClient alloc] initWithPublish:YES];
    }
    return _webRTCClient;
}

@end

当点击按钮开启rtc推流。效果图如下

六、WebRTC视频文件推流

WebRTC还为我们提供了视频文件推流RTCFileVideoCapturer

if ([capturer isKindOfClass:[RTCFileVideoCapturer class]]) {
        RTCFileVideoCapturer *fileVideoCapturer = (RTCFileVideoCapturer *)capturer;
        [fileVideoCapturer startCapturingFromFileNamed:@"beautyPicture.mp4" onError:^(NSError * _Nonnull error) {
            NSLog(@"startCaptureLocalVideo startCapturingFromFileNamed error:%@", error);
        }];
    }

推送的本地视频效果图如下

至此实现了WebRTC音视频通话的iOS端调用ossrs视频通话服务功能。内容较多，描述可能不准确，请见谅。

七、小结

WebRTC音视频通话-实现iOS端调用ossrs视频通话服务。内容较多，描述可能不准确，请见谅。本文地址：https://blog.csdn.net/gloryFlow/article/details/132262724

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