The growing interest in integrating interactive multimedia features into web applications has recently led to the creation of the W3C WebRTC and the IETF RTCWEB working groups. Such groups are jointly defining both the application programming interfaces and the underlying communication protocols for the setup and management of a reliable communication path between any pair of next-generation web browsers. While the ongoing work is focusing on peer-to-peer communication between browsers, engineers are also facing a new issue, associated with the coexistence of legacy SIPbased systems with the upcoming browserenabled architectures. We herein discuss how we tackled such an issue, by first identifying interoperability requirements and then presenting a real-world interoperability example dealing with the integration of RTCWEB clients into an existing standards-based collaboration platform
This paper takes an in-depth look at the performance of the Janus WebRTC gateway. Janus is a modular, open-source gateway allowing WebRTC clients to seamlessly interact with legacy real-time communication technologies, both standard and proprietary, and with each other. This is achieved by attaching technology-specific plugins on top of a barebones core implementing all of the functions and protocols mandated by the RTCWEB/WebRTC specification suites. The paper focuses on assessing the scalability of the Janus architecture, by selecting three representative use cases, followed by a detailed analysis of a real-world scenario associated with multi-point audio conferencing.
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