arXiv subjects

Kundan Singh

Publications and source records attributed to Kundan Singh.

5 recordsLinked to original sources

arXiv2026

A Generic Web Component for WebRTC Pub-Sub

We present video-io, a generic web component to publish or subscribe to a media stream in WebRTC (web real-time communication) applications. Unlike a call or conference room abstraction of existing video conferencing services, it uses a named stream abstraction, which is useful in many scenarios beyond just a call or conference. It keeps most of the application logic in the endpoint using the extensive application interface of this component, and keeps any vendor specific access control or signaling negotiation in a service-specific connector implementation. This allows an app developer to write once, and be able to run the web app on different servers or services. We also demonstrate its flexibility by implementing the connector for ten different existing systems and services. Decoupling the app from the hosted vendor service promotes innovation in the endpoint beyond what a single vendor locked client app can offer.

cs.MM↗

arXiv2026

Lightweight Call Signaling and Peer-to-Peer Control of WebRTC Video Conferencing

We present the software architecture and implementation of our web-based multiparty video conference application. It does not use a media server. For call signaling, it either piggybacks on existing push notifications via a lightweight notification server, or utilizes email messages to further remove that server dependency. For conference control and data storage, it creates a peer-to-peer network of the clients participating in the call. Our prototype client web app can be installed as a browser extension, or a progressive web app on desktop and mobile. It uses WebRTC data channels and media streams for the control and media paths in implementing a full featured video conferencing with audio, video, text and screen sharing. The challenges faced and the techniques used in creating our lightweight or serverless system are useful to other low-end WebRTC applications that intend to save cost on server maintenance or paid subscriptions for multiparty video calls.

cs.MM↗

arXiv2025

Unlocking WebRTC for End User Driven Innovation

We present a software architecture to enable end user driven innovation of web multimedia communication applications. RTC Helper is a simple and easy-to-use software that can intercept WebRTC (web real-time communication) and related APIs in the browser, and change the behavior of web apps in real-time. Such customization can even be driven by the end user on third-party web apps using our flexible and general purpose browser extension. It also facilitates rapid prototyping of ideas by web developers in their existing web apps without having to rebuild or redeploy after every change. It has more than ten customization categories, and over a hundred built-in examples covering a wide range of novel use cases in web-based audio/video communication.

cs.MM↗

arXiv2011

SIP APIs for Voice and Video Communications on the Web

Existing standard protocols for the web and Internet telephony fail to deliver real-time interactive communication from within a web browser. In particular, the client-server web protocol over reliable TCP is not always suitable for end-to-end low latency media path needed for interactive voice and video communication. To solve this, we compare the available platform options using the existing technologies such as modifying the web programming language and protocol, using an existing web browser plugin, and a separate host resident application that the web browser can talk to. We argue that using a separate application as an adaptor is a promising short term as well as long-term strategy for voice and video communications on the web. Our project aims at developing the open technology and sample implementations for web-based real-time voice and video communication applications. We describe the architecture of our project including (1) a RESTful web communication API over HTTP inspired by SIP message flows, (2) a web-friendly set of metadata for session description, and (3) an UDP-based end-to-end media path. All other telephony functions reside in the web application itself and/or in web feature servers. The adaptor approach allows us to easily add new voice and video codecs and NAT traversal technologies such as Host Identity Protocol. We want to make web-based communication accessible to millions of web developers, maximize the end user experience and security, and preserve the huge global investment in and experience from SIP systems while adhering to web standards and development tools as much as possible. We have created an open source prototype that allows you to freely use the conference application by directing a browser to the conference URL.

cs.NI↗

arXiv2011

Flash-based Audio and Video Communication in the Cloud

Internet telephony and multimedia communication protocols have matured over the last fifteen years. Recently, the web is evolving as a popular platform for everything we do on the Internet including email, text chat, voice calls, discussions, enterprise apps and multi-party collaboration. Unfortunately, there is a disconnect between web and traditional Internet telephony protocols as they have ignored the constraints and requirements of each other. Consequently, the Flash Player is being used as a web browser plugin by many developers for web-based voice and video calls. We describe the challenges of video communication using a web browser, present a simple API using a Flash Player application, show how it supports wide range of web communication scenarios in the cloud, and describe how it can interoperate with Session Initiation Protocol (SIP)-based systems. We describe both the advantages and challenges of Flash Player based communication applications. The presented API could guide future work on communication-related web protocol extensions.

cs.NI↗