mediasoup | Cutting Edge WebRTC Video Conferencing | Runtime Evironment library
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Cutting Edge WebRTC Video Conferencing
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QUESTION
I was wondering what this means: level=error msg="Handler for GET /system/df returned error: error getting build cache usage: failed to get usage for h23kzrlo9o1ntzqg8djaak2ql: snapshot 6p6vktbybxdisl2r4iovhkxaf not found while I'm building this https://github.com/versatica/mediasoup-demo. I get this error on macos when docker desktop on macos crashes. I have big sur on a 2018 macbook pro, a core i7 and 32 gb of ram. Thank you.
...ANSWER
Answered 2021-Apr-12 at 15:59Assuming part of the build process is using Docker containers, you might want to check to be sure Docker for Mac is allowed to use more than the default 2GB memory. More often than not, that has been the reason Docker crashes on MacOS for me.
Docker -> Preferences -> Resources -> Memory
Increase to 8GB (or more since you have 32GB available).
QUESTION
I'm trying to launch this on AWS Ubuntu.
It works fine under Chrome on localhost. (There was an issue with Firefox, hopefully running remotely with HTTPS will make the problem disappear. But that's unrelated to this question.)
I opened the ports that are specified on readme.MD using the AWS console (inbound TCP to port 3000, inbound UDP to ports 40000-49999, all outgoing traffic is allowed.)
Then adapted config.json to:
...ANSWER
Answered 2021-Mar-15 at 12:24What's up with this?
QUESTION
I'm trying to run this demo on AWS Ubuntu, to check if the bug I reported in firefox is caused by running locally under HTTP; all's working on Chrome, so it may not be. Possibly.
I set the demo up on AWS Ubuntu, and generated a self-signed cert using:
...ANSWER
Answered 2021-Mar-13 at 12:21You are correct that this is a pain in the xxx neck when doing casual demos and bug-reproductions.
I do one of two things when I need to do this.
put the app on https://glitch.com . They give you a durable https://yourprojectname.glitch.me URL to work with. Good for one-offs. I have had bug reports against both Chromium and Firefox accepted with glitch.com examples.
deploy the app to Heroku. They give you a durable https://example-whatever-12345.herokuapp.com URL to use. Good for bigger stuff.
Others use https://netlify.com for purely static stuff. I haven't used it myself.
If you have your own domain and server you can also use https://letsencrypt.org/ to obtain certs. But it takes a mess of server rigging, and might not be best for casual stuff.
QUESTION
ERROR in src/app/components/dashboard/dashboard.component.html:1:1 - error NG8001: 'StackLayout' is ot a known element:
- If 'StackLayout' is an Angular component, then verify that it is part of this module.
- To allow any element add 'NO_ERRORS_SCHEMA' to the '@NgModule.schemas' of this component.
1
dashboard.component.tns.html
...ANSWER
Answered 2020-Oct-13 at 18:48Just realized the app.module.tns.ts doesn't have the Dashboard component declared which resolves the issue.
QUESTION
I'm just building a demo webrtc app with mediasoup npm. I am just trying to record the conversation for recording feature. but there is no docs related to this feature in mediasoup readme.
Is there any way to accomplish this using this npm or any other alternatives that makes building feature rich webrtc application using nodejs ?
...ANSWER
Answered 2020-Aug-05 at 16:16Here is the project with mediasoup recording implemented mediasoup3-record-demo
QUESTION
I'm passing a handful of STUN and TURN servers for my WebRTC application (built on top of mediasoup). When I do this, I get a message in the console telling me: "Using more than two STUN/TURN servers slows down discovery"
I can cut down the servers to 2... but... why does more hurt? Wouldn't I want the most options available to make a connection?
...ANSWER
Answered 2019-Oct-03 at 22:32Because how ICE works is that initially the browser looks at what IP based network interfaces it can find on your machine.
Then it takes the number of network interfaces and combines it with ever single STUN server and every single TURN server the service provides. Next it needs to send request to every single of the these combinations and keep track of the responses. In case of STUN it's just a single response, in case of TURN it is usually multiple round trips. So the more STUN and TURN servers you have the longer it takes to get answers from all of them.
