欢迎使用CSDN-markdown编辑器

之前使用WebRTC 旧版本, 更新到了最新版本. version: 1.1.20621

1.1.19878

(应该区别不大. cocoapods更新太费劲了)

交换消息处依旧会使用伪代码. 别担心我已经注释掉了. 不会出现满目红色错误滴

更新

还是我们的小伙伴cocoapods. 只支持9.0以上咯

source 'https://github.com/CocoaPods/Specs.git'
target 'xxx' do
platform :ios, '9.0'
pod 'GoogleWebRTC'
end

pod install

变化之处

RTCIceServer 创建方式

url

为Turn Server地址. 以数组方式呈现, 多个. 第一篇文章有介绍

RTCIceServer *server = [[RTCIceServer alloc] initWithURLStrings:url
username:@"u1"
credential:@"p1"];

RTCPeerConnection

创建方式

//RTCConfiguration 有很多配置信息但是我都没有使用[��]. 其实是我没去看呢...有时间补充上
RTCConfiguration *config = [[RTCConfiguration alloc] init];
config.iceServers = stunServerArray;
RTCPeerConnection *connection = [factory peerConnectionWithConfiguration:config
constraints:[self  peerConnectionConstraints]
delegate:self];
//self peerConnectionConstraints]此方法和以前一致

代理方法

代理方法名字全部改变 但是和之前对比一样就能看出来就是改了几个字母而已….

RTCRtpSender和RTCMediaStream

RTCMediaStream 虽然未被删除, 但是我觉得RTCRtpSender就是来取代他的.

static NSString *const kARDMediaStreamId = @"ARDAMSa0";

创建AudioSender

- (RTCRtpSender *)createAudioSender {
RTCMediaConstraints *constraints = [self defaultMediaAudioConstraints];
RTCAudioSource *source = [_factory audioSourceWithConstraints:constraints];
RTCAudioTrack *audioTrack = [_factory audioTrackWithSource:source trackId:kARDAudioTrackId];
RTCRtpSender *sender =
[_peerConnection senderWithKind:kRTCMediaStreamTrackKindAudio
streamId:kARDMediaStreamId];
if (audioTrack) {
sender.track = audioTrack;
}
return sender;
}
//MARK : kRTCMediaConstraintsLevelControl : 按照字面意思理解是等级控制. 暂时为true. 具体功能没测试出来
- (RTCMediaConstraints *)defaultMediaAudioConstraints {
RTCMediaConstraints *constraints = [[RTCMediaConstraints alloc] initWithMandatoryConstraints:
@{
kRTCMediaConstraintsLevelControl : kRTCMediaConstraintsValueTrue} optionalConstraints:nil
];
return constraints;
}

创建VideoSender

- (RTCRtpSender *)createVideoSender {
RTCRtpSender *sender = [_peerConnection senderWithKind:kRTCMediaStreamTrackKindVideo streamId:kARDMediaStreamId];
RTCVideoTrack *videoTrack = [self createVideoTrackWithDirecion:AVCaptureDevicePositionFront];
if (videoTrack) {//这个代理和之前相同, 将视频Track 返回到界面上去渲染.
if (self.delegate && [self.delegate respondsToSelector:@selector(dspersonWebRTCdidReceiveLocalVideoTrack:)]) {
[self.delegate dspersonWebRTCdidReceiveLocalVideoTrack:videoTrack];
}
sender.track = videoTrack;
}
return sender;
}
//默认创建为前置摄像头(需求问题. 自定义即可)
- (RTCVideoTrack *)createVideoTrackWithDirecion:(AVCaptureDevicePosition)position {
RTCVideoTrack *localVideoTrack = nil;
#if !TARGET_IPHONE_SIMULATOR && TARGET_OS_IPHONE
//更新方法
localVideoTrack =
[_factory videoTrackWithSource:self.source
trackId:kARDVideoTrackId];
#endif
return localVideoTrack;
}

这个时候已经将音视频Track 添加到

_peerConnection中

了. 和之前

mediaStream

道理相同

删除了RTCPair 这个类

之前创建媒体约束(

RTCMediaConstraints

)都需要此类, 现在更改为字典创建

删除了RTCSessionDescriptionDelegate

删除了这个代理, 我的心一凉(默默的在想这得改多少啊…..)但是还好谷歌还是比较亲民的. 这个代理删除了但是居然不用删除这段代码(还能废物再利用��).

