android WebRtc 视频通话(P2P)
概述
WebRTC名称源自网页实时通信(Web Real-Time Communication)的缩写,是一个支持网页浏览器进行实时语音对话或视频对话的技术,是谷歌2010年以6820万美元收购Global IP Solutions公司而获得的一项技术。Google于2011年6月3日开源的即时通讯项目,旨在使其成为客户端视频通话的标准。其实在Google将WebRTC开源之前,微软和苹果各自的通讯产品已占用很大市场份额(如Skype),Google也是为了快速扩大市场,所以将他给开源。在行业内得到了广泛的支持和应用,成为下一代视频通话的标准。更多介绍可以去官网上看。
WebRTC被誉为是web长期开源开发的一个新启元,是近年来Web开发的最重要创新。WebRTC允许Web开发者在其web应用中添加视频聊天或者点对点数据传输,不需要复杂的代码或者昂贵的配置。目前支持Chrome、Firefox和Opera,后续会支持更多的浏览器,它有能力达到数十亿的设备。
目前,WebRTC的应用已经不局限在浏览器与浏览器之间,通过官方提供的SDK,我们可以很容易的实现本地应用间的音视频传输。在Android平台上,我们也非常容易的集成WebRTC框架,用非常简洁的代码就能实现强大、可靠的音视频传输功能。
实现
说明
本文代码修改自meshenger-android
True P2P Voice- and video phone calls without the need for accounts or access to the Internet. There is no discovery mechanism, no meshing and no servers. Just scan each others QR-Code that will contain the contacts IP address. This works in many local networks such as community mesh networks, company networks or at home.
翻译如下:
真正的 P2P 语音和视频电话呼叫,无需帐户或访问互联网。 没有发现机制,没有网格化,也没有服务器。 只需相互扫描包含联系人 IP 地址的二维码即可。 这适用于许多本地网络,例如社区网状网络、公司网络或家庭网络。
版本: meshenger-android-3.0.3 最后的JAVA版本, 最新版本已经改用kotlin
基本情况:
功能满足一个局域网P2P的视频通话功能
- 二维码身份生成-----正常
- 二维码扫码添加通讯录-----正常
- 连接发起通话------异常
主要问题在于无法创建两端之间的Socket连接, 与WebRTC本身无关 - 通讯加解密 ----- 异常
- 视频通话 ---- 只显示对方视频不显示自身
感谢作者
- 增加WebRtc支持
build.gradle
dependencies {implementation 'org.webrtc:google-webrtc:1.0.32006'
}
- 关键代码及应用
创建 PeerConnectionFactory
private void initRTC() {log("initRTC");eglCtxRemote = EglBase.create().getEglBaseContext();eglCtxLocal = EglBase.create().getEglBaseContext();//创建 PeerConnectionFactory//这种方法存在兼容性问题, 在一些平台上, 会导致后续流程不能正常执行./*PeerConnectionFactory.initialize(PeerConnectionFactory.InitializationOptions.builder(context).createInitializationOptions());factory = PeerConnectionFactory.builder().createPeerConnectionFactory();*/constraints = new MediaConstraints();constraints.optional.add(new MediaConstraints.KeyValuePair("offerToReceiveAudio", "true"));constraints.optional.add(new MediaConstraints.KeyValuePair("offerToReceiveVideo", "true"));constraints.optional.add(new MediaConstraints.KeyValuePair("DtlsSrtpKeyAgreement", "true"));//initVideoTrack();PeerConnectionFactory.initialize(PeerConnectionFactory.InitializationOptions.builder(context).createInitializationOptions());PeerConnectionFactory.Options options = new PeerConnectionFactory.Options();//https://yuriyshea.com/archives/androidwebrtc%E8%B8%A9%E5%9D%91%E6%8C%87%E5%8D%97DefaultVideoEncoderFactory enVdf = new DefaultVideoEncoderFactory(eglCtxRemote, true, true);VideoDecoderFactory deVdf = new DefaultVideoDecoderFactory(eglCtxRemote);//创建 PeerConnectionFactoryfactory = PeerConnectionFactory.builder().setOptions(options).setVideoEncoderFactory(enVdf).setVideoDecoderFactory(deVdf).createPeerConnectionFactory();log("initRTC done");}
创建PeerConnection 发起呼叫
