Questions tagged [rtcpeerconnection]

140 questions
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Determine why Webrtc failed

I am using webRTC for sharing video between two computers. The problem I have is that sometimes it does not work, but usually, it does work. I think my companies firewall is breaking the connection, but I have no way of knowing why the video just…
Samuel Thompson
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WebRTC crash during CreateModularPeerConnectionFactory

I am trying to create a basic chat application using WebRTC peer connection on OSX Catalina and I do the following: signaling_thread = rtc::Thread::Create(); signaling_thread->SetName("signaling_thread",…
Anil
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WebRTC Peer Connection Error: Failed to set remote answer sdp: Called in wrong state: kStable

I am having an issue with a piece of javascript code that is supposed to connect to users to a webRTC video call using a Viagenie server and Firebase. When one of the users presses the call button they receive the following error: Uncaught (in…
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WebRTC peer iceconnectionstatechange and connectionstatechange dont fire "closed" event from Chrome 80

WebRTC iceconnectionstatechange and connectionstatechange dont fire "closed" event in Chrome 80. Does anyone have any workaround to detect the connectionstatechange when peer is closed? It doesnot fire even in plan-b
Anthony
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Not getting WebRTC status chanegd to RTCICEConnectionConnected

below are my steps: Initialise webRTC socket connected onRenegotiationNeeded addLocalMediaStrem peerConnectionCreateOffer didCreateSessionDescription setLocalDescriptionWith SDP send SDP to socket RTCICEGatheringGathering get local…
Krutika Sonawala
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How to put metadata on MediaStream when adding to RTCPeerConnection?

I am developing a multiuser video chat application, where each user has a different role. While listening for RTCPeerConnection.onAddStream event, I am not able to figure out the source of the mediaStream. Is there an easy way to add metadata at…
rethna2
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How to Remove Blank Frames in RecordRTC

I am stuck on some point. Please help me to figure it out. When any peer connection (In Video Conferencing) is disconnected and I reconnect them, then a blank frame is added in Recording, I am using RecordRTC and testing it in Chrome 74.0 I think it…
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how to install JRTPLIB in ubuntu?

I try to install JRTPLIB on the ubuntu machine, but I don't know how to do that can anyone help me, please. thank you
siva
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Calling createAnswer doesn't change `signallingState`

Calling createAnswer doesn't change signalingState, it still in have-remote-offer. What may be the possible reason for this. offerSdp = { "sdp": offerSdp, "type": "offer" }; pc.setRemoteDescription(new…
Shashidhara
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chrome 72.0.3626.81 and RTCPeerConnection.setLocalDescription

With chrome prior version 71.0.3578.98 and firefox 64.0.2 or 65.0 the same code works fine. It fails creating the offer, so the localDescription is never updated. Any clue of the root cause or how to progress to find a solution? Debugging with…
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WebRTC connection between peers connected on localhost 127.0.0.1

I have slightly different usecase here. I want to do a webrtc connection between two peers running on the same host. My hardware has no connectivity (no WiFi, Ethernet), only interface I have is a lo (127.0.0.1). I am doing signaling using direct…
pragnesh
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how to close mediastream after multiple remote peers attached?

I am creating multiple webrtc peer connections and creating a single mediastream using if (mediaStream == undefined) { navigator.mediaDevices.getUserMedia({ audio: true, video: true …
Croftie
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How to get webrtc jitter buffer in firefox?

I am using webrtc peerconnection.getstats to get various parameters to check the call quality in Firefox. I want to get the jitter buffer parameter, but its not exposed in the getstats response (but it is available in chrome). Also the buffer is…
Anthony
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WebRTC problem with renegotiation (after addTrack, removeTrack)

I have an application where every call starts with audio, but user have opportunity switching video chat (there is a button). And here is problem, after the first switch all work fine but then after switch to audio and again to video remote stream…
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WebRTC: Tracks are Empty in ontrack Event - Frontend and Backend Implementation

I am working on a WebRTC application where I am trying to establish a video streaming connection between a client (React frontend) and a server (Node.js backend) using WebRTC. However, I am facing an issue where the ontrack event on the server sides…