I'm assuming that WebRTC is an API that decodes/encodes audio and video, although the communication between the server and the clients is done via web sockets, or some other network protocol? I'm a bit confused. Does WebRTC have its own communications protocol?
There's two sides to WebRTC.
JavaScript APIs (getUserMedia) that allow an app to access camera and microphone hardware. You can use this access to simply display the stream locally (perhaps applying effects), or send the stream over the network. You could send the data to your server, or you could use...
PeerConnection, an API that allows browsers to establish direct peer-to-peer socket connections. You can establish a connection directly to someone else's browser and exchange data directly. This is very useful for high-bandwidth data like video, where you don't want your server to have to deal with relaying large amounts of data.
Take a look at the demos to see both parts of WebRTC in action.
So in a nutshell:
WebSockets allow full-duplex communication between a browser and a web server.
WebRTC's PeerConnection allows full-duplex communication between two browsers.
WebRTC uses RTP (a UDP based protocol) for the media transport, but requires an out-of-band signaling channel to setup the communication. One option for the signaling channel is WebSocket.
Instead of peerConnection you can also look at the WebRTC data channel draft: https://datatracker.ietf.org/doc/html/draft-jesup-rtcweb-data-protocol-00 which is basically bidirectional udp. Which can be a really valuable alternative to WebSockets as doesn't have the "negative" sides of a tcp connection.
No, Signaling is not defined by WebRTC.
Here is an post by the IETF which explains it pretty good why it is not:
http://www.ietf.org/mail-archive/web/rtcweb/current/msg01143.html
This means that you are free to choose how you exchange network information. I.e. you could use websockets, HTTP and even Email, but that would be a bit of a struggle :)
Related
My current use case is that I'm trying to mock a system that uses WebRTC for live video streaming (for a robot). This way, I don't have to be connected to the robot to develop the client.
My issue as of now is that I have no idea how to stream a video using WebRTC to connected peers. I've seen many examples of how to do this from client to client using a signaling server, but other than directly sending the video buffer using socket.io, I haven't seen an example of server -> client WebRTC streaming.
I'm planning to use Node.JS for mocking the video stream as I've been using it for the rest of the robot's systems.
It really isn't that different though client to client or server to client. You want to stream/broadcast a video to all the connected peers. Think of your server will be a client in the setup.
You can also use a WebRTC solution like Janus Repo it is a simple gateway and completely open source. Refer to - WebRTC & Dev API's for more info.
If you find latency issues after peers have increased in number you can check - Mesh, Routing, Multi peer architecture for some solutions for it.
Hope it helps.
I am trying to do a one-to-many broadcast.
I do not want to use WebRTC for this because a p2p connection is not ideal when many clients are involved.
Is there a way to take a MediaStream from client A, send it to a server, then broadcast it to many other viewer clients?
I have looked into socket.io-stream, but this seems to be more for filestreams and not video streams.
I have thought of sending a screenshot of the screen every so many milliseconds, and sending from client A using socket.emit("frameUpdate",screenshotImgString), but this seems rather resource intensive and does not include handy things like video compression.
Is there a way to simply take a stream object, pass it to the server, and then have the server share it with everyone?
A possibility when using WebRTC is using the Selective Forwarding Unit architecture. But this requires p2p connections between multiple clients with the server, possibly each of which is mediated by a TURN server. So there's servers upon servers processing all this!
Ideally, if I'm gonna be using a TURN server anyways, I would want a single server relaying everything.
Not going Client 1 --> TURN server --> SFU server --> TURN server --> {Client 2...N}
But instead Client 1 --> Video Relay Server --> {Client 2...N}
Everything online seems very WebRTC focused, but I am unsure if this is the best solution to my problem.
My development environment is in Node.js
An SFU is exactly what you're looking for.
With an SFU, a TURN server is normally not needed, data flows directly from the sender to the SFU and then from the SFU to the receiver. (In some rare cases, such as when the client is behind a very restrictive firewall, a TURN server may be useful, but that will be negotiated automatically by the WebRTC stack.)