connection = factory.createPeerConnection(Collections.emptyList(), new DefaultObserver() {@Overridepublic void onIceGatheringChange(PeerConnection.IceGatheringState iceGatheringState) {super.onIceGatheringChange(iceGatheringState);log("Outgoing.onIceGatheringChange " + iceGatheringState.name());if (iceGatheringState == PeerConnection.IceGatheringState.COMPLETE) {log("Outgoing.connect call from remote address: " + contact.getAddresses());reportStateChange(CallState.CONNECTING);//发送信息给接收方,告知发起通话.getPublisher().sendCall(contact.getAddresses(), connection.getLocalDescription().description);}@Overridepublic void onIceConnectionChange(PeerConnection.IceConnectionState iceConnectionState) {log("Outgoing.onIceConnectionChange " + iceConnectionState.name());super.onIceConnectionChange(iceConnectionState);if (iceConnectionState == PeerConnection.IceConnectionState.DISCONNECTED) {reportStateChange(CallState.DISCONNECTED);}else if (iceConnectionState == PeerConnection.IceConnectionState.CLOSED) {hangUp("Outgoing.onIceConnectionChange CLOSED");}}@Overridepublic void onAddStream(MediaStream mediaStream) {log("Outgoing.onAddStream");super.onAddStream(mediaStream);handleMediaStream(mediaStream);}@Overridepublic void onDataChannel(DataChannel dataChannel) {log("Outgoing.onDataChannel");super.onDataChannel(dataChannel);RTCCall.this.dataChannel = dataChannel;dataChannel.registerObserver(RTCCall.this);}});//初始化音视频通道connection.addStream(createStream());//创建数据通道, 可用于收发消息.this.dataChannel = connection.createDataChannel("data", new DataChannel.Init());this.dataChannel.registerObserver(this);log("Outgoing.createOffer");connection.createOffer(new DefaultSdpObserver() {@Overridepublic void onCreateSuccess(SessionDescription sessionDescription) {log("Outgoing.onCreateSuccess");super.onCreateSuccess(sessionDescription);connection.setLocalDescription(new DefaultSdpObserver(), sessionDescription);}}, constraints);
呼叫方收到后同样初始化, 并在点击接听后创建PeerConnection
public void accept(OnStateChangeListener listener) {log("accept");this.listener = listener;new Thread(() -> {connection = factory.createPeerConnection(this.iceServers, new DefaultObserver() {@Overridepublic void onIceGatheringChange(PeerConnection.IceGatheringState iceGatheringState) {super.onIceGatheringChange(iceGatheringState);if (iceGatheringState == PeerConnection.IceGatheringState.COMPLETE) {log("Incoming.onIceGatheringChange");//通知已接听getPublisher().sendAnswer(contact.getAddresses(),connection.getLocalDescription().description);reportStateChange(CallState.CONNECTED);}}@Overridepublic void onIceConnectionChange(PeerConnection.IceConnectionState iceConnectionState) {log("Incoming.onIceConnectionChange " + iceConnectionState.name());super.onIceConnectionChange(iceConnectionState);if (iceConnectionState == PeerConnection.IceConnectionState.DISCONNECTED) {reportStateChange(CallState.DISCONNECTED);}else if (iceConnectionState == PeerConnection.IceConnectionState.CLOSED) {hangUp("Incoming.onIceConnectionChange CLOSED");}}@Overridepublic void onAddStream(MediaStream mediaStream) {log("Incoming.onAddStream");super.onAddStream(mediaStream);handleMediaStream(mediaStream);}@Overridepublic void onDataChannel(DataChannel dataChannel) {super.onDataChannel(dataChannel);RTCCall.this.dataChannel = dataChannel;dataChannel.registerObserver(RTCCall.this);}});connection.addStream(createStream());//this.dataChannel = connection.createDataChannel("data", new DataChannel.Init());log("Incoming.setting remote description");//设置会话, 创建响应应答connection.setRemoteDescription(new DefaultSdpObserver() {@Overridepublic void onSetSuccess() {super.onSetSuccess();log("creating answer...");connection.createAnswer(new DefaultSdpObserver() {@Overridepublic void onCreateSuccess(SessionDescription sessionDescription) {log("Incoming.onCreateSuccess");super.onCreateSuccess(sessionDescription);connection.setLocalDescription(new DefaultSdpObserver(), sessionDescription);}@Overridepublic void onCreateFailure(String s) {super.onCreateFailure(s);log("Incoming.onCreateFailure: " + s);}}, constraints);}}, new SessionDescription(SessionDescription.Type.OFFER, offer));}).start();}
