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WebRTC官方Demo代码深度解析与Node.js信令服务器搭建

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通过使用Node.js来建立信令服务器,这个资源能够帮助开发者更好地理解和学习WebRTC中的信令流程和peer-to-peer(P2P)连接的建立。" WebRTC(Web Real-Time Communication)是一项支持网页浏览器进行实时语音对话、视频对话或P2P文件共享的通信技术。在WebRTC的实现过程中,信令服务器扮演着关键角色,它负责协调双方或多方的连接建立过程,交换必要的信息,如网络信息、媒体能力和其他控制信息,以便建立一个直接的P2P连接。 以下是对该项目中各个组件的理解和知识点展开: ### 1. peerConnection(PC1和PC2) WebRTC中的`RTCPeerConnection`对象允许两个浏览器之间直接交换音频、视频和任意数据。它是WebRTC连接的核心,负责维护底层网络连接和传输数据。在实际应用中,每个想要建立连接的设备都需要创建一个`RTCPeerConnection`实例。 - **pc1**:表示第一个端点,通常可以看作是客户端或发起连接请求的一方。 - **pc2**:表示第二个端点,可以看作是服务端或者等待连接请求的一方。 这两个实例之间的通信需要通过信令服务器来协调,它们各自收集本地状态信息并交换给对方,如编解码器支持、IP地址等,以便建立连接。 ### 2. 信令服务器(server) 信令服务器不是WebRTC规范的一部分,但它对于WebRTC通信的建立至关重要。信令服务器的主要任务是传递信号,这些信号包含了建立直接连接所需的所有信息。信号可以是多种类型,如SDP(Session Description Protocol)信令、ICE候选(Interactive Connectivity Establishment),以及其他控制信息。 在本项目中,使用Node.js创建了一个简单的信令服务器。Node.js由于其异步、事件驱动的特性,非常适合用于实时通信的场景,比如信令服务器这种需要处理大量并发连接和消息的场景。 信令服务器通常需要处理以下任务: - **用户连接管理**:管理客户端的连接和断开。 - **会话管理**:为每个通信对创建一个唯一的会话标识。 - **信令交换**:接收、转发和存储来自各个端点的信令数据。 - **ICE候选交换**:收集并交换NAT穿透所需的信息。 - **协商和建立会话**:在PC1和PC2之间交换SDP信息,使得双方能达成一致的通信参数,并建立连接。 ### 3. JavaScript JavaScript作为Web开发中最广泛使用的脚本语言之一,是实现WebRTC应用的主要语言。在这个项目中,JavaScript被用来实现以下功能: - **创建和操作RTCPeerConnection对象**:通过JavaScript实例化RTCPeerConnection,管理连接的生命周期。 - **处理WebRTC事件**:监听WebRTC事件,如连接状态改变、候选信息收集等。 - **发送和接收信号**:JavaScript用于处理从信令服务器接收信号的逻辑,以及将信号发送到信令服务器的过程。 ### 实际应用场景 在实际开发中,WebRTC技术可以应用于多种场景,如: - **视频会议系统**:实现实时视频和音频通信。 - **在线游戏**:实现玩家之间的实时互动和数据同步。 - **文件共享应用**:通过P2P直接传输文件,降低服务器负载。 - **在线教育平台**:教师和学生之间进行实时互动教学。 在使用WebRTC时,需要注意安全性问题,如传输数据的加密,以及使用HTTPS等安全协议来保护信令数据的安全。 ### 结论 webRTC-test项目通过整理官方demo代码,分离出三个组件,并用Node.js建立了一个信令服务器。这个项目不仅为开发者提供了WebRTC技术实现的参考,还展示了如何在实际应用中构建信令机制,是学习和研究WebRTC的重要资源。通过深入分析和实践该项目,开发者可以加深对WebRTC工作原理的理解,进而在自己的应用中实现可靠和高效的实时通信功能。

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现在在unity中,已导入webrtc(版本3.0.0-pre.8)和mirror(版本96.0.1),已有如下代码实现了在本地测试使用webrtc进行音视频通话的功能,如何结合mirror实现局域网内两台设备间的音视频通话功能.请在现有代码上进行修改调整.using System; using System.Collections; using System.Collections.Generic; using System.Linq; using UnityEngine; using UnityEngine.Experimental.Rendering; using UnityEngine.UI; namespace Unity.WebRTC.Samples { class VideoReceiveSample2 : MonoBehaviour { #pragma warning disable 0649 [SerializeField] private RawImage sourceImage; [SerializeField] private AudioSource sourceAudio; [SerializeField] private RawImage receiveImage; [SerializeField] private AudioSource receiveAudio; #pragma warning restore 0649 private RTCPeerConnection _pc1, _pc2; private List<RTCRtpSender> pc1Senders; private VideoStreamTrack videoStreamTrack; private AudioStreamTrack audioStreamTrack; private MediaStream receiveAudioStream, receiveVideoStream; private DelegateOnIceConnectionChange pc1OnIceConnectionChange; private DelegateOnIceConnectionChange