Open Yamakasi opened 10 years ago
Hi Matt, Thanks so much for the kind remarks! Can you give me a little more info on what you are trying to accomplish?
Hi,
Thanks for your response and email! What I actually do visiting the webpage with 2 PC's and the users don't see eachother.
So I'm figuring out how an ID is set and how I can set this manually, or using a GET variable from the URL.
Cheers,
Matt
Hi Matt,
Ok, so I am assuming you followed step-by-step tutorial on this page:
http://bearcoda.com/index.php/7-webrtc-setup-a-one-to-one-video-call
If so you have the following steps completed:
Let me know, Joe Lopez
Hi,
Yes I did all stpes, but I thought you build a node.js signaling server in... or am I wrong ?
2014-03-21 16:05 GMT+01:00 BearCoda notifications@github.com:
Hi Matt,
Ok, so I am assuming you followed step-by-step tutorial on this page:
http://bearcoda.com/index.php/7-webrtc-setup-a-one-to-one-video-call
If so you have the following steps completed:
1.
Setup a signaling server to trade WebRTC watch id's. If you don't have one then you setup and used the sample I setup in the signalServer folder. 2.
You have a setup TURN and STUN server to connect to and have opened the webRTCSample.html file and configured line 62 to connect to your servers. 3.
If you used my signal server sample then also in that html file you uncommented lines 14 and 15. 4.
You tested the demo on the latest Chrome, Firefox, or Opera stable builds.
Let me know, Joe Lopez
Reply to this email directly or view it on GitHubhttps://github.com/bearcoda/webRTCSample/issues/2#issuecomment-38284578 .
Hi Matt, Oh also for step 3 you also have to uncomment line 57 hehe.
I included a front-end sample that you can plugin in case you don't already have a signaling server you can plugin yourself. However, you still have to setup the back-end Node.js with the files I included in the signalServer folder. I did include a readMe txt file to guide you through the setup. Apart from that keep in mind you will also need to have access to a TURN and STUN server which is separate from the signaling server.
Hi,
Thanks! yeah indeed, I need to checkup on the code about line 57 but I think I uncommented it.
Why do you still need a TURN/STUN server when node.js is the signaling server already ?
Thanks again mate!
Cheers,
Matt
2014-03-21 17:49 GMT+01:00 BearCoda notifications@github.com:
Hi Matt, Oh also for step 3 you also have to uncomment line 57 hehe.
I included a front-end sample that you can plugin in case you don't already have a signaling server you can plugin yourself. However, you still have to setup the back-end Node.js with the files I included in the signalServer folder. I did include a readMe txt file to guide you through the setup. Apart from that keep in mind you will also need to have access to a TURN and STUN server which is separate from the signaling server.
Reply to this email directly or view it on GitHubhttps://github.com/bearcoda/webRTCSample/issues/2#issuecomment-38296824 .
Hi Matt, Yea I was confused by all this when I started so I definitely understand. The ICE servers (TURN and STUN) are the ones that handle the actual connection process to be able to stream the live video. When you initially make a request for a live stream connection the browser API provides ids for both the caller and receiver. The way these ids are exchanged between the viewers is not handled by the ICE servers, that part is left up to the developer. Developers setup signaling servers (well that's what they call it) and these can be setup however you want using the platform of your choice as long as the ids go to their proper destinations.
The demo I built handles the exchange of id's using the signaling server sample I included but it also by default gives you the option to include your own signaling server. However, you will still have to provide and configure your own ICE servers. Now there are a couple of locations I saw on the web that have a public open server you can have access to. I did a quick search and this turned up:
http://www.voip-info.org/wiki/view/STUN
I haven't tested them so I don't know if they work and they only provide STUN so you will also need a TURN to fallback on for NAT traversal. A better choice would be to setup your own via this project:
https://code.google.com/p/rfc5766-turn-server/
I actually tested the demo with this project and it worked great. A third option as well is to sign up with someone that host these servers. The only one that I really know of is Xirsys and from what I heard they provide a pretty good service with the ICE and signaling servers. I haven't used them myself but if you are interested the link to their page is below.
Hope this helps, Joe Lopez
Hi,
You define IceServers as one server, but we need stun and turn, so it should be comma seperated.
I see people running stun/turn on the same IP but with seperate ports. This is possible also ?
Like here on slide 15/16 http://www.slideshare.net/amiteshawa/web-rtc-media-stra
Thanks again!
Hi Matt,
I suggest you check out this link below.
http://www.html5rocks.com/en/tutorials/webrtc/infrastructure/
It explains the infrastructure pretty well. Hope this helps, Joe Lopez
Hi Joe,
Yes thank you for that link, I have seen it before and will read up further.
Was it possible in your signaling script to connect using a set hash ? So if the has matched the right people see eachtoher ? Or should we add the create room part there ?
Cheers,
Matt
Hi Matt, No, this demo is really just a simple sample on setting up a basic one-to-one call. It only supports one session with 2 users talking to each other. You can freely use what I have here but it's going to require custom development to get it working how you want. Best Regards, Joe Lopez
Hi,
I love this project very much as it's very native!
I'm not able to connect 2 clients using the same ID. Can I set some ID using a GET variable ?
Thanks for this project!
Cheers,
Matt