呼叫方处理应答
private void handleAnswer(String remoteDesc) {log("handleAnswer");connection.setRemoteDescription(new DefaultSdpObserver() {@Overridepublic void onSetSuccess() {super.onSetSuccess();}@Overridepublic void onSetFailure(String s) {super.onSetFailure(s);}}, new SessionDescription(SessionDescription.Type.ANSWER, remoteDesc));}
启动摄像头
public void setVideoEnabled(boolean enabled) {log("setVideoEnabled enabled=" + enabled);this.videoEnabled = enabled;try {if (enabled) {this.capturer.startCapture(640, 480, 30);} else {this.capturer.stopCapture();}JSONObject object = new JSONObject();object.put(StateChangeMessage, enabled ? CameraEnabledMessage : CameraDisabledMessage);log("setVideoEnabled: " + object);dataChannel.send(new DataChannel.Buffer(ByteBuffer.wrap(object.toString().getBytes()), false));} catch (JSONException e) {e.printStackTrace();} catch (InterruptedException e) {e.printStackTrace();}}
挂断
public void hangUp(String res) {if(state == CallState.ENDED){log("hangUp Ignored already ENDED:" + res);return;}log("hangUp:" + res);reportStateChange(CallState.ENDED);closePeerConnection();//通知挂断.new Thread(() -> {getPublisher().sendHangup(contact.getAddresses());}).start();}
//Fatal signal 11 (SIGSEGV), code 1 (SEGV_MAPERR), fault addr 0x0 in tid 13128 (signaling_threa), pid 13067 (d.d.meshenger)//type=1400 audit(0.0:28585): avc: granted { nlmsg_readpriv } for scontext=u:r:untrusted_app_29:s0:c5,c257,c512,c768 tcontext=u:r:untrusted_app_29:s0:c5,c257,c512,c768 tclass=netlink_route_socket app=d.d.meshenger//pid: 13067, tid: 13128, name: signaling_threa >>> d.d.meshenger <<<private void closePeerConnection() {log("closePeerConnection");if(localRenderer != null){localRenderer.release();localRenderer = null;}if(remoteRenderer != null){remoteRenderer.release();remoteRenderer = null;}if(capturer != null){try {capturer.stopCapture();capturer.dispose();capturer = null;} catch (InterruptedException e) {e.printStackTrace();}}if (connection != null) {try {PeerConnection conn = connection;connection = null;conn.close();} catch (Exception e) {e.printStackTrace();}//connection = null;log("closePeerConnection done");}}
完整呼叫流程LOG如下
[呼叫方]
## 开始呼出
2022-12-01 14:50:09.016 23145-23717 RTCCall D RTCCall created
2022-12-01 14:50:09.016 23145-23717 RTCCall D initRTC
2022-12-01 14:50:09.041 23145-23717 RTCCall D initRTC done
2022-12-01 14:50:09.042 23145-23728 RTCCall D createPeerConnection
2022-12-01 14:50:09.052 23145-23728 RTCCall D createStream
2022-12-01 14:50:09.058 23145-23728 RTCCall D createCapturer
2022-12-01 14:50:09.139 23145-23728 RTCCall D Outgoing.createOffer
2022-12-01 14:50:09.144 23145-23725 RTCCall D Outgoing.onCreateSuccess
2022-12-01 14:50:09.212 23145-23725 RTCCall D Outgoing.onIceGatheringChange GATHERING
2022-12-01 14:50:09.327 23145-23725 RTCCall D Outgoing.onIceGatheringChange COMPLETE
2022-12-01 14:50:09.327 23145-23725 RTCCall D transferring offer...
