Let's say I have access to a server via ssh and my own laptop. However, I wish to use webrtc to transmit data via a peer connection for it's ultra low latency over UDP. It's also crucial that I have NO communication between a third server.
What I would like to do, is use ssh to get the signalling information for the two machines (offers and answers) to then create a webRTC channel. This way I can say that no information ever went to a third server (including the signalling part).
I understand the requirement is a bit silly, but it's due to how we network things currently, and the important part is the super low latency webRTC provides.
Or is this the wrong tool for the job perhaps? Open to other suggestions :)
I'm developing a web application that want to share the posts that one user posted in one topic argument via webRTC.
The problem is that webRTC allow only one-to-one peer connection.
What can i do for solve this problem using webRTC?
I need that my information is transfered broadcast to the other peer that are connected in the same room.
For this you need some signaling server like easyrtc, which packages some really awesome functionalities. One of those is multy-party videocalls.
If you want to read more about, and see some cool demos check https://easyrtc.com/ and https://demo.easyrtc.com/demos/
After you establish the connection via signalling server, you can open the DataChannel and make the transfer that you said..
I'm trying to achieve peer to peer video conference using google stun server.
I can connect anyone by stun server randomly.Because stun gives multiple and random addresses and connect with it.
But is there any way to connect specific peer by stun server for a login based system or room based system?
I want to achive something like - https://apprtc.appspot.com/
You need to design your signalling method (this is up to the application developer), which is independent of STUN.
WebRTC does not specify the mechanism for signalling. Signalling is the method whereby users discover each other and establish that a call (media streams between two peers) is going to take place.
The 'discovery' process could involve a registration-based system (eg using SIP proxy) or room based where two users have access to a 'room' (by knowing the credentials or some means of authentication). Once two peers have found each other, their browsers then need to share and negotiate network topology and media capabilities to ensure that the streams can reach the intended destination and can be encoded/decoded properly.
What I want is, basically, to create a connection between two different computers on same local network. But i want to do this by computers' local IP's. (like 192.168.2.23 etc)
This must be a totally local connection. no TURN or STUN Servers. I am not sure if this is possible. Because there are not much documentation/example/information about WebRTC.
So, how can I create a connection from my computer to another one just passing its local IP as parameter?
Update: To be more clear; imagine there is an html page contains some code that activates my camera and audio services. and another -almost same- page is open in other computer. Waiting a connection request... And there is a textbox in my page to type an IP belongs to other computer on my local network. type 192.168.2.xx and bingo! i have connection between me and other computer.
I want this process as IP based, because there may be more than 2 devices on the network. And all of them are possible devices to create connection. So i need to reach them by their IP's.
Any example code or explanation would be great! even if it tells that this is not possible.
Thanks
Peer discovery is a vital part in any WebRTC application. It's an expensive term for saying: "Hi, I'm computer 4 and I want to talk to you!".
See it as calling a friend over the phone. You need to dial his number first.
This part is not defined in the WebRTC standards. You need to implement this logic in your application. Once you know who you want to call, you need a way of exchanging vital information. This is called signaling, like flo850 put in his answer.
Signaling is needed before any peer-to-peer connection can be set up.
To come up with an idea for your use case of 7 devices in a LAN.
If you have these devices connected to for example a WebSockets server and are in the same channel.
The WebSockets server can be written to route messages to specific receivers.
Devices connected to the channel often are identified with some kind of ID, imagine you use the device's IP.
When you want to talk to computer 4 with IP 192.168.0.4 you send the exchange messages (signaling) on the channel to the receiver with ID, the IP of the device you want to connect with.
How to send the signaling (offer, answer) is described here with example code.
Hope this helps
Users usually sit behind NATs; that's why ICE concept implemented in WebRTC.
If both users are sitting behind same NAT; you can skip ICE servers by passing "NULL" parameter value over "RTCPeerConnection" constructor:
var peer = new [webkit|moz]RTCPeerConnection ( null );
Now, browser will use "host" candidates, also known as "local" candidates.
you still need a signaling server. During the ICE candidate search, your clients will exchange their local ip through this signaling server
I'm interested in Peer-to-Peer connections in the browser. Since this seems to be possible with WebRTC, I'm wondering how it works exaclty.
