Alessandro Amirante

On the seamless interaction between webRTC browsers and SIP-based conferencing systems

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.

Improving the scalability of an IMS-compliant conferencing framework through presence and event notification

In this article we present a thorough performance analysis of an actual implementation of a distributed conferencing framework compliant with the IP multimedia core network subsystem specification. The subject of our experimental campaign has been an architecture implemented by exploiting existing achievements in the field of conferencing. The work presented is based on the standard centralized conferencing framework defined by the IETF XCON working group, for which we recently provided an open source implementation. Starting from here, we devised a distributed conferencing solution that has been implemented as an overlay network of centralized conferencing clouds able to: (i) discover each other by means of presence mechanisms; (ii) exchange conference related information through an event notification paradigm; (iii) manage distributed conferences through out-of-band dedicated channels. The study is conducted by first measuring the main performance figures of the centralized solution and then assessing the capability of the distributed framework to improve such performance by scaling both in the number of supported users/conferences and in the overall resource consumption.

Janus: a general purpose WebRTC gateway

This paper deals with the design and implementation of Janus, a general purpose, open source WebRTC gateway. Details will be provided on the architectural choices we took for Janus, as well as on the APIs we made available to extend and make use of it. Examples of how the gateway can be used for complex WebRTC applications are presented, together with some experimental results we collected during the development process.

Performance analysis of the Janus WebRTC gateway

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.

Improving the Scalability of an IMS-Compliant Conferencing Framework Part II: Involving Mixing and Floor Control

At IPTComm 2007 we presented a paper in which we analyzed the performance of a distributed conferencing framework we have recently designed and implemented. The paper focused on the scalability of the overall framework and demonstrated that the switch from a centralized to a distributed approach can significantly improve the achievable performance. Though, in the cited work, we mainly addressed the signalling aspects of the framework. We did not make any attempt at assessing the performance of our architecture (neither in the centralized, nor in the distributed case) in the presence of mixed media streams. This left us, and perhaps the readers, with a fine sensation of unfinished and motivated us to keep on working on the measurements, this time focusing on the two main aspects that come into play when a real conference is activated in a moderated environment like the one we propose: (i) mixing of the real-time streams generated by the users; (ii) floor control.

Meetecho Mobile: Accessing an IETF-compliant conferencing framework from cellular devices

In this article we present Meetecho Mobile, an innovative conferencing client designed for mobile devices. Such a client provides access to conferences compliant with the most recent standard proposals currently under definition within the IETF. It allows mobile users to participate in multimedia conferences involving, besides an XMPP-based chat, both audio and video streams. The article discusses the design and implementation of the conferencing client and highlights the most notable solutions we devised in order to effectively handle a whole suite of protocols, ranging from signaling (SIP/SDP, XMPP) to real-time streaming (RTP). We also provide information about the main performance figures characterizing the clients behavior.

Standard multimedia conferencing in the wild: the Meetecho architecture

We present a conferencing architecture called Meetecho. To the purpose, we embrace an engineering approach, by describing an actual implementation of an open source centralized video-conferencing system capable to offer advanced communication experience to end-users through the effective exploitation of mechanisms like session management and floor control. Meetecho has been designed to be fully compliant with the latest standard proposals coming from both the IETF and the 3GPP and can be considered as an outstanding example of a real-time application built on top of the grounds paved by the SIP protocol. We will discuss both the design of the overall conferencing framework and the most important issues we had to face during the implementation phase.

Online non-intrusive diagnosis of one-way RTP faults in VoIP networks using cooperation

We address the well-known issue of one-way RTP flows in VoIP communications. We investigate the main causes that usually lead to this type of fault, and we propose a methodology allowing for their automated online detection and diagnosis. The envisaged approach exploits node cooperation and is based on a more general framework for network faults diagnosis called DYSWIS (Do You See What I See). As most of the problems associated with one-way RTP can be ascribed to the presence of NAT elements along the communication path, one of the key features of the proposed methodology resides in the capability to detect such type of devices. Besides, another important aspect of this work is that the diagnosis is non-intrusive, meaning that the whole process is based on the passive observation of flowing packets, and on silent active probing that is transparent to the users. In this way, we also avoid the possibility of being classified as SPIT (SPam over Internet Telephony). We provide a thorough description of the various steps the diagnosing process goes through, together with some implementation details as well as the results of the validation process.

Recording and playout of multimedia conferencing sessions: a standard approach

This paper deals with the challenge of session recording, with special reference to multimedia conferences, both centralized and distributed. Standardization efforts related to both multimedia conferencing and session recording are presented in the paper, and taken into account in our proposal. In the approach we herein present, each involved medium is recorded separately, and is then properly tagged in order to maintain related metadata information. A SMIL (Synchronized Multimedia Integration Language) description is used as a standard presentation format in order to put the individual recordings altogether and handle their synchronization. Such SMIL metadata can subsequently be exploited by an interested user by means of a compliant player in order to start the playout of the integrated multimedia conference session. The paper also describes a complete implementation of the proposed recording architecture made available by our Meetecho conferencing platform.

Meetecho: A Standard Multimedia Conferencing Architecture

Herein, we introduce Meetecho , an architecture capable to provide multimedia conferencing functionality in a standard way. This architecture is an actual implementation of both the framework and the protocols defined within the IETF by the XCON Working Group, and already revealed itself as a useful testbed supporting the ongoing standardization activities.