ontrack | Continuous delivery monitoring | Continous Integration library
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QUESTION
I'm trying to establish peer connection between two clients via WebRTC and then stream the video from camera through the connection. The problem is, there's no video shown on the remote side, although I can clearly see the remotePc.ontrack
event was fired. Also no error was thrown. I do NOT want to use the icecandidates mechanism (and it should NOT be needed), because the result application will only be used on a local network (the signaling server will only exchange the SDPs for the clients). Why is my example not working?
ANSWER
Answered 2021-Jun-06 at 16:49ICE candidates are needed, as they tell you the local addresses where the clients will connect to each other.
You won't need STUN servers though.
QUESTION
I have succesfully managed to establish a WebRTC connection between Node (server) and a browser. Server gets the video track on onTrack callback inside the RTCPeerConnection. Is there any way I can potentially convert the video track and make it work on ffmpeg so I can output it to rtmp.
Thanks in advance.
...ANSWER
Answered 2021-May-14 at 22:35The way I have done this is to use a socket to the node server, and then use ffmpeg to convert to RTMP:
I spawn FFMPEG
QUESTION
I'm doing a video conference website. The use case is a user are showing camera and everyone already see here camera. It mean the connection is stable. And user want to share screen. After I have screen stream, I add track to peerConnection but remote computer not fire ontrack event.
Here is my code after I got screen stream:
...ANSWER
Answered 2021-May-14 at 19:22You need to renegotiate after addTrack. You can either do so manually by calling createOffer, setLocalDescription and setRemoteDescription or rely on the onnegotiationneeded callback to happen as described in https://blog.mozilla.org/webrtc/perfect-negotiation-in-webrtc/
QUESTION
I'm having problems with the logic to build behind the webRTC multi peer connections handling. Basically I'm trying to make a Room full of people in a videoconference call. I'm using the basic WebSocket library provided by js, and React for the frontend and Java (spring boot) for the backend.
As of my understanding right now this is what I managed to write down (filtered based on what I "think" is relevant)
This is my web socket init method (adding listeners)
...ANSWER
Answered 2021-May-04 at 08:17This feels actually wierd but I fixed it calling the getUserMedia() (which means calling the addTrack before adding the onicecandidate event definition to the peerConnection.
This at least fixed my problem
QUESTION
I am working on webrtc based Audio/Video conferencing app. For that, I have created a frontend in Angular 11 and the Backend node(express server). So I have created this app in using vanilla JS, HTML, CSS and it worked. But when I tried to convert the frontend into Angular app that where the problem comes.
Working of the app goes like this, first I create the room and I will get a roomId
which I send to another person and he uses that roomId
to join the room for a chat.
When another person joins the room then onRoomJoined()
gets executed on my side.
ANSWER
Answered 2021-Apr-24 at 14:11if onRoomJoined function is called as a callback function like onontrack = onRoomJoind, then this
is no longer the Angular component. You can get around this by passing this or setUpConnection function the as a param in onRoomJoined and use/call it inside.
QUESTION
I have a web app that plays video broadcasted using WebRTC. I'm using a COTURN simple implementation (as described here: https://gabrieltanner.org/blog/turn-server). My turnserver.conf looks like this:
...ANSWER
Answered 2021-Apr-04 at 22:56I'm not sure if it was some sort of update on the RTCPeerConnection constructor (because it just stopped working without anyone touching the code), but in case anyone needs this, I solved it by adding
QUESTION
- A webPage uses RTCPeerConnection to create a connection. It uses a STUN and a TURN server in its configuration. It creates an offer and sends the Offer to the remote client only after all the candidates have been gathered
ANSWER
Answered 2021-Mar-22 at 09:22That sounds like packet loss to me. There are a few things that could be causing it.
Can you try lowering the bitrate of what you are sending? See if congestion is causing it.
Can you trying lowering the MTU? If you are forwarding packets from FFMPEG try appending pkt_size to the URL like this
QUESTION
I'm trying to establish a WebRTC connection with two video tracks in the same connection, just after receiving an offer.
The caller can't receive all the video tracks added by the callee when answering the call. However, the caller can start a connection offering two or more video tracks.
That's what the caller (sender) is doing:
...ANSWER
Answered 2021-Feb-18 at 17:33Apparently, this is a case of WebRTC renegotiation.
Adding one more track to the receiver's side will fire its negotiationneeded
event.
According to the docs,
This occurs both during the initial setup of the connection as well as any time a change to the communication environment requires reconfiguring the connection.
So I modified the POC to support a bidirectional negotiation process, as follows:
QUESTION
According to RTCPeerConnection.ontrack documentation, "ontrack" event suppose to fire for each incoming streams. I have a PeerConnection with two video streams, after connection, "ontrack" fires two times (up to here everything is OK). But both times it sends same stream out, so I end up with two identical video, I am sure sender is sending two different streams, dimension and frame rate of them are different and I can clearly see in chrome://webrtc-internals/ that 2 video streams have different frame size/rate.
Here is PeerConnection ontrack code:
...ANSWER
Answered 2021-Feb-17 at 17:09I solved it after few hours of struggling with different browsers and reading documentations several times!
Problem starts at MediaStream.id, it suppose to be unique but element in HTML5 only listens to first track inside each stream. PeerConnection adds new transceivers (as MediaStreamTrack) to same MediaStream, so no matter how many times
ontrack
handler fires, you get exact same MediaStream
objects, but each time you have new unique MediaStreamTrack
inside the RTCTrackEvent
.
Solution is to create new MediaStream
object for each new MediaStreamTrack
inside ontrack
handler.
QUESTION
My video chat app works properly in the same network. It generates and connects IceCandidate using stun as well. But for some reason the peer videos don't play in different networks. I am having trouble debugging the issue.
I haven't used any turn server but I doubt that is the problem as the peers from different networks already joins using stun, only videos don't play
...ANSWER
Answered 2021-Jan-25 at 23:18WebRTC can connect in a few ways, and falls down progressively to lower preference choices as it fails at its first choices.
- naive direct p2p with own ip
- if that fails, use a STUN server to determine what ip (e.g., router) we're behind
- if that fails, true p2p is not possible, use a TURN server instead to relay traffic.
WebRTC tries everything it can do to make a p2p connection, but there are times that it will fail. The turn server acts as a last resort so that the peers can both connect through the turn server. Obviously this is not a p2p connection, so there will be extra latency, and you will have to make sure that your turn server has enough bandwidth to cover all of the connections you expect.
Usually about 20% of connections require a TURN server. It may work fine for you on your network, but try accessing your webRTC service from a different network which has firewall and different network configurations (which will usually require TURN), and you'll see that not all connections are equal when it comes to p2p. so basically this is what is happening to you that different peer are in different network so you are not getting video of peers.
Basically what happens is that since peers have different networks so It become harder to do correct Ice candidate exchange so no media transmission happens that was decided during sdp negotiation, so we need a a public common server(TURN server) which acts as a peer to other peers for smooth ice-candidate exchange so that media transmission can happens (where this TURN server act as a relay for the media transmission between peers) I am sure in your case along with video, audio too is not working for this reasons.
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