WebRTC-SS

WebRTC Signaling Server & Simple Video Chat

Simple Signaling Server to provide basic WebRTC functionality

1. Create chat rooms providing their links web publishing;
2. Establish client-service WebSocket connections;
3. Implement WebRTC SDP Offers and ICE Candidates negotiation;
4. Test peer-to-peer video chatting.

Project Desription

This project has been built with Gradle, server side is written in Java 8 using Spring Boot, client is in Java Script. Application web entry: http://localhost:8080 (https://localhost:4883 for the Secure Test version on the test branch). The back side uses the following frameworks and technologies:

• Spring Boot 2.0.5 with starter web, validation, and thymeleaf dependencies needed for the web server;
• Spring Web Socket to establish connection between signalling server and clients;
• HTML 5 to provide Web RTC interaction between clients;
• Spring Boot Starter-test provides JUnit 4, Mockito, AssertJ, other libraries used to test back-side.

Git Repository Structure

This GitHub repository has 3 branches:

• main - latest stable version working on http://localhost:8080
• develop - to maintain product development;
• test - includes self-signed sertificates to test video chatting on mobile devices (such as Android Chromium, which forbid http connection to local host), use https://localhost:4883 to reach this version, and let your browser ingnore a warning about the certificate.

Project Structure

• MainController provides HTTP requests handling, model processing and view presentation;
• Domain package includes domain model and service;
• Web Socket based Web RTC Signalling Server is located under the Socket directory;
• Config and Util packages contain configuration and utility classes respectively;
• Test section contains unit and integration backside tests;
• Resources folder keeps application.yml configuration file, certificates (in test branch), frontside templates and static content (CSS, Java Script files).

API

Method	URI	Description
Get	"", "/", "/index", "/home", "/main"	main page application web entry point
Post	"/room"	process room selection form
Get	"/room/{sid}/user/{uuid}"	select a room to enter; sid - room number, uuid - user id
Get	"/room/{sid}/user/{uuid}/exit"	a room exit point for the user selected
Get	"/room/random"	generates random room number
Get	"/offer"	demonstrates sample SDP offer
Get	"/stream"	demonstrates streaming video resolution selection

Building and Running Docker Image

1. Build a Docker image running in the Terminal ./gradlew build docker --info
2. Select recently built image, tagged benkoff/webrtcss-spring-boot-docker:latest
3. Push selected image to Docker hub
4. Pull and run uploaded image in local Docker with docker run -p 8080:8080 -t benkoff/webrtcss-spring-boot-docker

Useful Links and Sources

Documentation and Tutorials:

• https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API/Signaling_and_video_calling
• https://codelabs.developers.google.com/codelabs/webrtc-web/
• https://webrtc.github.io/samples/
• https://andrewjprokop.wordpress.com/2014/07/21/understanding-webrtc-media-connections-ice-stun-and-turn/
• https://nextrtc.org/
• https://www.html5rocks.com/en/tutorials/webrtc/basics/
• http://w3c.github.io/webrtc-pc/
• https://www.scaledrone.com/blog/webrtc-chat-tutorial/
• http://builds.kurento.org/release/stable/docs/tutorials/node/tutorial-4-one2one.html

Sample Implementations:

• https://github.com/webrtc/samples
• https://github.com/webrtc/apprtc/blob/master/src/collider/collider/collider.go
• https://github.com/mslosarz/nextrtc-signaling-server
• https://github.com/Kurento/kurento-tutorial-java/blob/master/kurento-one2one-call-advanced/src/main/java/org/kurento/tutorial/one2onecalladv

WebSockets:

• https://docs.spring.io/spring/docs/current/spring-framework-reference/web.html#websocket
• https://keyholesoftware.com/2017/04/10/websockets-with-spring-boot/
• https://www.baeldung.com/websockets-spring