2022-12-01 14:50:09.328 23145-23735 RTCCall D Outgoing.connect call from remote address: 192.168.7.239
2022-12-01 14:50:09.328 23145-23735 RTCCall D reportStateChange CONNECTING## 对方接听
2022-12-01 14:50:15.331 23145-23168 RTCCall D reportStateChange CONNECTED
2022-12-01 14:50:15.331 23145-23168 RTCCall D handleAnswer
2022-12-01 14:50:15.409 23145-23725 RTCCall D Outgoing.onIceConnectionChange CHECKING
2022-12-01 14:50:15.415 23145-23725 RTCCall D Outgoing.onAddStream
2022-12-01 14:50:15.415 23145-23725 RTCCall D handleMediaStream ava=false
2022-12-01 14:50:15.416 23145-23725 RTCCall D handleAnswer.onSetSuccess
2022-12-01 14:50:15.512 23145-23725 RTCCall D Outgoing.onIceConnectionChange CONNECTED
2022-12-01 14:50:15.528 23145-23725 RTCCall D onStateChange## 对方开启摄像头, 并推送
2022-12-01 14:50:24.939 23145-23725 RTCCall D onMessage: {"StateChange":"CameraEnabled"}
2022-12-01 14:50:30.932 23145-23725 RTCCall D Outgoing.onIceConnectionChange COMPLETED## 开启摄像头并推送
2022-12-01 14:50:39.122 23145-23145 RTCCall D setVideoEnabled enabled=true
2022-12-01 14:50:39.122 23145-23145 RTCCall D setVideoEnabled: {"StateChange":"CameraEnabled"}
2022-12-01 14:50:39.124 23145-23725 RTCCall D onBufferedAmountChange l=31## 挂断
2022-12-01 14:51:03.363 23145-23145 RTCCall D hangUp:UI.callDecline click
2022-12-01 14:51:03.363 23145-23145 RTCCall D reportStateChange ENDED
2022-12-01 14:51:03.363 23145-23145 RTCCall D closePeerConnection
2022-12-01 14:51:03.370 23145-23145 RTCCall D closePeerConnection state=CONNECTED
2022-12-01 14:51:03.372 23145-23725 RTCCall D Outgoing.onIceConnectionChange CLOSED
2022-12-01 14:51:03.372 23145-23725 RTCCall D hangUp Ignored already ENDED:Outgoing.onIceConnectionChange CLOSED
2022-12-01 14:51:03.547 23145-23725 RTCCall D onStateChange
2022-12-01 14:51:03.547 23145-23725 RTCCall D onStateChange
2022-12-01 14:51:03.580 23145-23145 RTCCall D closePeerConnection done
2022-12-01 14:51:04.059 23145-23145 RTCCall D releaseCamera
2022-12-01 14:51:04.149 23145-23168 RTCCall D hangUp Ignored already ENDED:publisher msg
[被叫方]
## 来电并响铃
2022-12-01 14:22:30.319 8916-8943 RTCCall D initRTC
2022-12-01 14:22:30.363 8916-8943 RTCCall D initRTC done
2022-12-01 14:22:30.364 8916-8943 RTCCall D reportStateChange RINGING## 接听
2022-12-01 14:22:35.625 8916-8916 RTCCall D setRenderer
2022-12-01 14:22:35.625 8916-8916 RTCCall D accept
2022-12-01 14:22:35.661 8916-14310 RTCCall D createStream
2022-12-01 14:22:35.662 8916-14310 RTCCall D createCapturer
2022-12-01 14:22:35.677 8916-14310 RTCCall D Incoming.setting remote description
2022-12-01 14:22:35.768 8916-14256 RTCCall D Incoming.onAddStream
2022-12-01 14:22:35.768 8916-14256 RTCCall D handleMediaStream ava=false
2022-12-01 14:22:35.769 8916-14256 RTCCall D creating answer...
