RTCPeerConnection | tiny browser module that gives normalizes | Runtime Evironment library
kandi X-RAY | RTCPeerConnection Summary
kandi X-RAY | RTCPeerConnection Summary
A tiny browser module that normalizes and simplifies the API for WebRTC peer connections. It gives us a cleaner (cross-browser) way to handle offer/answer and is based on an event emitter. If you're not using browserify or you want AMD support use rtcpeerconnection.bundle.js.
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Top functions reviewed by kandi - BETA
- Creates a new PeerConnection .
- format a value into a string
- The base implementation of _ . clone .
- Format a property .
- Clones an object .
- Try to run clear
- Run setTimeout .
- Inspect a value .
- Drushes messages to the queue .
- Reduce the output to a single string
RTCPeerConnection Key Features
RTCPeerConnection Examples and Code Snippets
Lines of Code : 60
License : No License
Copy/* Minimal RTC Wrapper
*
* @param {Object={}} options They can be:
* {Object|Boolean} dataChannel Does this peer have a DataChannel? If so,
* you can setup some custom config for it
* {MediaStream} stream The MediaStream o {
video : {
basic : true
},
audio : {
basic : true,
webAudio : true
},
rtcPeerConnection : true,
rtcDataChannel : false,
getUserMedia : "prefix- Community Discussions
Trending Discussions on RTCPeerConnection
QUESTION
I was learning a bit about using 'useRef' hooks in React Native functions. While examining some code I found on the Internet...I saw usage including question marks ("?"), like this (as noted in the code):
...ANSWER
Answered 2022-Mar-12 at 08:26it's look you don't need ?. on your example, it's optional chaining, and one more, please add you cunstruction to the ref
QUESTION
I am using pure HTTP polling for establish connections.
PC1 : opens cam, create and send his offer.
PC2 : accepts offer, create answer, send answer.
PC1 and PC2 are connected and PC2 can watch PC1 cam.
Now PC2 opens his cam :I display the local webcam and add video and audio to the connected pc (RTCPeerConnection)
...ANSWER
Answered 2022-Feb-15 at 15:41Since adding the streams changes the SDP you need to renegotiate, i.e. call createOffer, setLocalDescription and (after getting an answer from the remote end) setRemoteDescription. This is called renegotiation (and you will notice the negotiationneeded event firing).
This sample shows upgrading an audio-only call to an audio-video one but the same principle applies.
Note that in your case you are changing who sends the offer so you might run into a problem called 'glare' which is explained nicely in this Mozilla blog post.
QUESTION
I'm having a weird issue with a WebRTC connection between 2 users on different networks.
User A is using a phone hotspot, user B is on my home WiFi.
When user A is the first to join the page, this user sends an offer and some ICE candidates to user B. User B sends an answer and some ICE candidates back to user A, but after about 20 seconds the ICE connection state changes to "failed" (safari) or "disconnected" (chrome). User A and user B are unable to setup a working WebRTC connection.
However, when user B is the first to join the page & creates the offer, and user A sends back an answer, both users are able to connect without issue.
In both cases, user A & user B both have a correct looking localDescription & remoteDescription on their end, but in the second case where the connection fails, the connectionState & ICEConnectionState are "failed" (safari) or "disconnected" (chrome) for both users.
I am using both a TURN and STUN server in my RTCPeerConnection's configuration, and tested the server is working properly using the Trickle ICE WebRTC example
I suspect that somehow user A or user B are unable to see or use all the available ICE candidates for the other user, however I'm unsure how best to troubleshoot the issue. Is there an easy way to see what ICE candidates are available / active for a given RTCPeerConnection object?
Update:
Looking at the chrome://webrtc-internals (as philipp-hancke sugested) shows that
if user B sends the offer:
4 ICE candidate pairs are available (2 host, 1 srflx, 1 relay)
An ICE candidate pair using the external IP for user B, and a TURN server relay for user A is selected. The connection works.
If user A sends the offer:
Again 4 ICE candidate pairs are available (2 host, 1 srflx, 1 relay), but the ICE Candidate pair shows (not connected) and no WebRTC connection is made
Update 2: I've accepted Philipp Hancke's answer since it answered the initial question I asked, even it I haven't solved my problem yet. I've asked a new question here.
...ANSWER
Answered 2022-Jan-31 at 20:47This sounds like a long-standing bug in webrtc. Using TURN servers typically solves the problem which explains why this hasn't been a priority for anyone to work on.
chrome://webrtc-internals is the one-shop-stop location to go for debugging issues like this.
QUESTION
I am using RTCPeerconnections to submit video and audio in a web-RTC-based video-messenger. I am able to detect network disconnects after approximately 7 seconds - yet, this is 7 seconds in which the user is staring at a frozen video and starts to randomly click buttons in the app. I would like to improve the user experience by shortening this timespan - e.g. by informing the user about a network issue if the video freezes for more than 1 second.
Status Quo: I am currently detecting respective situations by listening to the onconnectionstatechange event of the RTCPeerConnection. Yet, the event is only fired approximately 7 seconds after the disconnect. I determined the ~7 seconds by connecting two machines via normal WiFi, using a hardware switch on one of the laptops to switch off the wireless (such switches exist on some older Lenovo models / guarantee an immediate disconnect) and wait for the other machine to detect the event.
Consideration: The root cause being the interruption of the underlying network connection, it would be ideal to detect the changed network status as early as possible (even if its just transport delays). This said, the disturbance faced by the user ultimately stems from the video that instantly freezes when interrupting the network. If there was no way to detect the connection issue earlier, it could be an option to detect the frozen video instead. Is any of these two things possible (ideally event-driven, so that I don't need to poll things every second)?