pc2OnIceConnectionChange; private DelegateOnIceCandidate pc1OnIceCandidate; private DelegateOnIceCandidate pc2OnIceCandidate; private DelegateOnTrack pc2Ontrack; private DelegateOnNegotiationNeeded pc1OnNegotiationNeeded; private WebCamTexture webCamTexture; private Texture2D webcamCopyTexture; private Coroutine coroutineConvertFrame; private void OnDestroy() { if (webCamTexture != null) { webCamTexture.Stop(); webCamTexture = null; } } private void Start() { pc1Senders = new List<RTCRtpSender>(); pc1OnIceConnectionChange = state => { OnIceConnectionChange(_pc1, state); }; pc2OnIceConnectionChange = state => { OnIceConnectionChange(_pc2, state); }; pc1OnIceCandidate = candidate => { OnIceCandidate(_pc1, candidate); }; pc2OnIceCandidate = candidate => { OnIceCandidate(_pc2, candidate); }; pc2Ontrack = e => { if (e.Track is VideoStreamTrack video) { video.OnVideoReceived += tex => { receiveImage.texture = tex; }; } if (e.Track is AudioStreamTrack audioTrack) { receiveAudio.SetTrack(audioTrack); receiveAudio.loop = true; receiveAudio.Play(); } }; pc1OnNegotiationNeeded = () => { StartCoroutine(PeerNegotiationNeeded(_pc1)); }; StartCoroutine(WebRTC.Update()); } private static RTCConfiguration GetSelectedSdpSemantics() { RTCConfiguration config = default; config.iceServers = new[] { new RTCIceServer { urls = new[] { "stun:stun.l.google.com:19302" } } }; return config; } private void OnIceConnectionChange(RTCPeerConnection pc, RTCIceConnectionState state) { switch (state) { case RTCIceConnectionState.New: Debug.Log($"{GetName(pc)} IceConnectionState: New"); break; case RTCIceConnectionState.Checking: Debug.Log($"{GetName(pc)} IceConnectionState: Checking"); break; case RTCIceConnectionState.Closed: Debug.Log($"{GetName(pc)} IceConnectionState: Closed"); break; case RTCIceConnectionState.Completed: Debug.Log($"{GetName(pc)} IceConnectionState: Completed"); break; case RTCIceConnectionState.Connected: Debug.Log($"{GetName(pc)} IceConnectionState: Connected"); break; case RTCIceConnectionState.Disconnected: Debug.Log($"{GetName(pc)} IceConnectionState: Disconnected"); break; case RTCIceConnectionState.Failed: Debug.Log($"{GetName(pc)} IceConnectionState: Failed"); break; case RTCIceConnectionState.Max: Debug.Log($"{GetName(pc)} IceConnectionState: Max"); break; default: throw new ArgumentOutOfRangeException(nameof(state), state, null); } } IEnumerator PeerNegotiationNeeded(RTCPeerConnection pc) { Debug.Log($"{GetName(pc)} createOffer start"); var op = pc.CreateOffer(); yield return op; if (!op.IsError) { if (pc.SignalingState != RTCSignalingState.Stable) { Debug.LogError($"{GetName(pc)} signaling state is not stable."); yield break; } yield return StartCoroutine(OnCreateOfferSuccess(pc, op.Desc)); } else { pp报错提示(op.Error); } } public void AddTracks() { var videoSender = _pc1.AddTrack(videoStreamTrack); pc1Senders.Add(videoSender); pc1Senders.Add(_pc1.AddTrack(audioStreamTrack)); if (WebRTCSettings.UseVideoCodec != null) { var codecs = new[] { WebRTCSettings.UseVideoCodec }; var transceiver = _pc1.GetTransceivers().First(t => t.Sender == videoSender); transceiver.SetCodecPreferences(codecs); } } public void RemoveTracks() { var transceivers = _pc1.GetTransceivers(); foreach (var transceiver in transceivers) { if (transceiver.Sender != null) { transceiver.Stop(); _pc1.RemoveTrack(transceiver.Sender); } } pc1Senders.Clear(); } public void Call() { Debug.Log("GetSelectedSdpSemantics"); var configuration = GetSelectedSdpSemantics(); _pc1 = new RTCPeerConnection(ref configuration); Debug.Log("Created local peer connection object pc1"); _pc1.OnIceCandidate = pc1OnIceCandidate; _pc1.OnIceConnectionChange = pc1OnIceConnectionChange; _pc1.OnNegotiationNeeded = pc1OnNegotiationNeeded; _pc2 = new RTCPeerConnection(ref configuration); Debug.Log("Created remote peer connection object pc2"); _pc2.OnIceCandidate = pc2OnIceCandidate; _pc2.OnIceConnectionChange = pc2OnIceConnectionChange; _pc2.OnTrack = pc2Ontrack; CaptureAudioStart(); StartCoroutine(CaptureVideoStart()); } private void CaptureAudioStart() { if(Microphone.devices.Length == 0) { Debug.Log("Microphone device not found"); return; } var deviceName = Microphone.devices[0]; Microphone.GetDeviceCaps(deviceName, out int minFreq, out int maxFreq); var micClip = Microphone.Start(deviceName, true, 1, 48000); while (!(Microphone.GetPosition(deviceName) > 0)) { } sourceAudio.clip = micClip; sourceAudio.loop = true; sourceAudio.Play(); audioStreamTrack = new AudioStreamTrack(sourceAudio); } private IEnumerator CaptureVideoStart() { if (WebCamTexture.devices.Length == 0) { Debug.LogFormat("WebCam device not found"); yield break; } yield return Application.RequestUserAuthorization(UserAuthorization.WebCam); if (!Application.HasUserAuthorization(UserAuthorization.WebCam)) { Debug.LogFormat("authorization for using the device is denied"); yield break; } int width = WebRTCSettings.StreamSize.x; int height = WebRTCSettings.StreamSize.y; const int fps = 30; WebCamDevice userCameraDevice = WebCamTexture.devices[0]; webCamTexture = new WebCamTexture(userCameraDevice.name, height, height, fps); webCamTexture.Play(); yield return new WaitUntil(() => webCamTexture.didUpdateThisFrame); var supportedFormat = WebRTC.GetSupportedGraphicsFormat(SystemInfo.graphicsDeviceType); if (webCamTexture.graphicsFormat != supportedFormat) { webcamCopyTexture = new Texture2D(width, height, supportedFormat, TextureCreationFlags.None); videoStreamTrack = new VideoStreamTrack(webcamCopyTexture); coroutineConvertFrame = StartCoroutine(ConvertFrame()); } else { videoStreamTrack = new VideoStreamTrack(webCamTexture); } sourceImage.texture = webCamTexture; } IEnumerator ConvertFrame() { while (true) { yield return new