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Summary The network convergence time may take hundreds of milliseconds or more. Since this repair performance may be unsatisfactory, IP fast reroute (IPFRR) approaches have been developed to decrease the network convergence time and minimize traffic loss. These IPFRR mechanisms are usually based on the proactive precomputation of a backup path before a failure occurs. Once a failure is detected, the router running IPFRR will immediately use this backup path in response to the failure, rather than computing a replacement path only after a failure has occurred. The computation of an alternative path in many IPFRR mechanisms requires that the computing node (router) knows the exact network topology. In this paper, we present a new IPFRR mechanism called the multicast repair (M‐REP) IPFRR mechanism, which provides an advanced fast reroute technique for Internet service providers‘ (ISP‘s) core networks. The M‐REP IPFRR mechanism is based on IP multicast and utilizes Protocol Independent Multicast – Dense Mode (PIM‐DM) with the modification of an internal reverse path forwarding (RPF) check. The M‐REP IPFRR mechanism does not depend on any particular routing protocol type (distance‐vector or link‐state) and requires less system resources because of its precomputation‐less character but still provides immediate reaction to failure.

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The interest in integrating real-time and realtime multimedia communication features into IP networks and its applications, the web including, started a development effort, which brings several new communication technologies. These technologies are influencing many aspects of our lives now, education not excluding. The Session Initiation Protocol (SIP) and the Web Real-Time Communication (WebRTC) belongs among the most important. Both of them allow us, the teachers, communicate with our students through audio and video. In this paper we provide a technological overview describing the aspects of integration of these two important internet technologies, SIP with WebRTC. Both together used to easily extend any web based e-learning or collaboration environment for unlimited communication experiences. I.INTRODUCTION Real-time communication (RTC) is an important part of modern convergence networks. Interactive voice communication, video, conferencing, chat, messaging and presence are today very popular and widely used IP network services and applications. These services totally broke down barriers for global communications. Significantly have enhanced the way how we communicate, affecting many aspects of our lives, education including.The Session Initiation Protocol (SIP) is a crucial communication technology of the last decade. SIP brought new and standardized control mechanisms. This allows the transformation of IP networks into real multimedia communication platforms, which are now able to provide real-time communication and presence services. Wider acceptance of SIP by the internet community and telco/ISP providers then has started the process of network convergence. The network convergence provides us with better and seamless integration of computer communication services with communication services of legacy telecommunication networks (telephone and mobile). SIP allows to integrate and mix together different kind of communication services (internet and telco, realtime and interactive with data/non-interactive services) into a new kind of communication environment. The communication environment is usually accessible through feature rich terminals and computer applications.On the other side we have another very popular internet technology. It is, among other, very often used within elearning practice and theirs learning management systems (LMS) -the web. However, the web technology was missing and did not support a true real-time mentioned limitations and is promising bring a true and native real-time communication experience to the web, the Web Real-Time Communication (WebRTC). The WebRTC technology natively integrates multimedia communication capabilities into modern web browsers, and enables them to communicate on a peer-to-peer basis using audio, video and data [4], without a need for special purpose client, neither external plug-ins, nor extensions. The design evolution, standardization and technology integration of WebRTC are subjects of several standardization bodies, mainly the World Wide Web Consortium (W3...