I'm new to WebRTC, actually just heard about it a few days ago and I've read a lot about it. However, I still have a few questions.
What do I need to explore the usage of WebRTC? E.g.: do I need a server, any libraries etc.? I'm aware that new version of Chrome and Firefox support WebRTC, but besides these two browsers, is there anything else that is necessary?
What is the main purpose of WebRTC when addressing practical usage? To video chat? Audio chat? What about text-chatting?
Does WebRTC need a server for any kind of browser-to-browser interaction? I've seen some libraries, such as PeerJS that don't explicitly mention any kind of server... so is it possible to connect two clients directly? There's also a PeerServer, which supposedly helps broker connections between PeerJS clients. Can I use WebRTC without such a server?
What are the most commonly used libraries for WebRTC?
What's a good starting point for someone who's totally new in WebRTC? I'd like to setup a basic google-talk kind of service, to chat with one person.
Thank you so much guys.
You can find many docs here E.g. this one, this one and this one!
You can find a few libraries here.
A simple multi-user WebRTC app needs following things:
Signalling server to exchange sdp/ice/etc. ---- e.g. socket.io/websockets/xmpp/sip/XHR/etc.
ICE server i.e. STUN and/or TURN; to make sure Firewalls doesn't block UDP/TCP ports
JavaScript app to access/invoke RTCWeb JavaScript API i.e. RTCPeerConnection.
It just takes a few minutes to setup WebRTC peer-to-peer connection. You can setup peer-to-server connections as well where media-servers can be used to transcode/record/merge streams; or to relay to PSTN networks.
WebRTC DataChannels can be used for gaming, webpage synchronizing; fetching static contents, peer-to-peer or peer-to-server data transmission, etc.
What do I need to explore the usage of WebRTC? E.g.: do I need a
server, any libraries etc.? I'm aware that new version of Chrome and
Firefox support WebRTC, but besides these two browsers, is there
anything else that is necessary?
WebRTC it is JavaScript API for web developers which can be used for audio and video streaming.
But there are 2 notices:
You need a signaling path.
For example, if your first user is Alice using Firefox and second user is Bob using Chrome,
they should negotiate used codecs and streams.
WebRTC does not offer the signalling implementation. So you need to implement the signaling yourself. It is quite simple. You need to send SDP(stream config) to participant and receive an SDP answer. You can use plain HTTP via apahe server or use Websockets or any other transport to negotiate SDP.
So, it seems you need an intermediary signaling server workning with websockets or HTTP/HTTPS.
Once you negotiated the streams you are sending your audio or video stream, but the distanation user might have a simmetric NAT. It means that you stream will not be delivered to the target user. In such situation you need a TURN server to traverse the NAT.
Finally you will need 2 server-side logic items:
1) Signaling server
2) TURN or proxy server
To start, take a look Web Call Server.
The server implements HTML5 Websocket signaling and SRTP proxying as a TURN server.
You can also learn the webrtc application open source code.
First steps:
1. Download the signaling and streaming server.
2. Download and unzip web client.
3. Start the web client and debug javascript code to learn more how webrtc works.
How do you send a UDP Packet using Web RTC?
You can't send a UDP packet directly with WebRTC. That would violate the basic security constraints required by the browser.
You can send SRTP to an ICE-enabled host. That is probably not what you are looking for.
If a browser permitted sending arbitrary UDP packets, then malicious applications could send packets to any host.
This might not sound so bad, after all, hosts on the Internet need to be able to deal with that right? The problem is that some browsers are in protected environments where access to the network is restricted. Within those networks, some hosts are far less protected than a host on the public Internet might be. This would be OK, since access to the network is controlled.
If it were possible for a browser to send arbitrary packets, a user on a browser in that environment could be convinced to send specifically crafted packets to one of these poorly protected hosts. Likely, that would result in the network operator banning the browser, which is something browser-makers want very much to avoid.
WebRTC only sends certain types of UDP packet under specific conditions. If the host that you are interested in talking to understands ICE and is able to consume RTP with SRTP or SCTP over DTLS (unlikely methinks). Then perhaps you could force the browser to send something.