SDP创建都改为

RTCPeerConnection

block形式创建

Offer 和 Answer 的约束

- (RTCMediaConstraints *)defaultOfferConstraints {
NSDictionary *mandatoryConstraints = @{
@"OfferToReceiveAudio" : @"true",
@"OfferToReceiveVideo" : @"true"
};
RTCMediaConstraints* constraints = [[RTCMediaConstraints alloc]
initWithMandatoryConstraints:mandatoryConstraints
optionalConstraints:nil];
return constraints;
}

创建Offer

[_peerConnection offerForConstraints:[self defaultOfferConstraints]
completionHandler:^(RTCSessionDescription * _Nullable sdp, NSError * _Nullable  error) {
[self peerConnection:manager.peerConnection
didCreateSessionDescription:sdp error:error];//这个方法就是以前的代理方法. 废物再利用了.
}];

废物再利用代码

旧版本创建本地SDP代理

- (void)peerConnection:(RTCPeerConnection *)peerConnection
didCreateSessionDescription:(RTCSessionDescription *)sdp
error:(NSError *)error {
//dispatch_async_on_main_queue(^{//一个主线程的block. 自己添加主线程
if (error) {
if (DSpersonKit.isENABLED_DEBUG) {
[MBProgressHUD showErrorMessage:error.description];
}
//DSpersonKitLog(@"\n����������������发送本地SDP 出现错误����������������\n%@", error);
//此处出现错误我会直接发送关闭并且发给对方
//[[DSWebRTCManager shareInstance] closeAndRemoveWithP2P:self receive:false];
return;
}
//DSpersonKitLog(@"发送本地SDP");
//@weakify(self);//weak
[peerConnection setLocalDescription:sdp completionHandler:^(NSError *_Nullable error) {
//@strongify(self); strong
[self peerConnection:self.peerConnection didSetSessionDescriptionWithError:error];
}];
//此处需要将获得的sdp
//[_webSocket send:sdp];
//});

}

旧版本设置远程SDP

- (void)peerConnection:(RTCPeerConnection *)peerConnection
didSetSessionDescriptionWithError:(NSError *)error {
// dispatch_async_on_main_queue(^{
if (error) {
if (DSpersonKit.isENABLED_DEBUG) {
[MBProgressHUD showErrorMessage:error.description];
}
// DSpersonKitLog(@"\n����������������远程SDP协议出现错误����������������\n%@", error);
// [[DSWebRTCManager shareInstance] closeAndRemoveWithP2P:self receive:false];
//同1. 旧版本创建本地SDP代理
return;
}

//如果我们正在回答我们,远程Offer. 我们需要创建的answer, 和一个本地描述()
if (!_isInititor && !_peerConnection.localDescription) {
// DSpersonKitLog(@"接收到远端发来的Offer, 创建本地Answer");
//他应该在SetRemoteDescription之后调用, 否则报错
// @weakify(self);
[_peerConnection answerForConstraints:[self defaultOfferConstraints]
completionHandler:^(RTCSessionDescription * _Nullable sdp, NSError * _Nullable error) {
//@strongify(self);
[self peerConnection:self.peerConnection
didCreateSessionDescription:sdp
error:error];
}];
}
//});

}

拿到offer 设置到本地

从服务器获取到的SDP设置到本地

[self setLocalDescription:sdp completionHandler:^(NSError * _Nullable error) {
[self peerConnection:self.peerConnection didSetSessionDescriptionWithError:error];
}];

RTCICECandidate 变化

更名为RTCIceCandidate

代理名字改变但是不影响和之前方式一致

- (void)peerConnection:(RTCPeerConnection *)peerConnection
didGenerateIceCandidate:(RTCIceCandidate *)candidate  {
if (!_peerConnection) {
//DSpersonKitLog(@"因为出现了错误, 或者收到了Bye 消息. 已经关闭了P2P. 不再发送 ICE Candidate return ")
return;
}
//dispatch_run_in_async(^{//因为要发送很多次记得要异步哦
//[_webSocket send: candidate];
//});
}

///此代理为新增的, 经过测试没有调用. 估计是时机不正确
- (void)peerConnection:(nonnull RTCPeerConnection *)peerConnection didRemoveIceCandidates:(nonnull NSArray<RTCIceCandidate *> *)candidates;

状态名字发生改变. 改为

RTCIceConnectionState

RTCDataChannel 变化

代理名字发生改变(也是一眼就能认出来什么意思)