Here's a very simple code snippet describing my current disconnect detection:
...ANSWER
Answered 2022-Jan-29 at 08:23(ice)connectionstatechange is the right event in general.
If you want more granularity you'll need to poll getStats and looks for stats like framesReceived. But there is no guaranteed frame rate sent from the other side (e.g. in screensharing you go below 1/s).
While the actual ICE statistics like requestsSent seem more useful they happen much less frequently, only once per second and you can loose a packet or it comes late.
In general this is a question of how reliable the detection of the network failure is. If it is too aggressive you end up with a poor UX showing a warning too often. You might not end up that is significantly better than at the cost of introducing complexity that you need to maintain.
QUESTION
In local server, video streaming is running proper on both ends but while network changed video of one peer is not going to display to other peer.
I am using google stun server for connection of peer with another network, but then video streaming of one user is not visible here is my code
I am not able to understand what I am doing wrong here.I am novice in webrtc
...ANSWER
Answered 2022-Jan-12 at 06:27as you move out side of the network the webrtc ICE candidate gathering process may fail because of Routers NAT and firewalls therefore you must have a turn server in your configuration that will relay the traffic if the direct p2p connection establishment fails
QUESTION
I have a task, but I can't seem to get it done. I've created a very simple WebRTC stream on a Raspberry Pi which will function as a videochat-camera. With ionic I made a simple mobile application which can display my WebRTC stream when the phone is connected to the same network. This all works.
So right now I have my own local stream which shows on my app. I now want to be able to broadcast this stream from my phone to a live server, so other people can spectate it.
I know how to create a NodeJS server which deploys my webcam with the 'getUserMedia' function. But I want to 'push' my WebRTC stream to a live server so I can retrieve a public URL for it.
Is there a way to push my local Websocket to a live environment? I'm using a local RTCPeerConnection to create a MediaStream object
...ANSWER
Answered 2021-Dec-10 at 16:54Is there a way to push my local Websocket to a live environment?
It's not straightforward because you need more than vanilla webrtc (which is peer-to-peer). What you want is an SFU. Take a look at mediasoup.
To realize why this is needed think about how the webrtc connection is established in your current app. It's a negotiation between two parties (facilitated by a signaling server). In order to turn this into a multi-cast setup you will need a proxy of sorts that then establishes separate peer-to-peer connections to all senders and receivers.
QUESTION
First, I want to mention that I am very new to WebRTC, so any advice would be very helpful.
Currently I am using aiortc library to build my own WebRTC app.
Here is what I am trying to do.
I have 2 peers, one is web browser, which is written in javascript, and another one is python script, which is working as signaling server and peer at the same time. So If you access to my web page, you will send video frame to server and then the server will make modification of that then send it back.
So I finished testing my app on LAN environment and everything worked as I expected. But once I deployed my app to remote server (Google cloud run) , I encountered Ice connection state failing issue. And gets this log on remote server.
(I think it is due to disconnection between peers, not low memory problem. I tried with 16GB RAM and 4 cpus and still didn't work)
Then, I dig into more information, and found that TURN/STUN server is necessary to build WebRTC app over Internet. So I added google STUN server to my RTCPeerConnection like this. [{'urls': 'stun:stun.l.google.com:19302'}, {'urls': 'stun:stun1.l.google.com:19302'}, {'urls': 'stun:stun2.l.google.com:19302'}] (I added both side on javascript and python because both side is working as peer) Unfortunately, it still didn't work.
Now, I am planning to build my own TURN server, but I am afraid if TURN server wouldn't solve this problem. So I would like to have any advice from you since I am quite stuck within my situation.
p.s I have done SSL encryption.(So GetUserMedia is working fine)
Sdp details(Offer/Answer):
SDP
Offer
...ANSWER
Answered 2021-Dec-10 at 15:13If everything work on local, and this ice server are set, verify that your gcloud server have the correct firewall for webrtc port (not only your signaling port, check the sdp/ice you exchange). also this Webrtc page allow you to check is a stun/turn work on your client
You will not need stun on your python side, as it's a server his ip may be public (unless you don't want to). Stun allow to find your public ip and allow the port to remain open.
On your server you need to open your signaling port (certainly the WS where you exchange the sdp) and the P2P port (candidate lines in the sdp), the media/data will go through this one. For each media (sdp m line) there are usually one used port.
QUESTION
I have this code and I need my table to show the first 10 patients and, after 10 seconds, show the next 10 without touching any button (automatically).
I'm looking for something similar to this: https://embed.plnkr.co/ioh85m5OtPmcvPHyl3Bg/
But with an OData model (as specified on my view and controller).
This is my view:
...ANSWER
Answered 2021-Sep-30 at 06:39- You could bind you items using bindItems, pass skip,top parameters and wrap the whole thing in a setInterval
QUESTION
I'm trying to get a sort of voice chat working using WebRTC and a WebSocket for exchanging offers.
First I create my RTCPeerConection
...ANSWER
Answered 2021-Nov-07 at 18:13So it turns out I completely missed a step. Both Caller and Callee need to exchange their ice candidates. So I added the following code:
QUESTION
I want to make a simple video web chat tool, using webrtc and weosocket.
Use coturn as a stun and turn server.
It can be used normally in the local area network, but it will be iceConnectionState has been checking on the public network.
The following is the implementation code, basically from https://github.com/webrtc/samples/blob/gh-pages/src/content/peerconnection/pc1/js/main.js caller.js:
...ANSWER
Answered 2021-Sep-23 at 18:27The cause of the problem is that the IceCandidate of server.js is not sent to caller.js
Modify server.js
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