I was going through this PubNub WebRTC demo. https://kevingleason.me/SimpleRTC/minivid.html
Which works fine within same network (same browser or different devices across same network). But I tried using it over internet, I am able to connect a call but can not see anything but a black screen. This is the source for same tutorial
https://github.com/pubnub/SimpleRTC
I have gone through many such application, such as AndroidRTC
and I face same problem (black screen after connection over internet). I am unable to figure out why, any help is appreciated.
You need some sort of signaling mechanism (PubNub, Firebase, or your own software [nodejs seems the preferred choice these days]) to get the webRTC API communicating P2P on your local network. To get webRTC to work from one network to another you need a STUN server/service. Google provides free stun servers (stun:stun.l.google.com:19302). To get webRTC to traverse strict firewall settings and complicated networks you need a TURN server/service like xirsys.com.
This article covers it all ...
http://www.html5rocks.com/en/tutorials/webrtc/infrastructure/
Related
I am trying out the WebRTC examples of MuazKhan. It is working perfectly fine when the broadcaster is on AWS/Azure(or any other network) and the receiver is through my phones 4G network. But as soon as I switch to my broadband from 4G, the video stream is unable to connect and I the video player keeps on trying. Therefore I assumed that the problem is of the router NATting and will be resolved if I use a tried-n-tested TURN service such as Xirsys.
Sadly, even after using their TURN servers, I still am blocked with my broadband connection issue as mentioned above. Here are a few queries that I wanted to discuss:-
This issue seems to be due to NATting through my broadbands routers. Shouldn't using the TURN server solve it?
How can I verify if even the TURN servers are getting used and its not just the STUN servers.
Can this issue be due to the Signalling Server?
Do i need to enable some specific protocols in my router to make this work?
What else could be responsible for the issue that I should debug?
After webrtc passes by a signaling server on Internet, how it works when two machines are running in the same network?
The data will be exchanged only on the network or will it still use internet ?
I am asking this because of our internet is not good, it's too slow. but our local network speed is very fast.. So I would like to know if the internet signal will affect the audio and video conversation.
Thanks a lot!
Depending on the network configuration, the devices should connect directly over the local network. Please note that some browsers, such as Safari, may not share with the signalling service local ICE Candidates unless configured to do so (false concern over sharing network info). The devices must share local ICE Candidates, or else they will still stream via the external network or a TURN server, if available.
Is there a simple guide from where I can start creating a stun / turn and signaling server ?
I spend over a week searching for those things and couldn't find any guide where I can say:
okay, I am on the right track now - this is clear.
So far, everything is so abstract without any examples.
This is what I'm trying to achieve: a simple video stream on my local network where I'll have a server with installed usb camera on it, and an application on my iis which will connect to the usb camera and stream it to the clients, and every time when a client opens the application, will see the video stream from the server camera.
Note: since I want to use it on my local network do I really need a stun/turn server, or is there a guide that shows how to avoid it ?
Media streamed over dedicated servers HTTP/HTTPS rarely needs a NAT traversal solution. Instead, just have your web server with camera attached, on the public Internet or behind your NAT with port-forwarding enabled.
There are LOTS of streaming media solutions available as open source, free downloads, or commercially sold. A good list is here:
https://en.wikipedia.org/wiki/List_of_streaming_media_systems
I've created a webrtc p2p app using socket io for signalling.
This work perfectly with one connection pair. When there are more than one connection pair, the video works well but the audio isn't hear on the other side.
I've deployed TURN server, it works as intended.
I'm not able to isolate the issue as in where does this issue lie.
I've been searching on google with no luck. It's been 3 days now
It would be great if anyone would point me to a right direction
I've got WebRTC peer to peer working but when I want to broadcast a single camera to multiple clients obviously peer to peer isn't suitable.
I've found solutions like
http://lynckia.com
and
http://www.medooze.com/products/mcu/webrtc-support.aspx
But the first I can't get setup (and it seems to have cross browser issues)
the second just feels like we're hitting a nail with a nuclear missile.
All I need is a relay, I don't need to decode / recode streams.
I just need
The Broadcaster to connect to the server (peer to peer)
The clients to connect to the server (peer to peer)
The server to relay the stream from the broadcaster to the clients.
Is there any software out there that offers this solution that I've missed? is there an alternative working and scalable alternative?
Thanks
Jitsi Video Bridge works pretty much exactly how you describe.
On your server you can run Janus, to which your broadcaster can provide a stream via RTP.
Have a look at an example configuration file.
After writing a configuration file which defines how the server receives stream from the broadcaster, you should be able to launch janus in the background via a command line interface tool:
$ janus --daemon --config=config_file.conf
Also, see streaming test demo.
Note: I have not tested this thoroughly.
Have a look at this github-repo inspired from muaz khan's WebRTC p2p scalable broadcast. This can work great on LAN. On internet, I am not sure how well it can work as of now though we are improving it on the go.
If you just want to broadcast from a peer to a set of peers, if they don't care about the latency, the best solution is to covert WebRTC to live streaming, without transcoding just muxing:
Peer(Publisher) ---WebRTC--> Server --RTMP/HLS/DASH--> Peers/Players
If this works good for you, SRS is able to covert WebRTC to live streaming.
Because live streaming allows you to use CDN or TCP to deliver the streams, and the latency is about 3~5s, so this solution is only available when Peers/Players never need to communicate to the Peer(Publisher).
If you want all those peers to talk to each other, it's very complex and need a WebRTC SFU cluster to do this, there will be a huge number of streams. For example, if allows 100 peers to talk to each other, there will be 100x100 = 10k streams.
It's too complicated, so I don't think there is good open-source solution right now(at 2022.02).