2022-12-01 14:22:35.773 8916-14256 RTCCall D Incoming.onCreateSuccess
2022-12-01 14:22:35.873 8916-14256 RTCCall D Incoming.onIceConnectionChange CHECKING
2022-12-01 14:22:36.035 8916-14256 RTCCall D Incoming.onIceGatheringChange## 连接已建立
2022-12-01 14:22:36.048 8916-14256 RTCCall D reportStateChange CONNECTED
2022-12-01 14:22:36.256 8916-14256 RTCCall D Incoming.onIceConnectionChange CONNECTED
2022-12-01 14:22:36.278 8916-14256 RTCCall D onStateChange## 开启摄像头并推送
2022-12-01 14:22:45.635 8916-8916 RTCCall D setVideoEnabled enabled=true
2022-12-01 14:22:45.636 8916-8916 RTCCall D setVideoEnabled: {"StateChange":"CameraEnabled"}
2022-12-01 14:22:45.638 8916-14256 RTCCall D onBufferedAmountChange l=31## 对方开启摄像头
2022-12-01 14:23:02.916 8916-14256 RTCCall D onMessage: {"StateChange":"CameraEnabled"}## 对方已挂断
2022-12-01 14:23:24.318 8916-14256 RTCCall D Incoming.onIceConnectionChange DISCONNECTED
2022-12-01 14:23:24.318 8916-14256 RTCCall D reportStateChange DISCONNECTED
2022-12-01 14:23:24.320 8916-14256 RTCCall D onStateChange
2022-12-01 14:23:24.320 8916-14256 RTCCall D onStateChange
2022-12-01 14:23:24.430 8916-8943 RTCCall D hangUp:publisher msg
2022-12-01 14:23:24.430 8916-8943 RTCCall D reportStateChange ENDED
2022-12-01 14:23:24.431 8916-8943 RTCCall D closePeerConnection
2022-12-01 14:23:24.445 8916-8943 RTCCall D closePeerConnection state=CONNECTED
2022-12-01 14:23:24.449 8916-14256 RTCCall D Incoming.onIceConnectionChange CLOSED
2022-12-01 14:23:24.449 8916-14256 RTCCall D hangUp Ignored already ENDED:Incoming.onIceConnectionChange CLOSED
2022-12-01 14:23:24.705 8916-8943 RTCCall D closePeerConnection done
2022-12-01 14:23:24.924 8916-8916 RTCCall D releaseCamera
一些问题
首先是源码中Socket连接的问题
Utils.java 与IPV6有关, 这里改成了IPV4.
并在后续的网络相关部分改为使用IP地址.