Now if STUN server #1 reports your external IP address as A, with port 1, most likely STUN server #2 is also going to tell you it sees A as your external IP address, with the only difference it reports port 2. In almost all cases all the STUN servers are going to report back the exact same external IP A. But reporting the same external IP address over and over again to the ICE agent on the other end does not increase the chances of establishing the connection at all since its request are all going to hit the same router/NAT/firewall.
With TURN servers one can argue that more TURN servers could help more, as each TURN server should give the browser a different IP address for relaying. But it would be highly unusual if a given browser is able to reach one TURN server and not the another.
In the end all of these servers result in the browser emitting more ICE candidates. And so more ICE candidates need to get send to the other browser or ICE agent. As a result the ICE checking table which tries all possible permutations of local and remote ICE candidates grows bigger and bigger with each ICE candidate. And so it produces a lot more network traffic, while not much increasing the chances of establishing a connection.
QUESTION
I have a webcam chat room application (so it's many-to-many video sharing) using WebRTC and a mediasoup server.
I am having problems with SOME of my users not being able to get an incoming video feeds to work. It's a difficult problem because I can't reproduce it at all, and I can't easily "remote-debug" the problem since most of my users are very non-technical. So far the only thing I can tell for certain is that it seems to be network-related, not browser-related, as I have had bug reports from people using Firefox, Chrome, Safari, and Edge. I'm running my server (mediasoup v2) on port 443 with no firewall on the server box, so that should make the door as wide as possible. I just don't know what the exact problem is yet so I'm feeling around in the dark.
So, I'm trying solutions. I don't think(?) I have a TURN server set up but from what I have read, it seems like adding one certainly can't hurt, and could help with my situation.
I don't fully understand the entire WebRTC protocol or RFC 7118 (this stuff is really complicated!) or exactly what/where/how a TURN server fits into the bigger picture. It would help, right? A lot of Googling has led to no clear answers. Would love some help! Thank you!
...ANSWER
Answered 2019-Aug-20 at 23:01WebRTC 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.
TL;DR, If you need 100% connection rates, you should have a turn server.
I believe AWS has a ready made instance you can spin up, or if you could use this open source coturn server https://github.com/coturn/coturn
On a debugging note... Check your ice candidates type. You should see host and srflx if you only have a STUN server, but if you have a TURN server you will also see relay. You can replicate this issue by discarding the ice candidates that have host and srflx types.
QUESTION
I'm running the popular open-source mediasoup WebRTC server on a standalone box with no firewalls, fresh SSL certs installed and literally nothing else running on the machine (no web server, no database server, nothing). When my video chat app tries to stream video through this server, to a client using Firefox, mediasoup emits an SSL error (2nd line below):
...ANSWER
Answered 2019-Aug-14 at 18:46the mediasoup forum is here: https://mediasoup.discourse.group
BTW, as Philipp says, mediasoup "is not listening in port 8443 or 443". That's the WebSocket server of the mediasoup-demo project. mediasoup does not listen by itself for HTTP or WebSocket at all.
If you don't provide your TLS certificate to mediasoup.createWorker()
then mediasoup will create its own random certificates for DTLS. And it's in DTLS where you see that error.
BTW I've tested in latest Firefox stable (68.0.2) and Nightly (70.0a1 (2019-08-14)) in OSX and I don't get any SSL error in DTLS, so honestly no idea.
PS: If you can provide more info (for example which mediasoup version you are using), please open an issue in mediasoup Discourse Group in the "mediasoup-demo" category.
QUESTION
I installed mediasoup and ran it as well!
But it's working on 3000
port.
I couldn't find listen port in its sources, How i can change this port to443
?
ANSWER
Answered 2017-Oct-22 at 13:05If you need to listen bellow than 1024 port then you need to run as administrator on windows and as a root(sudo
) user on Linux/MAC that is sudo node server.js
and as documentation of https://github.com/versatica/mediasoup-demo, you must update port on config.js
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