I've read some explanations and saw diagrams about it and now it's clear to me, that the connection establishmet works over the server. The server seems to exchange some data between the client that are willing to connect to each other, so that they can start a direct connection, that is independent of the server.
But that's exaclty what I don't understand. Until now, I thought the only way to create connections is to listen on a port on computer A and connect to that port from computer B. But this does not seem to be the case in WebRTC. I think none of the clients starts to listen on a port. Somehow, they can create a connection without listening on ports and accepting connections. Neither client A, nor client B starts acting as a server.
But how? What data is exchanged over the WebRTC server, that the clients can use to connect to each other?
Thanks for your explanations for this :)
Edit
I found this article. It's not related to WebRTC, but I think it answers a part of my question. I'm not sure, tough. It still would be cool, if someone could explain it to me and give me some additional links.
WebRTC gives SDP Offer to the client JS app to send (however the JS app wants) to the other device, which uses that to generate an SDP Answer.
The trick is that the SDP includes ICE candidates (effectively "try to talk to me at this IP address and this port"). ICE works to punch open ports in the firewalls; though if both sides are symmetric NATs it won't be possible generally, and an alternative candidate (on a TURN server) can be used.
Once they're talking directly (or via TURN, which is effectively a packet-mirror), they can open a DTLS connection and use it to key the SRTP-DTLS media streams, and to send DataChannels over DTLS.
Edit:
Acronyms here: http://blog.1click.io/10-jargons-abbreviations-for-webrtc-fans/ for the rest, there is Google. Most of these are defined by the IETF (http://ietf.org/)
Edit 2:
Firefox and Chrome (and the spec) have moved to using "trickle" for ICE candidates, so the ICE candidates are generally added after-the-face to the PeerConnection and exchanged independently of the initial SDP (though you can wait until the initial candidates are ready before sending an offer, and bundle them together).
See https://webrtcglossary.com/trickle-ice/ and https://datatracker.ietf.org/doc/draft-ietf-ice-trickle/
How WebRTC Works
This document provides a quick and abstract introduction to WebRTC. In order to get more information about WebRTC please look at the Further Reading section at the end of this document.
WebRTC
WebRTC(Web Real-Time Communication) is a set of technologies that is developed for peer to peer duplex real-time communication between browsers. As its name mentions it is compatible with Web and it is a standard in W3C One of the important feature of WebRTC is that it works even behind NAT addresses.
WebRTC uses several technologies to provide real-time peer to peer communication between browsers. These technologies are
SDP (Session Description Protocol)
ICE (Interactivity Connection Establishment)
RTP (Real Time Protocol)
There is one more thing which is Signalling Server is needed for running WebRTC. However, there is no defined standart in implementing signalling server. Each implementation creates its own style. There will give some more information about Signalling Server later in this section.
Let's give some quick info about technologies above.
SDP (Session Description Protocol)
SDP is a simple protocol and it is used for which codecs are supported in browsers. For instance, assume that there are two peers(Client A and Client B) which will be connected through WebRTC. Client A and Client B create SDP strings that defines which codecs they support. For example, Client A may support H264, VP8 and VP9 codecs for video, Opus and PCM codecs for audio. Client B may support only H264 for video and only Opus codec for audio. For this case, the codecs that will be used between Client A and Client B are H264 and Opus. If there are no common codecs between peers, peer to peer communication cannot be established.
You may have a question about how these SDP strings are sent between each others. This is where Signalling Server takes place.
ICE (Interactivity Connection Establishment)
ICE is the magic that establishes connection between peers even if they are behind NAT. Let's assume again Client A and Client B will get connected and take a look at how ICE is used for that.
Client A finds out their local address and public Internet address by using STUN server and sends these address to Client B through Signalling Server. Each addresses received from STUN server is called ICE candidate
In the image above, there are two servers. One of them is STUN and other of them is TURN server.
STUN server is used to let Client A learn its all addresses. Let me give an example for this, our computers generally has one local address in the 192.168.0.0 network and there is a second address we see when we connect to www.whatismyip.com, this IP address is actually the Public IP address of our Internet Gateway(modem, router, etc.) so let's define STUN server; STUN servers lets peers know theirs Public and Local IP addresses. Btw, Google provides free STUN server(stun.l.google.com:19302).
There is a one more server, TURN Server, in the image. TURN Server is used when peer to peer connection cannot be established between peers. TURN server just relays the data between peers.
