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构建跨平台的WebRTC PeerConnection服务端与客户端

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这些组件分别在Ubuntu 18和Ubuntu 20操作系统上编译,但可以在CentOS 7上运行。这表明它们具有良好的跨平台兼容性,这对于WebRTC的部署和使用非常重要。WebRTC(Web Real-Time Communication)是一种支持网页浏览器进行实时语音对话或视频对话的API,它由Google主导开发。WebRTC包括多个API,其中最重要的是RTCPeerConnection API,它是WebRTC的核心,负责管理点对点连接。" 知识点详细说明: 1. WebRTC定义及其重要性:WebRTC是一种开放源代码的项目,允许网络应用或站点,在不需要中间媒介的情况下,建立浏览器之间点对点(Peer-to-Peer)的连接,实现视频流和(或)音频流或者其他任意数据的传输。WebRTC的出现极大的推动了实时通信技术的发展,尤其是在视频会议、在线教育、直播互动等方面的应用。 2. RTCPeerConnection API:RTCPeerConnection是WebRTC技术的核心,主要功能是提供浏览器之间的点对点连接。它使得网页应用可以直接在用户之间传输音频、视频和数据。RTCPeerConnection管理音视频数据的传输和编解码,确保低延迟和高质量的实时通信。 3. peerconnection_server与peerconnection_client概念:在这个上下文中,peerconnection_server和peerconnection_client很可能是指基于WebRTC技术实现的服务器端和客户端组件。这些组件通常涉及媒体交换、会话控制、NAT穿透、编解码器协商等功能,是建立一个完整WebRTC通信系统的关键部分。 4. 跨平台兼容性:根据描述,peerconnection_server和peerconnection_client均被设计为可以在不同的Linux发行版上运行,具体来说,peerconnection_server是在Ubuntu 18上编译的,但可以在CentOS 7上运行。这说明了编译时考虑到了二进制兼容性或者使用了兼容层技术,如使用静态库编译或利用容器技术,以确保应用能在不同环境中运行,而无需重新编译。 5. Ubuntu与CentOS操作系统:Ubuntu和CentOS是两种流行的Linux发行版。Ubuntu以其易用性和强大的社区支持而广受欢迎,而CentOS则因其稳定性和企业级支持而受到许多服务器环境的青睐。一个能够在多个发行版上运行的WebRTC组件,增强了其在不同部署环境下的可用性和灵活性。 6. 编译和运行环境要求:虽然文件没有详细说明编译和运行环境的详细要求,但从中可以推断出,编译这两个组件时,可能使用了兼容的库和工具链,以确保跨平台兼容性。运行时,可能还需要依赖其他的系统库或服务,例如SSL库、网络服务和媒体处理相关的库等。 通过以上详细的知识点说明,我们可以看出,该文件涉及的内容主要围绕WebRTC技术及其应用展开,特别是对于在不同Linux系统中部署WebRTC组件进行了深入探讨。这对于从事实时通信、网络编程以及相关技术的开发者来说,提供了重要的参考价值。

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信令服务器代码如下: using Microsoft.AspNetCore.SignalR; using Newtonsoft.Json; using System.Collections.Concurrent; using webrtc_net_api.Models; namespace webrtc_net_api.Service { public class WebRtcHub : Hub { private static readonly ConcurrentDictionary<string, string> _peerConnections = new(); public override async Task OnConnectedAsync() { Console.WriteLine($"新客户端连接: {Context.ConnectionId}"); await base.OnConnectedAsync(); } public override async Task OnDisconnectedAsync(Exception exception) { Console.WriteLine($"客户端断开: {Context.ConnectionId}"); _peerConnections.TryRemove(Context.ConnectionId, out _); await base.OnDisconnectedAsync(exception); } public async Task JoinRoom(string peerId) { Console.WriteLine($"加入房间: {peerId}"); _peerConnections.TryAdd(Context.ConnectionId, peerId); await Groups.AddToGroupAsync(Context.ConnectionId, peerId); } public async Task SendSignal(SignalingMessage message) { Console.WriteLine($"收到信令: {JsonConvert.SerializeObject(message)}"); // 需引用Newtonsoft.Json Console.WriteLine($"目标PeerId: {message.PeerId}, 当前连接数: {_peerConnections.Count}"); // 查找目标连接(根据目标peerId) var targetConnection = _peerConnections .FirstOrDefault(p => p.Value == message.PeerId).Key; if (!string.IsNullOrEmpty(targetConnection)) { Console.WriteLine($"返回数据:{message.Type}"); await Clients.Client(targetConnection).SendAsync("receive_signal", message); } } } } namespace webrtc_net_api.Models { public class SignalingMessage { public string PeerId { get; set; } public string Type { get; set; } public string Sdp { get; set; } public RTCIceCandidate Candidate { get; set; } } public class RTCIceCandidate { public string Candidate { get; set; } public string SdpMid { get; set; } public int SdpMLineIndex { get; set; } } } WebRtcClient.vue代码如下: <template>
</template> <script> import SimplePeer from 'simple-peer'; import SignalRService from '@/socket'; export default { data() { return { peer: null, signalRService: null, isInitiator: false }; }, async mounted() { this.signalRService = new SignalRService(); await this.signalRService.startConnection('target_peer_id'); // 获取媒体设备 const stream = await navigator.mediaDevices.getUserMedia({ video: true, audio: true }); this.$refs.localVideo.srcObject = stream; // 创建 Peer 连接 this.peer = new SimplePeer({ initiator: this.isInitiator, trickle: true, stream: stream, config: { iceServers: [ { urls: 'stun:stun.l.google.com:19302' } ] } }); // 处理信令 this.peer.on('signal', (data) => { this.signalRService.sendSignal({ type: data.type, sdp: data.sdp, peerId: 'target_peer_id' }); }); // 处理远程流 this.peer.on('stream', (remoteStream) => { this.$refs.remoteVideo.srcObject = remoteStream; }); // 处理 ICE 候选 this.peer.on('iceCandidate', (candidate) => { if (candidate) { this.signalRService.sendSignal({ type: 'ice-candidate', candidate: { candidate: candidate.candidate, sdpMid: candidate.sdpMid, sdpMLineIndex: candidate.sdpMLineIndex }, peerId: 'target_peer_id' }); } }); } }; </script> socket.js代码如下: import { HubConnectionBuilder, LogLevel } from '@microsoft/signalr'; export default class SignalRService { constructor() { this.connection = null; } async startConnection(peerId) { this.connection = new HubConnectionBuilder() .withUrl('http://192.168.10.114:5012/signalr/webrtc', { accessTokenFactory: () => localStorage.getItem('jwt_token') // 可选认证 }) .configureLogging(LogLevel.Information) .withAutomaticReconnect() .build(); this.connection.on('receive_signal', (message) => { this.handleSignal(message); }); try { await this.connection.start(); await this.connection.invoke('JoinRoom', peerId); console.log('SignalR 连接已建立'); } catch (err) { console.error('SignalR 连接失败:', err); setTimeout(() => this.startConnection(peerId), 5000); } } handleSignal(message) { this.peer.signal(message); } sendSignal(message) { this.connection.invoke('SendSignal', message); } } 请提供一个完整的,基于ROS2的WEBRTC视频流推送的功能,代码需完整,并且都在一个文件内,推送地址为:http://192.168.10.114:5012/signalr/webrtc

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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(); 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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}"); } } }

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