A WebRTC Video Chat Implementation Within the Yioop Search Engine

Web real-time communication (abbreviated as WebRTC) is one of the latest Web application technologies that allows voice, video, and data to work collectively in a browser without a need for third-party plugins or proprietary software installation. When two browsers from different locations communicate with each other, they must know how to locate each other, bypass security and firewall protections, and transmit all multimedia communications in real time. This project not only illustrates how WebRTC technology works but also walks through a real example of video chat-style application. The application communicates between two remote users using WebSocket and the data encryption algorithm specified in WebRTC technology. This project concludes with a description of the WebRTC video chat application’s implementation in Yioop.com, a PHP-based internet search engine

Delivering Live Multimedia Streams to Mobile Hosts in a Wireless Internet with Multiple Content Aggregators

We consider the distribution of channels of live multimedia content (e.g., radio or TV broadcasts) via multiple content aggregators. In our work, an aggregator receives channels from content sources and redistributes them to a potentially large number of mobile hosts. Each aggregator can offer a channel in various configurations to cater for different wireless links, mobile hosts, and user preferences. As a result, a mobile host can generally choose from different configurations of the same channel offered by multiple alternative aggregators, which may be available through different interfaces (e.g., in a hotspot). A mobile host may need to handoff to another aggregator once it receives a channel. To prevent service disruption, a mobile host may for instance need to handoff to another aggregator when it leaves the subnets that make up its current aggregator�s service area (e.g., a hotspot or a cellular network).\ud In this paper, we present the design of a system that enables (multi-homed) mobile hosts to seamlessly handoff from one aggregator to another so that they can continue to receive a channel wherever they go. We concentrate on handoffs between aggregators as a result of a mobile host crossing a subnet boundary. As part of the system, we discuss a lightweight application-level protocol that enables mobile hosts to select the aggregator that provides the �best� configuration of a channel. The protocol comes into play when a mobile host begins to receive a channel and when it crosses a subnet boundary while receiving the channel. We show how our protocol can be implemented using the standard IETF session control and description protocols SIP and SDP. The implementation combines SIP and SDP�s offer-answer model in a novel way

Options for Securing RTP Sessions

The Real-time Transport Protocol (RTP) is used in a large number of different application domains and environments. This heterogeneity implies that different security mechanisms are needed to provide services such as confidentiality, integrity, and source authentication of RTP and RTP Control Protocol (RTCP) packets suitable for the various environments. The range of solutions makes it difficult for RTP-based application developers to pick the most suitable mechanism. This document provides an overview of a number of security solutions for RTP and gives guidance for developers on how to choose the appropriate security mechanism

Rtp and the datagram congestion control protocol

We describe how the new Datagram Congestion Control Protocol (DCCP) can be used as a bearer for the Real-time Transport Protocol (RTP) to provide a congestion controlled basis for networked multimedia applications. This is a step towards deployment of congestion control for such applications, necessary to ensure the future stability of the best-effort network if high-bandwidth streaming and IPTV services are to be deployed outside of closed QoS-managed networks

Evaluating Extensions to IMS Session Setup for Multicast-based Many-to-Many Services

Telecommunication networks are converging towards an all-IP paradigm that integrates a broad set of value-added services. In this context, the IP Multimedia Subsystem (IMS) is being developed by the 3GPP as a key element to achieve the convergence. Additionally, multiparty services are nowadays acquiring an increasing interest from the industry. In this respect, network multicast provides a cost-effective solution to deliver these services to the user. Nevertheless, although network multicast is being considered as an enabler for one-to-many services (e.g. IPTV) in the IMS, the specifications for many-to-many services still follow a unicast approach (e.g. push-to-talk and conference). This paper describes extensions to the session control procedures in the IMS, to support multicast based multi-user services. The idea was first described in a prior work, but this paper presents enhancements to provide a comprehensive solution and to improve the grade of service (GOS) perceived by the users. In addition, the GOS achieved by the proposal is evaluated. First, the bandwidth utilization for the multicast-based multi-user services is analyzed and compared against the unicast scenario. Next, the GOS is evaluated using an analytical approach, by obtaining the mathematical expressions for the session and user plane setup delays. Finally, the GOS is also evaluated using an experimental approach, and the results are compared with values recommended by the ITU-T