public static List<InetSocketAddress> getAddressPermutations(String contact_mac, int port) {byte[] contact_mac_bytes = Utils.macAddressToBytes(contact_mac);ArrayList<InetSocketAddress> addrs = new ArrayList<InetSocketAddress>();try {List<NetworkInterface> all = Collections.list(NetworkInterface.getNetworkInterfaces());for (NetworkInterface nif : all) {if (nif.isLoopback()) {continue;}for (InterfaceAddress ia : nif.getInterfaceAddresses()) {InetAddress addr = ia.getAddress();if (addr.isLoopbackAddress()) {continue;}android.util.Log.d("Utils", "getAddressPermutations " + addr.getHostName() + "," + addr.getHostAddress() + ",");if (addr instanceof Inet4Address) {addrs.add(new InetSocketAddress(addr, port));/*Inet6Address addr6 = (Inet6Address) addr;byte[] extracted_mac = getEUI64MAC(addr6);if (extracted_mac != null && Arrays.equals(extracted_mac, nif.getHardwareAddress())) {// We found the interface MAC address in the IPv6 assigned to that interface in the EUI-64 scheme.// Now assume that the contact has an address with the same scheme.InetAddress new_addr = createEUI64Address(addr6, contact_mac_bytes);if (new_addr != null) {addrs.add(new InetSocketAddress(new_addr, port));}}*/}}}} catch (Exception e) {e.printStackTrace();}
增加本地摄像头预览显示
方法是传入本地的SurfaceViewRenderer并修改getVideoTrack
private boolean enablePreview = true;private VideoTrack getVideoTrack() {this.capturer = createCapturer();if(!enablePreview) {return factory.createVideoTrack("video1", factory.createVideoSource(false));}else {VideoSource videoSource = factory.createVideoSource(false);//EglBase.Context eglBaseContext = EglBase.create().getEglBaseContext();SurfaceTextureHelper surfaceTextureHelper = SurfaceTextureHelper.create("CaptureThread", eglCtxLocal);capturer.initialize(surfaceTextureHelper, App.getApp(), videoSource.getCapturerObserver());//capturer.startCapture(480, 640, 30);VideoTrack track = factory.createVideoTrack("video1", videoSource);track.addSink(localRenderer);return track;}}
初始化方式问题导致无正常回调
constraints = new MediaConstraints();constraints.optional.add(new MediaConstraints.KeyValuePair("offerToReceiveAudio", "true"));constraints.optional.add(new MediaConstraints.KeyValuePair("offerToReceiveVideo", "true"));constraints.optional.add(new MediaConstraints.KeyValuePair("DtlsSrtpKeyAgreement", "true"));
//方式1:PeerConnectionFactory.initialize(PeerConnectionFactory.InitializationOptions.builder(context).createInitializationOptions());factory = PeerConnectionFactory.builder().createPeerConnectionFactory();//方式2: PeerConnectionFactory.initialize(PeerConnectionFactory.InitializationOptions.builder(context).createInitializationOptions());PeerConnectionFactory.Options options = new PeerConnectionFactory.Options();DefaultVideoEncoderFactory enVdf = new DefaultVideoEncoderFactory(eglCtxRemote, true, true);VideoDecoderFactory deVdf = new DefaultVideoDecoderFactory(eglCtxRemote);factory = PeerConnectionFactory.builder().setOptions(options).setVideoEncoderFactory(enVdf).setVideoDecoderFactory(deVdf).createPeerConnectionFactory();
方式1 会导致呼出时的错误如下
