116 research outputs found

    Rtp and the datagram congestion control protocol

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    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

    Options for Securing RTP Sessions

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    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

    Media Processing in Video Conferences for Cooperating Over the Top and Operator Based Networks

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    Telecom operators have dominated the communication industry for a long time by providing services with guaranteed quality of service. Such services are provided by the operator at the cost of maintaining a high grade network. With the introduction of broadband and internet, many over the top (OTT) services have emerged. These services use the underlying operator networks as a mere bit pipe while all service intelligence resides in the application running on the client device. Introduction of OTT services has seen a good response from general users who are no longer bound to services provided by the network operator. This in turn has caused operators and telecom companies to loose the ownership of their customers. This thesis takes media processing in video conferencing as a case study to compare the two competing domains of operator networks and OTT networks. Both domains offer video conferencing to end users, but they follow different architectures. The study shows that OTT services can perform much better if they utilize support of the underlying network. This will also bring the user base back to the network operator. The proposal is to turn the competition into cooperation between both parties. Assessments are done from both technical as well as business perspectives to assert that such cooperative agreements are possible and should be experimented in real life

    Service Oriented Architecture for VoIP Conferencing

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    Voice/Video over IP (VoIP) systems to date have been either highly centralized or dependent on the IP multicast in nature. Global Multimedia Collaboration System is a scalable, integrated and service-oriented VoIP conferencing system, based on the XGSP collaboration framework and NaradaBrokering messaging middleware. This system can provide media and session services to heterogeneous endpoints such as H.323, SIP, Access Grid, RealPlayer as well as cellular phone. In this paper, we address the challenges of scalability, interoperablity and heterogeneity in massive VoIP conferencing system. We believe that our approach opens up new opportunities for leveraging classic VoIP systems by using new technologies in service-oriented computing

    Signaling For Multimedia Conferencing in Stand-Alone Mobile Ad Hoc Networks

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    Mobile ad hoc networks (MANETs) are infrastructure-less and can be set up anywhere, anytime. They can host a wide range of applications in rescue operations, military, private, and commercial settings. Multimedia conferencing is the basis of a wealth of “killer†applications that can be deployed in MANETs. Some examples are audio/video conferencing, multiplayer games, and online public debating. Signaling is the nerve center of multimedia conferences—it establishes, modifies, and tears down conferences. This paper focuses on signaling for multimedia conferences in MANETs. We review the state of the art and propose a novel architecture based on application-level clusters. Our validation employed SIP as the implementation technology and OPNET as our simulation tool. Our clusters are constructed dynamically and the nodes that act as cluster heads are elected based on their capabilities. The capabilities are published and discovered using a simple application-level protocol. The architectural principles and the clustering operations are discussed. Our SIP-based implementation is also presented along with the performance evaluation. Keywords: MANET, SIP-technology, OPNET-simulation tool, cluste

    Options for Securing RTP Sessions

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    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

    On the Design of a SIP-Based Binding Middleware for Next Generation Home Network Services

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    Abstract. This paper proposes a two-layer component-based middleware frame-work that copes with the complexity of managing and constructing efficient and useful SIP-based home services. In the first layer, the device integration frame-work overcomes the heterogeneity of media home devices by providing protocol-independent components that reify the underlying devices. At the second layer, the binding framework allows constructing open mobile media bindings between SIP and non SIP communication protocol endpoints including media home de-vices. The openness of our framework is motivated by the need of constructing highly flexible home services such as context aware adaptation, session mobility, media session enrichment and QoS. Our framework is implemented as part of a context-aware adaptive middleware on top of the OSGi platform and an illustra-tive use case is shown.

    Design, implementation and evaluation of unified communications on-premises and over the Cloud

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    Unified Communication (UC) is the integration of two or more real time communication systems into one platform. Integrating core communication systems into one overall enterprise level system delivers more than just cost saving. These real-time interactive communication services and applications over Internet Protocol (IP) have become critical in boosting employee accessibility and efficiency, improving customer support and fostering business agility. However, some small and medium-sized businesses (SMBs) are far from implementing this solution due to the high cost of initial deployment and ongoing support. Cloud based UC solution, UC as a Service (UCaaS), is now itself a maturing technology in the marketplace and it has revolutionized the IT industry, being the powerful platform that many businesses are choosing to migrate their on-premises UC solution onto. UCaaS solution has the potential to reduce the capital and operational expenses associated with deploying UC on their own. In this paper, we will discuss and demonstrate an open source on-premises UC solution, viz. “Asterisk” for use by businesses, and report on some performance tests using SIPp. This paper also discusses and demonstrates an open source UCaaS solution. The contribution from this research is the provision of technical advice to businesses in deploying UC and UCaaS, which is manageable in terms of cost, ease of deployment and support

    INTENT BASED LOAD-BALANCING FOR VOICE OVER INTERNET PROTOCOL (VOIP) ELEMENTS

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    Presented herein is an intelligent call distribution/load balancing solution that performs distribution based on the type of call. The solution determines the type of call based on various factors and uses reinforced learning algorithms to select the element best suited for that type of call based on call-success-ratio for a particular call type (e.g., audio/video/fax/ etc.). This eliminates call failures, call delays and improves customer satisfaction. This solution can be extended to various other details of the call like dual tone multiple frequency (DTMF), codec, payload type, etc
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