Remote service usage through SIP with multimedia access as a use case

The IP Multimedia Subsystem is under deployment, as an IP-based service control and access infrastructure, but how it interconnects with residential appliances is currently unclear. With IMS access for the residential appliances they can be used as both service consumers and service providers. In this paper we present a protocol which allows residential services to be remotely invoked, through the IMS, and consumed in a different network, along with a prototype implementation and early results. With our protocol services of two distinct service protocol systems can cooperate

Reflections on security options for the real-time transport protocol framework

The Real-time Transport Protocol (RTP) supports a range of video conferencing, telephony, and streaming video ap- plications, but offers few native security features. We discuss the problem of securing RTP, considering the range of applications. We outline why this makes RTP a difficult protocol to secure, and describe the approach we have recently proposed in the IETF to provide security for RTP applications. This approach treats RTP as a framework with a set of extensible security building blocks, and prescribes mandatory-to-implement security at the level of different application classes, rather than at the level of the media transport protocol

TRIM: An architecture for transparent IMS-based mobility

In recent years, the development and deployment of new wired and wireless access net work technologies have made the ubiquitous Internet a reality. Users can access anywhere and anytime to the broad set of value added Internet services, which are delivered by means of the IP protocol. In this context, 3GPP is currently developing the IP Multimedia Subsystem (IMS), as a key element that allows to evolve from the ubiquitous access to the Internet services towards a next generation network model, by providing a set of essen tial facilities such as session control, QoS, charging and service integration. Nevertheless, several open issues still need consideration before the future Internet becomes real, such as supporting user mobility in IP networks. Although mobility support in the Internet is receiving much attention, IMS networks present inherent particularities that require fur ther analysis. The solutions proposed so far for IMS do not support mobility transparently to the end user applications, or address the problem by introducing complex changes to the IMS infrastructure. This paper presents TRIM, an architecture for transparent IMS based mobility. TRIM supports mobility in IMS networks transparently to the end user applications, which are unaware of the handover management procedures executed between the mobile node and the network. We have performed several experiments with a TRIM prototype, using a real IMS testbed with 3G and WLAN access networks, validating the proposal for UDP and TCP based applications.European Community's Seventh Framework ProgramPartially granted by the Madrid Community through the MEDIANET project (S 2009/TIC 1468)Publicad

A solution for transparent mobility with route optimization in the IP multimedia subsystem

This paper presents TRIM+, an architecture for transparent mobility management with route optimization in IMS based networks. The design of our architecture is based on a previous work referred to as TRIM. TRIM was originally devised to provide transparent mobility support in the IMS, although transparency came at the cost of using a suboptimal data path between communicating end points. TRIM+ maintains transparency as a design criterium, and thus end-user applications, running at the mobile node and its correspondent communication peers, are unaware of mobility management procedures. Additionally, the proposed design defines a set of route optimization procedures, allowing compliant devices to use the optimal data path for media communications. Furthermore, TRIM+ addresses packet loss management in scenarios where the media path cannot be maintained during the handover of the MN. To this end, our architecture enables the MN to request buffering capacity in its home network to temporarily store incoming media traffic during the handover, which would otherwise be dropped. This mechanism, as well as route optimization procedures, are executed transparently to the end-user applications running at the communicating end points. As a proof-of-concept, we have implemented a software prototype of the TRIM+ architecture, deploying it over a real IMS testbed. By means of a set of experiments, we have validated the mechanisms proposed in this paper, considering both UDP and TCP user traffic.This article has been partially granted by the Madrid Commu nity through the MEDIANET project (S 2009/TIC 1468), and by the European Community through the CROWD project (FP7 ICT 318115). The work of Ignacio Soto has been partially sup ported through the I MOVING project (TEC2010 18907).Publicad