connection.createOffer(new DefaultSdpObserver() {@Overridepublic void onCreateSuccess(SessionDescription sessionDescription) {log("Outgoing.onCreateSuccess");super.onCreateSuccess(sessionDescription);connection.setLocalDescription(new DefaultSdpObserver(){@Overridepublic void onSetFailure(String s) {super.onSetFailure(s);//出错时的LOG://onSetFailure s=//Failed to set local offer sdp: //Failed to set local video description recv parameters for m-section with mid='video'.log("Outgoing.onSetFailure s=" + s);}}, sessionDescription);}}, constraints);
多次调用PeerConnection.close()会导致崩溃
2022-12-01 14:31:57.847 22887-22887 DEBUG pid-22887 A Build fingerprint: 'google/blueline/blueline:12/SP1A.210812.016.C1/8029091:user/release-keys'
2022-12-01 14:31:57.847 22887-22887 DEBUG pid-22887 A Revision: 'MP1.0'
2022-12-01 14:31:57.847 22887-22887 DEBUG pid-22887 A ABI: 'arm64'
2022-12-01 14:31:57.847 22887-22887 DEBUG pid-22887 A Timestamp: 2022-12-01 14:31:57.642912791+0800
2022-12-01 14:31:57.847 22887-22887 DEBUG pid-22887 A Process uptime: 0s
2022-12-01 14:31:57.847 22887-22887 DEBUG pid-22887 A Cmdline: d.d.meshenger
2022-12-01 14:31:57.847 22887-22887 DEBUG pid-22887 A pid: 22778, tid: 22851, name: signaling_threa >>> d.d.meshenger <<<
2022-12-01 14:31:57.847 22887-22887 DEBUG pid-22887 A uid: 10261
2022-12-01 14:31:57.847 22887-22887 DEBUG pid-22887 A signal 11 (SIGSEGV), code 1 (SEGV_MAPERR), fault addr 0x0
2022-12-01 14:31:57.847 22887-22887 DEBUG pid-22887 A Cause: null pointer dereference
2022-12-01 14:31:57.847 22887-22887 DEBUG pid-22887 A x0 0000000000000000 x1 0000000000000005 x2 0000000000000000 x3 0000007c6f2be89d
2022-12-01 14:31:57.847 22887-22887 DEBUG pid-22887 A x4 0000007c584fd818 x5 0000007f12e7a555 x6 73656d2f642f644c x7 632f7265676e6568
2022-12-01 14:31:57.847 22887-22887 DEBUG pid-22887 A x8 0000007be2f5a000 x9 81248e9b13f44bd4 x10 0000000000430000 x11 0000000000000001
2022-12-01 14:31:57.847 22887-22887 DEBUG pid-22887 A x12 0000000000000004 x13 0000000000000004 x14 7ffbffff00000000 x15 0000000000000000
2022-12-01 14:31:57.847 22887-22887 DEBUG pid-22887 A x16 0000007c584fcf00 x17 0000007f23435688 x18 0000007be1bc6000 x19 0000007d918a0910
2022-12-01 14:31:57.847 22887-22887 DEBUG pid-22887 A x20 0000000000000005 x21 0000007c584fe000 x22 0000007c584fd960 x23 0000007c584fe000
2022-12-01 14:31:57.847 22887-22887 DEBUG pid-22887 A x24 0000007be2f16ab8 x25 00000000ffffffff x26 0000007c584fdff8 x27 00000000000fc000
2022-12-01 14:31:57.847 22887-22887 DEBUG pid-22887 A x28 0000007c58405000 x29 0000007c584fda10
2022-12-01 14:31:57.847 22887-22887 DEBUG pid-22887 A lr 0000007be34be6e0 sp 0000007c584fd950 pc 0000007be34be6f4 pst 0000000060000000
资源
源码及资料下载
Android WebRTC 的一些资料
参考
WebRTC
googlesource webrtc / src
git clone https://webrtc.googlesource.com/src (需要连接外网)
meshenger-android
WebRTC-Android 探索 - 创建音视频通话程序的基本姿势
WebRTC实现Android传屏demo
Android WebRTC踩坑指南
Google WebRtc Android 使用详解(包括客户端和服务端代码)
owt-client-android
android webrtc学习 一(源码下载和编译)
编译webrtc android源码
libwebrtc.aar
android WebRtc 视频通话(P2P)相关推荐
- WebRTC之P2P
WebRTC之P2P StoneLiu999 2020-11-19 11:35:39 802 已收藏 4 分类专栏: WebRTC 文章标签: webrtc p2p turn nat stun 版权声 ...
- Android WebRTC语音视频通话demo
Android WebRTC简介 https://blog.csdn.net/Charon_Chui/article/details/80510945?utm_term=%E6%89%8B%E6%9C ...
