I am a newbie to WebRtc. I would like to send the media stream from client ( java script) to my server (via websockets). In my server I will be doing some processing on those media content.
Could you please show me a client code snippet on sending media stream to Server via websocket.
Thanks
Ganesh.R
Nobody can show you this, because you cannot send the stream via Websockets. You need to read a little more about WebRTC.
WebRTC give you the possibility to request access to media devices from Javascript, and allows you to create a PeerConnection that will establish a connection to another endpoint to send the streams captured from the devices or some raw data (using DataChannel).
You won't have access to the streams data to send via WebSockets. Instead, the browser will send it over UDP or TCP using the SRTP protocol. If you want to get media streams on server side, you will need to implement this protocol and some negotiation to establish the connection.
HTML5Rocks have a great introduction with code snippets to start.
Related
I'm developing an application that receives an audio stream over a WebSocket and needs to forward the audio to a SIP server.
Currently, I've managed to connect to the audio source using a Websocket and receive the media stream (encoded u-law) using Node-Red, but I'm struggling to figure out how to send the media stream to the SIP server. Any advice would be much appreciated.
I looked into this for a similar question a while back, can't find where it was now.
As you probably know the media part of SIP is RTP, so its a fairly separate stack to the call signalling.
I didn't find any nodes that supported it and the few node.js libraries for RTP were all very incomplete and out of date.
In theory it might be possible to craft your own RTP streams using the UDP nodes and then create the relevant SDP in the SIP response but I'm not sure how robust or scalable this would be.
The other option is that there are a couple of Programmable Comms platforms out there that support both SIP and Web sockets so you could possible utilise one of those and connect from Node-RED via websocket letting them do the SIP work.
I've done SIP|<>Websocket stuff with both the Vonage API (Previously Nexmo) and Jambonz (open source)
I'm looking at a proto file which has a bidirectional stream between the client and the server. Does this mean that the client and server can send and receive messages arbitrarily? I'm more confused about the server side. How can the server send data over this bidirectional stream arbitrarily? What would be the trigger?
Thanks!
From the docs:
In a bidirectional streaming RPC, again the call is initiated by the
client calling the method and the server receiving the client
metadata, method name, and deadline. Again the server can choose to
send back its initial metadata or wait for the client to start sending
requests.
What happens next depends on the application, as the client and server
can read and write in any order - the streams operate completely
independently. [...]
This means: the client would establish the connection to the server and you'd then have a connection on wich both parties can read/write.
For my app I'm streaming audio data from a raspberry-pi client to my node.js audio service through socket.io. The problem is, to process the audio, I'm piping the audio stream from client in my service to an external service. Then this external service will give the result stream audio back to my service and my service will emit it to the client.
So my application flow is like
Client ---socket.io-stream---> audio_service ---stream---> external_service
external_service --stream---> audio_service ---socket.io-stream---> client
My questions is:
Is it possible that when a client connected to my audio_service, my audio_service will initiate a connection to external_service and emit that connection back to the client through socket.io? This way the client will stream audio directly to the external_service by using the returned connection instead of going through audio_service.
If it is possible, is it also possible that even though the client stream audio directly to the external_service, it will still send the stream result back to the audio_service?
Thank you very much for your help
It isn't possible to send a stream through Socket.IO the way it is set up today. Some folks have made some add-ons that emulate streams in the usual evented RPC way, but it isn't very efficient.
The best library I know for this is Binary.JS. This will give you streams multiplexed over a single binary WebSocket connection. Unfortunately, Binary.js hasn't been maintained in awhile, but it still works.
I'm new to socket.io. In Realtime (Web) Applications, we used to choose whether it should be WebRTC or WebSocket (or even SIP, still?) technologies.
What exactly is socket.io in this case please?
WebSockets
socket.io is a popular open source library implemented on both backend and client side. This library is based on WebSockets API which allow a communication between a SERVER and a CLIENT.
WebRTC
On the other hand, WebRTC is a WebAPI which comes with basically 3 things:
Real Time Communication between two browsers (no server needed), a peer to peer connection (P2P)
Media Streaming (Audio and Video)
Real Time Communication Data Chanel (stream any data on P2P)
The main difference is that WebSockets needs A SERVER and it is based on publish/subscribe pattern where you can send raw data back and forth, without having any special data handling by default. In contrast, WebRTC has a lot of functions already in place which can be used to handle Audio/Video streaming and also the raw data with data chanel.
For more info I recommend reading on MDN links I provided above and also check this very cool slides on sockets and webRTC
If you want to make video or audio communication services use WebRTC for browser build in support and write the discovery and signaling. webrtc have awesome features like P2P connections and data encryption.
WebRTC client side (browser) features like get video and audio data with good support in evergreen browsers: http://iswebrtcreadyyet.com/#interop
And socket.io is good for build centralized pub / sub apps like text chat
You can make connections with WebRTC without socket.io but both works fine if you use socket.io for help in signaling
I know it's possible to stream content from a server to a web browser. I'm curious if there is a method to allow the same in reverse; and once a page is requested/connection established to then try and keep it open and stream content to the server from the client without continually re-initiating a connection/POSTing to the server. I have tried searching for such a thing but I always find information about http streaming from server to client and not the other way around. This is specifically to target mobile devices and stream chunks of text up to the server.
Thanks in advance for any info or advice!