天道酬勤,学无止境

peer-connection

为什么我的回合服务器不工作?(Why my turn server doesn't work?)

问题 使用 appr.tc ice 服务器(google turn 服务器)时,我可以在任何情况下进行连接。 但我无法连接到我自己的回合服务器。 我确实通过coturn project配置了我自己的轮流服务器。 我正在使用谷歌的libjingle_peerconnection api 创建一个可以执行video call的Android Application 。 当我运行轮服务器时: <pre> RFC 3489/5389/5766/5780/6062/6156 STUN/TURN Server Version Coturn-4.5.0.5 'dan Eider' 0: Max number of open files/sockets allowed for this process: 4096 0: Due to the open files/sockets limitation, max supported number of TURN Sessions possible is: 2000 (approximately) 0: ==== Show him the instruments, Practical Frost: ==== 0: TLS supported 0: DTLS supported 0: DTLS 1.2 is not supported 0: TURN

2021-12-01 22:44:32    分类:技术分享    webrtc   rfc5766turnserver   coturn   peer-connection

只适用于一对一,其中是多对多,webrtc(Only works on one-to-one of which was to be many-to-many, webrtc)

问题 我正在为这种风格的视频通话开发一个会议风格的应用程序(多对多)。 代码在 GitHub 上可用,但我没有太多的 node.js 经验,因此我决定使用 PHP 创建自己的服务器。 我使用 WebSockets 创建了服务器。 这很简单——它接收消息并将它们转发给所有其他连接的客户端(即,不是发送消息的客户端)。 仅此而已 - 仅此而已; 没有什么。 但我的问题是这种架构不允许客户端与多个人连接,即当客户端尝试与第三人连接时,额外的流失败。 客户端只能一对一连接。 我不知道错误是在 JavaScript 中还是我是否需要改进服务器。 我该怎么做才能让它连接到所有加入的客户? 看我的代码: HTML <script type="text/javascript" src="http://127.0.0.1/scr/js/jquery.js"></script> JavaScript var Server = new WebSocket('ws://127.0.0.1:1805/'), myStream = null, peerConn = null, mediaConstraints = { 'mandatory': { 'OfferToReceiveAudio': true, 'OfferToReceiveVideo': true } }; navigator

2021-11-29 20:32:34    分类:技术分享    javascript   php   webrtc   peer-connection

How i create peer-connection without localStream?

I just want achieve one client send mediaSteam and another received the mediaSteam. So Receiver client needn't add localSteam.and i just code pc.addStream(null).But not work. How i achieve this by WebRtc?

2021-11-26 08:05:17    分类:问答    webrtc   peer-connection

Why my turn server doesn't work?

I can connect in any situation when using appr.tc ice servers (google turn servers). but i can't connect with my own turn server. I did config my own turn server by coturn project. I'm using google's libjingle_peerconnection api to create an Android Application that can perform video call. When i run turn server: <pre> RFC 3489/5389/5766/5780/6062/6156 STUN/TURN Server Version Coturn-4.5.0.5 'dan Eider' 0: Max number of open files/sockets allowed for this process: 4096 0: Due to the open files/sockets limitation, max supported number of TURN Sessions possible is: 2000 (approximately) 0: ====

2021-11-23 17:13:08    分类:问答    webrtc   rfc5766turnserver   coturn   peer-connection

Only works on one-to-one of which was to be many-to-many, webrtc

I am developing a conference style application (many-to-many) for video calls this style. The code is available on GitHub but I do not have much node.js experience, hence I decided to create my own server using PHP. I created the server using WebSockets. It is simple - it receives messages and forwards them to all other connected clients (i.e., not the client that sent the message). Just that - nothing more; nothing less. But my problem is that this architecture does not allow clients to connect with more than one person, i.e., when a client tries to connect with the third person, the

2021-11-20 19:33:35    分类:问答    javascript   php   webrtc   peer-connection

当我调用 peerconnection->Close() 时 WebRtc Native-Crashed(WebRtc Native-Crashed when I call peerconnection->Close())

