本文详细介绍了从旧版WebRTC更新到version 1.1.20621过程中遇到的变化，包括RTCIceServer创建方式、RTCPeerConnection的代理方法变化、RTCPair和RTCSessionDescriptionDelegate的删除，以及RTCDataChannel和RTCICECandidate的调整。同时，文章提及了SDP错误和RTCDataChannel问题，并提供了Google VP8/VP9编码转为H264的方法。
之前使用WebRTC 旧版本, 更新到了最新版本. version:
1.1.206211.1.19878(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
RTCPeerConnectionblock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];
}
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