How can we make end-to-end encryption protocols found in chat applications such as whatsapp, telegram, signal, or how to make only encrypted messaging protocol.
Related
If the signalling part of webrtc is handled by sip, is it mendatory that sip signalling has to be encrypted by TLS. Is it mendatory that sip over websocket has be encrypted as TLS
WebRTC uses TLS sessions or QUIC for its signaling transport – both are encrypted in nature. All other avenues for non-encrypted signaling don’t really exist in WebRTC. Theoretically they might work, but some browsers will either block them altogether or require the user to grant access to the camera and microphone on each interaction. Deploying anything serious to production with WebRTC without encrypting signaling for their browser implementations is not a real alternative.
WebRTC “forces” you to encrypt your signaling. What is left out of scope of WebRTC are things like authentication, authorization and identity management. You are free to do as you please in that domain – just make sure to do something here – and not leave this wide open for pranksters or worse.
For native applications on mobile, desktop or embedded – you can do whatever you like. That said, the mindset must be the same – mandatory encrypted signaling.
reference : Best webrtc related blog
WebRTC uses DTLS for encryption of SCTP (data). Furthermore it uses DTLS for key exchange of SRTP (media). But I do not quite understand, whether it also uses DTLS for encryption of SRTP. So my question: Does WebRTC use DTLS solely for key exchange (DTLS-SRTP) or does it encrypt SRTP additionally to its internal encryption mechanism (at least optionally)?
Why I am asking:
RFC 6904 as well as RFC 3711 state that SRTP provides authentication, but not encryption, of the headers of RTP packets. So only the payload of SRTP is encrypted by design.
What confuses me:
Many sources claim WebRTC to be fully end-to-end encrypted.
I am especially interested in the implementation of WebRTC in gstreamer.
DTLS is used for the handshake, but then the keys are extracted and a SRTP context is initialized.
I am not familiar with GStreamer, but in Pion WebRTC we connect via DTLS here. The WebRTC clients negotiate which side is a DTLS Server and which is a DTLS Client via the SDP.
When the DTLS handshake is complete you then export the keying material, you can see that here
If you are interested in seeing how SRTP works you can check out pion/srtp. It is pretty simple though, it is just AES and then you generate an authentication tag. You can see it all here
I am interested in experimenting with WebRTC data streams as a method of low-latency communication between peers in multiplayer games in the browser, but have read that WebRTC is always encrypted. From http://webrtc-security.github.io/
Encryption is a mandatory feature of WebRTC, and is enforced on all components, including signaling mechanisms. Resultantly, all media streams sent over WebRTC are securely encrypted, enacted through standardised and well-known encryption protocols. The encryption protocol used depends on the channel type; data streams are encrypted using Datagram Transport Layer Security (DTLS) and media streams are encrypted using Secure Real-time Transport Protocol (SRTP).
It seems odd to me to have encryption so tightly coupled in this way. I can think of contexts where mandatory encryption is a hindrance, like multiplayer gaming where data transferred is non-sensitive and having to encrypt/decrypt data is unnecessary overhead, however small. For comparison, WebSockets communication is encrypted further down the stack by TLS if present, but the option of communicating over plain TCP is available as well. Am I misunderstanding this or is there a reason why it was decided that all WebRTC communication must be encrypted?
WebRTC is defining new realm for browsers which was not primarily intended. IMO Accessing camera/microphone rises enough resistance to use the technology solely so mandatory encryption gonna ease it.
The same reason as HTTPS communication is encrypted, they're trying to avoid a middle man to get advantage of the peer to peer communication
I have a microservices based web app. Microservices communicate with each other via a REST API exposed. I want an easy, yet secure solution to secure communication between my microservices. I've already used JWT protocol to secure my user-services communication but I can't figure out the best way to secure server-server communication.
Update:
I want an easy way to authenticate APIs. Is is a good way to hardcode key and secret or put them in configurations files and then use them to authenticate to an other end point?
I've heard about OAuth2 protocol but I'm afraid it's an overkill for my need.So What can be the easy and secure way to authenticate APIs?
You should use HTTPS in order to make communication between servers secure. As far as point to point security (transport layer security) is concerned this is the way to go.
But keep in mind that this still doesn't mean that you'll have message-level security (end-to-end security). Intermediaries (i.e. service agents or other services and applications) along the message path will be able to see what is in the message content while processing it.
REST relies on the uniform contract provided by HTTP, so you cannot use the advanced features of WS-Security as you would have with SOAP. The security features of SOAP provide a wider spectrum of options, so if security is key in your case, you should definitely check SOAP web services out.
Also, take a look at this question. It's relevant to yours and I'm sure you'll find it helpful.
Hope this helps!
By WebRTC standard all connections should be encrypted via DTLS. This is great. However, unless application verifies authenticity of the peer, the connection is vulnerable to man in the middle (MITM) attack.
The question is how to do this with libWebRTC (http://www.webrtc.org/native-code) specifically in Objective C interface. Ideally, I would like to be able to specify my own certificate for WebRTC connection. In this case I will be able to verify it through my application-specific secure channel. If this is not possible, then what is the suggested approach? I will be grateful for any hints.
The WebRTC specification does not include the signaling layer, i.e., how the fingerprints, ICE candidates, etc are exchanged between peers. This means that in almost all WebRTC applications, there's a signaling socket to rapidly signal the connection with a service which exchanges the information between two peers.
Instead of supplying your own certificates or fingerprints, what you want to do is authenticate the signaling over which those are sent.
Your signaling server effectively is the man-in-the-middle, especially in cases of using an MCU or SFU. It should verify the identity of clients exchanging signaling information, and ensure the signaling information is exchanged between mutually agreeing peers (i.e., calling one another by address, or joining a common "room").
So the answer is: use secure websockets, and authenticate clients sending/receiving WebRTC offer/answer information. If you do that, there's no risk of man-in-the-middle attacks, beyond the risk of any other HTTPS+WSS application.