问题 如何关闭或销毁 PeerConnectionInterface 对象? 当我尝试这样做时它崩溃了。 我有一个这样声明的对象: rtc::scoped_refptr<webrtc::PeerConnectionInterface> _peerConnection; 在我通过工厂创建 PeerConnectionInterface 后,它工作正常。 但是,当会话结束时,我尝试调用 _peerConnection->Close(); 程序崩溃了。 我也尝试调用 _peerConnection.release()->Release(); 也崩溃了。 我在WebRtc源代码的PeerConnection.cc中打印日志,发现这里崩溃了,在Close()函数和~PeerConnection()函数中: webrtc_session_desc_factory_.reset(); //PeerConnection.cc 声明是std::unique_ptr<WebRtcSessionDescriptionFactory> webrtc_session_desc_factory_; 于是我继续登录WebRtcSessionDescriptionFactory.cc,~WebRtcSessionDescriptionFactory()函数。 在此函数中崩溃:FailPendingRequests

2021-10-23 10:34:56    分类:技术分享    webrtc   peer-connection

如何使用 coturn (stun/turn) 服务器在 Web 应用程序中建立对等连接(How to establish peer connection in web app using coturn (stun/turn) server)

问题 我正在构建一个用于凸轮广播的网络应用程序。 我将 Django 用于 Web 应用程序,并将 coturn 作为 (STUN/TURN) 信号服务器。 我的目标是用 WebRTC 做到这一点。 我不知道如何将一个对等点连接到信令服务器以便其他对等点可以访问。 所以我需要知道的是如何建立“PeerConnection”。 在 Web 应用程序中,我拥有所有我需要的(我认为):user.id、共享机密、信令服务器 IP 和端口,......但我不知道如何将它混合在 HTML JS 脚本中以连接 coturn 服务器。 我已经阅读了 coturn 服务器文档并搜索了一些示例,但找不到这部分的示例。 有人可以给我举个例子吗? 回答1 我想你有点困惑, coturn不是信令服务器,它是一个 TURN/STUN 服务器。 通过信令服务器,您可以在对等方获得直接对等连接之前在对等方之间交换 sdp、ice 候选和其他数据,但 coturn 不会这样做。 我无法解释所有的位,但要点是 STUN 用于提供对等方的公共 IP,而 TURN 用作代理点,用于在无法直接访问时从对等方传输和接收数据,并且在大多数情况下,您require 是一个 STUN 服务器。 他们参与您的 WebRTC 应用程序的唯一时间是当您创建PeerConnection对象时,您在配置对象中传递 STUN/TURN

2021-10-14 19:53:26    分类:技术分享    javascript   webrtc   stun   peer-connection   coturn

How to establish peer connection in web app using coturn (stun/turn) server

I'm building a web app for cam broadcasting. I'm using Django for web app and coturn as (STUN/TURN) signalling server. My goal is to do it with WebRTC. I don't know how to connect a peer to the signalling server in order to be reachable by other peer. So what I need to know is how to stablish "PeerConnection". In web application, I have all I need (I think): user.id, shared secret, signalling server IP and port, ... But I don't know how to mix it in HTML JS scripts to connect with coturn server. I've read coturn server docs and searched for some examples but can't find examples for this part

2021-10-11 19:53:00    分类:问答    javascript   webrtc   stun   peer-connection   coturn

WebRtc Native-Crashed when I call peerconnection->Close()

How to close or destruct a PeerConnectionInterface object? It crashed when I'm trying to do so. I have an object declared like this: rtc::scoped_refptr<webrtc::PeerConnectionInterface> _peerConnection; It works fine after I create the PeerConnectionInterface by factory. However, when the session is over and I try to call _peerConnection->Close(); The program crashed. And I also try to call _peerConnection.release()->Release(); Crashed as well. I print logs in PeerConnection.cc which is from the source code of WebRtc, and find that it crashed here, which is in Close() function and

2021-10-03 03:23:11    分类:问答    webrtc   peer-connection

How to set remote description for a WebRTC caller in Chrome without errors?

I hope there is no flaw in the logic. Step 1: caller creates offer Step 2: caller sets localDescription Step 3: caller sends the description to the callee //------------------------------------------------------// Step 4: callee receives the offer sets remote description Step 5: callee creates answer Step 6: callee sets local description Step 7: callee send the description to caller //------------------------------------------------------// Step 8: caller receives the answer and sets remote description And here is the code for the above const socket = io(); const constraints = { audio: true

2021-04-11 05:58:36    分类:问答    javascript   webrtc   SDP   peer-connection   rtcpeerconnection