Client B does the same, gets local and public IP addresses from STUN server and sends these addresses to Client A through Signalling Server.
Client A receives Client B's addresses and tries each IP addresses by sending special pings in order to create connection with Client B. If Client A receives response from any IP addresses, it puts that address in a list with its response time and other performance credentials. At last Client A choose the best addresses according to its performance.
Client B does the same in order to connect to Client A
RTP (Real Time Protocol)
RTP is a mature protocol for transmitting real-time data. It is based on UDP. Audio and Video are transmitted with RTP in WebRTC. There is a sister protocol of RTP which name is RTCP(Real-time Control Protocol) which provides QoS in RTP communication. RTP is also used in RTSP(Real-time Streaming Protocol)
Signalling Server
The last part is the Signalling Server which is not defined in WebRTC. As mentioned above, Signalling Server is used to send SDP strings and ICE Candidates between Client A and Client B. Signalling Server also decides which peers get connected to each other. WebSocket technology is generally used in Signalling Servers for communication.
Compatibility
In the last one year, all browsers including Safari, Edge have released new versions supporting WebRTC. Chrome, Firefox and Opera have already supported WebRTC for a while. The video codec that is common to browsers are H264. For the audio, Opus is common in browsers. PCM can also be used for audio codec but AAC is not used even if AAC is supported in all browsers because of licensing issues. IP Cameras generally support H264 for video codec and PCM or AAC for audio codec.
Further Reading and References
WebRTC Samples
ICE Wikipedia
SDP
RTP RFC
Getting Started with WebRTC
WebRTC.org
STUN Server
RTSP Wikipedia
Btw, I am developer at Ant Media Server which supports scalable one-to-many WebRTC and peer to peer WebRTC connection
Establishing a p2p WebRTC connection has 3 steps (10.000 feet overview) :
Step 1: Signaling: both peers connect to a signaling server (using websockets over 80/443, comet, SIP,etc..) and exchange information (about their media capabilities, public IP:port pairs when they become available, etc.)
Step 2: Discovery: Devices connected to LAN or mobile networks are not aware of their public IP (and port) where they can be reached at so they use STUN/TURN servers located on the public Internet to discover their ip:port pair (ICE candidates). In the process they punch a hole through the NAT/router which is used in step3:
Step 3: P2P connection: once the ICE candidates are exchanged through the initial signaling channel each peer is aware of each other's ip:port (and holes have been punched in NATs/routers) so a peer to peer UDP connection can be established.
The scheme above explains the process with 2 devices connected to local networks. It's part of an article I wrote that deals with troubleshooting connection issues but it does a good job of explaining how WebRTC works.
A very good explanation can be found in this book "High Performance Browser Networking (O'Reilly)" http://chimera.labs.oreilly.com/books/1230000000545/ch03.html#STUN_TURN_ICE
which provides the fundamentals on how WebRTC uses ICE technology.
In particular assuming the IP address of the STUN server is known, the WebRTC application first sends a binding request to the STUN server. The STUN server replies with a response that contains the public IP address and port of the client as seen from the public network.
Now the application discovers its public IP and port tuple which can send to the other peer through SDP. (note that SDP are sent over an external signalling channel, f.i. websocket established through a web service)
With this mechanism in place, whenever two peers want to talk to each other over UDP, they can then use the established public IP and port tuples to exchange data.
Unfortunately, in some cases UDP may be blocked by a firewall. To address this issue, whenever STUN fails, we can use the Traversal Using Relays around NAT (TURN) protocol as a fallback, which can run over UDP and switch to TCP if all else fails.
WebRTC connection starts with WebRTC offer.
Caller creates the WebRTC offer and posts it to the Signaling server which will pass the offer to the callee. Users actually passing their SDP (Session Description Protocol) information each other.
Then we need to exchange the internet connection details. it allows clients to discover their public IP address and the type of NAT they are behind. this information is used to establish the media connection. This is handled by STUN server. this process is also known as getting the ICE Candidates. This data is also exchanged via the Signalling Server.
Final step is to exchange audio and video streams true TURN (Traversal Using Relay NAT) server. This ensures the connection even thought users are behind the firewall. this server process a lot of heavy calculations so its cost is high. when you test your app in dev with different browsers, you are directly connecting each other, you are not using TURN server