WaitForEndOfFrame(); Graphics.ConvertTexture(webCamTexture, webcamCopyTexture); } } public void HangUp() { if (webCamTexture != null) { webCamTexture.Stop(); webCamTexture = null; } if (coroutineConvertFrame != null) { StopCoroutine(coroutineConvertFrame); coroutineConvertFrame = null; } receiveAudioStream?.Dispose(); receiveAudioStream = null; receiveVideoStream?.Dispose(); receiveVideoStream = null; videoStreamTrack?.Dispose(); videoStreamTrack = null; audioStreamTrack?.Dispose(); audioStreamTrack = null; _pc1?.Dispose(); _pc2?.Dispose(); _pc1 = null; _pc2 = null; sourceImage.texture = null; sourceAudio.Stop(); sourceAudio.clip = null; receiveImage.texture = null; receiveAudio.Stop(); receiveAudio.clip = null; } private void OnIceCandidate(RTCPeerConnection pc, RTCIceCandidate candidate) { GetOtherPc(pc).AddIceCandidate(candidate); Debug.Log($"{GetName(pc)} ICE candidate:\n {candidate.Candidate}"); } private string GetName(RTCPeerConnection pc) { return (pc == _pc1) ? "pc1" : "pc2"; } private RTCPeerConnection GetOtherPc(RTCPeerConnection pc) { return (pc == _pc1) ? _pc2 : _pc1; } private IEnumerator OnCreateOfferSuccess(RTCPeerConnection pc, RTCSessionDescription desc) { Debug.Log($"Offer from {GetName(pc)}\n{desc.sdp}"); Debug.Log($"{GetName(pc)} setLocalDescription start"); var op = pc.SetLocalDescription(ref desc); yield return op; if (!op.IsError) { pp本地创建成功(pc); } else { var error = op.Error; pp本地创建失败(ref error); } var otherPc = GetOtherPc(pc); Debug.Log($"{GetName(otherPc)} setRemoteDescription start"); var op2 = otherPc.SetRemoteDescription(ref desc); yield return op2; if (!op2.IsError) { pp远程设置成功(otherPc); } else { var error = op2.Error; pp本地创建失败(ref error); } Debug.Log($"{GetName(otherPc)} createAnswer start"); var op3 = otherPc.CreateAnswer(); yield return op3; if (!op3.IsError) { yield return pp对等连接成功(otherPc, op3.Desc); } else { pp报错提示(op3.Error); } } private void pp本地创建成功(RTCPeerConnection pc) { Debug.Log($"{GetName(pc)} SetLocalDescription complete"); } static void pp本地创建失败(ref RTCError error) { Debug.LogError($"Error Detail Type: {error.message}"); } private void pp远程设置成功(RTCPeerConnection pc) { Debug.Log($"{GetName(pc)} 远程描述信息已经成功设置"); } IEnumerator pp对等连接成功(RTCPeerConnection pc, RTCSessionDescription desc) { Debug.Log($"Answer from {GetName(pc)}:\n{desc.sdp}"); Debug.Log($"{GetName(pc)} setLocalDescription start"); var op = pc.SetLocalDescription(ref desc); yield return op; if (!op.IsError) { pp本地创建成功(pc); } else { var error = op.Error; pp本地创建失败(ref error); } var otherPc = GetOtherPc(pc); Debug.Log($"{GetName(otherPc)} setRemoteDescription start"); var op2 = otherPc.SetRemoteDescription(ref desc); yield return op2; if (!op2.IsError) { pp远程设置成功(otherPc); } else { var error = op2.Error; pp本地创建失败(ref error); } } private static void pp报错提示(RTCError error) { Debug.LogError($"错误信息: {error.message}"); } } }