267 research outputs found

    Options for Securing RTP Sessions

    Get PDF
    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

    Congestion Control using FEC for Conversational Multimedia Communication

    Full text link
    In this paper, we propose a new rate control algorithm for conversational multimedia flows. In our approach, along with Real-time Transport Protocol (RTP) media packets, we propose sending redundant packets to probe for available bandwidth. These redundant packets are Forward Error Correction (FEC) encoded RTP packets. A straightforward interpretation is that if no losses occur, the sender can increase the sending rate to include the FEC bit rate, and in the case of losses due to congestion the redundant packets help in recovering the lost packets. We also show that by varying the FEC bit rate, the sender is able to conservatively or aggressively probe for available bandwidth. We evaluate our FEC-based Rate Adaptation (FBRA) algorithm in a network simulator and in the real-world and compare it to other congestion control algorithms

    Security aspects in voice over IP systems

    Get PDF
    Security has become a major concern with the rapid growth of interest in the internet. This project deals with the security aspects of VoIP systems. Various supporting protocols and technologies are considered to provide solutions to the security problems. This project stresses on the underlying VoIP protocols like Session Initiation Protocol (SIP), Secure Real-time Transport Procotol (SRTP), H.323 and Media Gateway Control Protocol (MGCP). The project further discusses the Network Address Translation (NAT) devices and firewalls that perform NAT. A firewall provides a point of defense between two networks. This project considers issues regarding the firewalls and the problems faced in using firewalls for VoIP; it further discusses the solutions about how firewalls can be used in a more secured way and how they provide security

    Options for Securing RTP Sessions

    Get PDF
    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

    AdamRTP: Adaptive multi-flow real-time multimedia transport protocol for Wireless Sensor Networks

    Get PDF
    Real-time multimedia applications are time sensitive and require extra resources from the network, e.g. large bandwidth and big memory. However, Wireless Sensor Networks (WSNs) suffer from limited resources such as computational, storage, and bandwidth capabilities. Therefore, sending real-time multimedia applications over WSNs can be very challenging. For this reason, we propose an Adaptive Multi-flow Real-time Multimedia Transport Protocol (AdamRTP) that has the ability to ease the process of transmitting real-time multimedia over WSNs by splitting the multimedia source stream into smaller independent flows using an MDC-aware encoder, then sending each flow to the destination using joint/disjoint path. AdamRTP uses dynamic adaptation techniques, e.g. number of flows and rate adaptation. Simulations experiments demonstrate that AdamRTP enhances the Quality of Service (QoS) of transmission. Also, we showed that in an ideal WSN, using multi-flows consumes less power than using a single flow and extends the life-time of the network

    Cloud-gaming:Analysis of Google Stadia traffic

    Full text link
    Interactive, real-time, and high-quality cloud video games pose a serious challenge to the Internet due to simultaneous high-throughput and low round trip delay requirements. In this paper, we investigate the traffic characteristics of Stadia, the cloud-gaming solution from Google, which is likely to become one of the dominant players in the gaming sector. To do that, we design several experiments, and perform an extensive traffic measurement campaign to obtain all required data. Our first goal is to gather a deep understanding of Stadia traffic characteristics by identifying the different protocols involved for both signalling and video/audio contents, the traffic generation patterns, and the packet size and inter-packet time probability distributions. Then, our second goal is to understand how different Stadia games and configurations, such as the video codec and the video resolution selected, impact on the characteristics of the generated traffic. Finally, we aim to evaluate the ability of Stadia to adapt to different link capacity conditions, including those cases where the capacity drops suddenly. Our results and findings, besides illustrating the characteristics of Stadia traffic, are also valuable for planning and dimensioning future networks, as well as for designing new resource management strategies

    Media Transport and Use of RTP in WebRTC

    Get PDF
    The framework for Web Real-Time Communication (WebRTC) provides support for direct interactive rich communication using audio, video, text, collaboration, games, etc. between two peers' web browsers. This memo describes the media transport aspects of the WebRTC framework. It specifies how the Real-time Transport Protocol (RTP) is used in the WebRTC context and gives requirements for which RTP features, profiles, and extensions need to be supported
    • …
    corecore