- 流媒体协议之WebRTC实现p2p视频通话(二)
阿里P7移动互联网架构师进阶视频(每日更新中)免费学习请点击:https://space.bilibili.com/474380680 简介 目的 帮助自己了解webrtc 实现端对端通信 # 使用流 ...
- WebRTC 实现P2P音视频通话——原生IOS端使用WebRTC实现一对一音视频通话
IOS端使用WebRTC实现一对一音视频通话 前言 环境 一.环境配置 搭建项目,配置权限,通过CocoaPods安装第三方库 二.音视频通话的实现 音视频通话实现主要分为两部分,信令客户端以及web ...
- WebRTC 实现P2P音视频通话——搭建信令服务器
WebRTC 实现P2P音视频通话--搭建信令服务器 文章目录 WebRTC 实现P2P音视频通话--搭建信令服务器 前言 一.安装NodeJS,npm 二.服务器端实现 1.引入库 2.代码实现 3 ...
- Android WebRTC 入门教程(二) -- 模拟p2p本地视频传输
Android WebRTC 入门教程(一) – 使用相机 Android WebRTC 入门教程(二) – 模拟p2p本地视频传输 源码工程: https://github.com/LillteZh ...
- Android WebRTC+SRS/ZLM视频通话(5):Android使用WebRTC从SRS/ZLMediaKit拉流
Android WebRTC+SRS/ZLM视频通话(5):Android使用WebRTC从SRS/ZLMediaKit拉流 来自奔三人员的焦虑日志 接着上一章内容,继续来记录Android是如何使用 ...
- WebRTC 实现P2P音视频通话——实现一对一音视频通话
WebRTC 实现P2P音视频通话 WebRTC 实现P2P音视频通话--搭建信令服务器 WebRTC 实现P2P音视频通话--搭建stun/trun P2P穿透和转发服务器 WebRTC 实现P2P ...
- WebRTC 实现P2P音视频通话——搭建stun/turn P2P穿透和转发服务器
WebRTC 实现P2P音视频通话 WebRTC 实现P2P音视频通话--搭建信令服务器 WebRTC 实现P2P音视频通话--搭建stun/turn P2P穿透和转发服务器 文章目录 WebRTC ...
最新文章
- ES6新特性:Javascript中的Reflect对象
- 安卓怎么下载python-教你在安卓手机上安装python程序
- MVVM架构~knockoutjs系列之验证成功提示显示
- CSS3中的2D和3D转换知识介绍
- nbiot开发需要掌握什么_什么是前端工程师?前端工程师需要掌握什么技能?
- 系统架构的演变 -----自 罗文浩
- java 做项目踩坑,web项目踩坑过程
- 《Python Cookbook 3rd》笔记(1.7):字典排序
- Codeplus2017 12月赛——可做题1
- jquery插件图片浏览
- vux安装中遇到的坑(转)
- java 改文件名的例子
- 结构方程模型_结构方程模型(Structural Equation Model, SEM) 三下
- 2019年最新淘宝联盟淘宝客升高佣规则
- matlab interp插值函数
- 软件中的易用性设计及测试(一)
- 阿里云被攻击封多久?
- 比较简单的初学者模仿毕业设计项目springboot人力资源管理系统.rar(项目源码+数据库文件)
- 程序人生 - 西瓜霜能吃下去吗?
- ap计算机知识点总结,AP微积分重要知识点总结(全)
热门文章
- HOperatorSet.GrabImageAsync(out ho_Image, hv_AcqHandle, -1);出现异常
- 提取Insight-MVT_Annotation_Train 数据集标签xml文件中的信息
- 【Unity2d】Sprite Renderer精灵渲染器
- 2021年茶艺师(中级)报名考试及茶艺师(中级)操作证考试
- 常用软件(Android)
- 百度搜索引擎爬行蜘蛛IP大全
- java是否过于笨重?
- php soap version,PHP的SOAP接口提示Wrong Version
- r语言的MASS包干什么的_R语言常用包汇总
- 【计算机网络】TCP为什么需要4次挥手