状态名字更改为

readyState

枚举名字改变. (此处不一一介绍了)

错误问题

SDP错误

Error Domain=org.webrtc.RTCPeerConnection Code=-1 "SessionDescription is NULL."
UserInfo={NSLocalizedDescription=SessionDescription is NULL.}

此错误是因为SDP错误所致, 虽然SDP不为空但是可能格式不争取仔细查看SDP正确格式是

```
v=0//开头
o=- 2225376018456128074 2 IN IP4 127.0.0.1
.......省略
a=ssrc:325890970 label:ARDAMSv0//结尾
```

RTCDataChannel 问题

还是未解决这个问题不清楚原因, 具体请看第一篇文章最后

Google VP8/VP9 编码转为H264

只需要处理SDP即可

#warnging  sdpString 此处使用的是description.sdp. demo中使用的是description.description
+ (RTCSessionDescription *)descriptionForDescription:(RTCSessionDescription *)description
preferredVideoCodec:(NSString *)codec {
NSString *sdpString = description.sdp;
NSString *lineSeparator = @"\n";
NSString *mLineSeparator = @" ";
// Copied from PeerConnectionClient.java.
// TODO(tkchin): Move this to a shared C++ file.
NSMutableArray *lines =
[NSMutableArray arrayWithArray:
[sdpString componentsSeparatedByString:lineSeparator]];
// Find the line starting with "m=video".
NSInteger mLineIndex = -1;
for (NSInteger i = 0; i < lines.count; ++i) {
if ([lines[i] hasPrefix:@"m=video"]) {
mLineIndex = i;
break;
}
}
if (mLineIndex == -1) {
//  RTCLog(@"No m=video line, so can't prefer %@", codec);
return description;
}
// An array with all payload types with name |codec|. The payload types are
// integers in the range 96-127, but they are stored as strings here.
NSMutableArray *codecPayloadTypes = [[NSMutableArray alloc] init];
// a=rtpmap:<payload type> <encoding name>/<clock rate>
// [/<encoding parameters>]
NSString *pattern =
[NSString stringWithFormat:@"^a=rtpmap:(\\d+) %@(/\\d+)+[\r]?$", codec];
NSRegularExpression *regex =
[NSRegularExpression regularExpressionWithPattern:pattern
options:0
error:nil];
for (NSString *line in lines) {
NSTextCheckingResult *codecMatches =
[regex firstMatchInString:line
options:0
range:NSMakeRange(0, line.length)];
if (codecMatches) {
[codecPayloadTypes
addObject:[line substringWithRange:[codecMatches rangeAtIndex:1]]];
}
}
if ([codecPayloadTypes count] == 0) {
// RTCLog(@"No payload types with name %@", codec);
return description;
}
NSArray *origMLineParts =
[lines[mLineIndex] componentsSeparatedByString:mLineSeparator];
// The format of ML should be: m=<media> <port> <proto> <fmt> ...
const int kHeaderLength = 3;
if (origMLineParts.count <= kHeaderLength) {
//RTCLogWarning(@"Wrong SDP media description format: %@", lines[mLineIndex]);
return description;
}
// Split the line into header and payloadTypes.
NSRange headerRange = NSMakeRange(0, kHeaderLength);
NSRange payloadRange =
NSMakeRange(kHeaderLength, origMLineParts.count - kHeaderLength);
NSArray *header = [origMLineParts subarrayWithRange:headerRange];
NSMutableArray *payloadTypes = [NSMutableArray
arrayWithArray:[origMLineParts subarrayWithRange:payloadRange]];
// Reconstruct the line with |codecPayloadTypes| moved to the beginning of the
// payload types.
NSMutableArray *newMLineParts = [NSMutableArray arrayWithCapacity:origMLineParts.count];
[newMLineParts addObjectsFromArray:header];
[newMLineParts addObjectsFromArray:codecPayloadTypes];
[payloadTypes removeObjectsInArray:codecPayloadTypes];
[newMLineParts addObjectsFromArray:payloadTypes];

NSString *newMLine = [newMLineParts componentsJoinedByString:mLineSeparator];
[lines replaceObjectAtIndex:mLineIndex
withObject:newMLine];

NSString *mangledSdpString = [lines componentsJoinedByString:lineSeparator];
return [[RTCSessionDescription alloc] initWithType:description.type
sdp:mangledSdpString];
}

原文地址