You should check sipml5, http://code.google.com/p/sipml5/
get the code and look into the folder:
sipml5/src/tinySIP/src/transports
#Martin Thomson has explained the problem very well. By taking
motivation from his post, this post might provide guidance to some
geeks and newbies.
It is being said that WebRtc is Real-time, Bi-directional, Secure communication between two or more peers. I will more focus on the word: Secure.
Security policies of enterprise networks typically require filtering of incoming unsolicited traffic, blocking certain protocols, and doing application-level filtering and scanning for spam, malware, and intellectual property.
Now two new questions come to mind;
Why is TCP traversal not problematic?
Why is UDP traversal problematic?
Why is TCP traversal not problematic?
TCP clearly indicates two things;
The beginning of a flow (SYN) and,
The end of a flow (FIN or RST).
This is used by firewalls to open and close pinholes. Exceptionally, a TCP connection that has received no traffic for a long time also has its pinhole closed (to accommodate network topology changes or failure of both TCP peers). Firewalls also perform protocol validation to clean up problems with
Out-of-window TCP segments and,
Overlapping TCP segments
This allows the firewall to protect the network and protect hosts from
several attack vectors (replay attacks, host IP address probing, DDOS
attack, etc.).
Why is UDP traversal problematic?
For UDP flows, the first outgoing packet on a 5-tuple will be used by the firewall as a start-of-session indicator. But UDP does not have an end-of-session indicator, so the firewall has only two ways to close a pinhole:
Timing out the pinhole after the interior host does not send traffic
for several seconds or
The interior host generates a fatal ICMP error.
Because there is no reliable way to determine that a session is being stopped, the firewall has a much harder job. It could implement an Application-level Gateway (ALG) and be aware of whatever semantics are imposed by the higher-level code on top of UDP.
It could also rely on a set of well-known application servers to inform it of sessions as they start and end, but that suffers from many challenges as application servers are hosted independently of the network on which they are used.
Using an ALG, a firewall can determine when the call is terminated and
close any dynamic mappings created for the media session. But the
problem is session signaling between the WebRTC application running in
the browser and the Web server that could be using TLS, in which case
the ALG no longer has access to the signaling.
Conclusion:
So, at the Application-Level Gateway, Webrtc utilizes a combination of SDP and ICE. WebRtc basically wraps UDP in such a way that
For Audio, Video Channels, WebRtc uses combinations of RTP, RTCP, SRTP over DTLS.
For Data Channels, webrtc uses the Stream Control Transmission Protocol (SCTP), defined in RFC 2960.
SCTP is a transport layer protocol that was intended as an alternative
to TCP or UDP. For WebRtc, we use it as an application layer
protocol that runs over our DTLS connection.
These protocols also come with some new protocols such as STUN, TURN. For the basic implementations of WebRtc Please follow;
WebRtc Experiments by Mauz Khan
WebRtc Experiments by Muhammad Usman Bashir
I hope this explanation might help some of the geeks. Thanks
I begin with this technology. I want to establish a TCP/IP connection with an electronic card that has an IP address (the server's map).
I wonder if the WebSocket allow me to make this connection, knowing that at present my interface communicates with the card through a socket implanted in an applet.
Does anyone know the syntax to connect with WebSocket as a parameter an IP address: 135.120.138.105
Thank you
WebSockets are not raw TCP sockets. They have many of the same characteristics (low overhead, persistent, bidirectional, full-duplex) as raw TCP sockets, but they have an initial HTTP-like handshake to implement CORS security and allow easier integration with web servers and existing firewall policies. WebSockets are also message based and have a small header on each frame (2 bytes overhead for small payloads).
You have a couple of options. You can use a program that bridges/proxies between WebSockets and raw TCP sockets such websockify (Disclaimer: I made websockify) or you can implement the server side of the WebSocket protocol in your server.
WebSockets won't work in your case. While they use TCP/IP, WebSockets have a different API that's designed for higher-level messages to be passed between server and client and it requires support on the server as well, so unless the card contains a WebSocket-enabled web server I think you're out of luck and you'll have to continue with your applet-based design.