peer-connection | 易学教程

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

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问题 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

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

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问题 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

how to close a webrtc datachannel?

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问题 I can not use the following methods. Because i use only datachannel (not use getUserMedia) <script> peerConnection.removeStream(remoteStream) remoteVideo.onerror = null; remoteVideo.pause(); remoteVideo.src = undefined; remoteStream.stop(); remoteStream.onended = null; remoteStream = null; peerConnection.removeStream(localStream) localVideo.onerror = null; localVideo.pause(); localVideo.src = undefined; localStream.stop(); localStream.onended = null; localStream = null; </script> Is there a

Why my turn server doesn't work?

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问题 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

WebRTC video and Audio in IOS not working with 3G /4G but works with WIFI

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问题 I am trying to connect WebRTC video and audio in iOS connected by 3g/4g gives error but works fine with WiFi network. And after the peer connection is found the error appears [carc] CAReportingClient.mm:320:-[CAReportingClient sendMessage:category:type:reporters:]_block_invoke: The operation couldn’t be completed. No valid RTCReporting and the session was started 回答1: Call setup failing on 3G/4G but working in a local network is symptomatic of a STUN failure. See if you are supplying at least

WebRTC video and Audio in IOS not working with 3G /4G but works with WIFI

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I am trying to connect WebRTC video and audio in iOS connected by 3g/4g gives error but works fine with WiFi network. And after the peer connection is found the error appears [carc] CAReportingClient.mm:320:-[CAReportingClient sendMessage:category:type:reporters:]_block_invoke: The operation couldn’t be completed. No valid RTCReporting and the session was started Call setup failing on 3G/4G but working in a local network is symptomatic of a STUN failure. See if you are supplying at least one valid STUN server to the WebRTC engine in your iOS code. Edit: It's also possible that a peer-to-peer

How i create peer-connection without localStream?

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问题 I just want achieve one client send mediaSteam and a nother 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? 回答1: Don't call pc.addStream with null . Instead, just don't call it. When the receiver is also the answerer , this is all there is to it. When the receiver is the offerer , you need a little extra work: You need to specify these RTCOfferOptions to createOffer : { offerToReceiveVideo: true,

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

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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

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

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问题 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