RTP control protocol (RTCP) extended report (XR) block for independent reporting of burst/fgp discard metrics

This document defines an RTP Control Protocol (RTCP) Extended Report (XR) block that allows the reporting of burst/gap discard metrics independently of the burst/gap loss metrics for use in a range of RTP applications

New security and control protocol for VoIP based on steganography and digital watermarking

In this paper new security and control protocol for Voice over Internet Protocol (VoIP) service is presented. It is the alternative for the IETF's (Internet Engineering Task Force) RTCP (Real-Time Control Protocol) for real-time application's traffic. Additionally this solution offers authentication and integrity, it is capable of exchanging and verifying QoS and security parameters. It is based on digital watermarking and steganography that is why it does not consume additional bandwidth and the data transmitted is inseparably bound to the voice content.Comment: 8 pages, 4 figures, 1 tabl

Error Probability in Redundant Packet Sending over IP Network

In this paper we calculate error probability of packetized signal when method of redundant packet sending is used in IP network. The number of repeated signaling packets from each interval of packetization is determined to achieve the desired error probability. The method for management of this number of repetitions is developed based on the new analysis. This method is especially important in the case of sending signaling criteria of classic telephony network over IP network, because it makes possible to reach the same error probability as in classic telephony network

Calidad de Experiencia en servicios multimedia sobre IP

En este trabajo abordamos un esquema de medida de calidad de experiencia para servicios multimedia sobre IP. Esta arquitectura, denominada QuEM (Qualitative Experience Measure), es más adecuada que los esquemas tipo MOS para la monitorización de un gran volumen de tráfico multimedia en tiempo real: facilita la agregación de resultados y la interpretación de las medidas por los operadores. La arquitectura se basa en la detección y caracterización de eventos que degraden la calidad de experiencia durante un tiempo determinado y que puedan ser descritos de forma cualitativa. Cada tipo de evento es monitorizado por un detector específico denominado QuID (Qualitative Impairment Detector). En el artículo desarrollamos la arquitectura QuEM y proponemos un conjunto de QuIDs adecuado para la monitorización de servicios como IPTV o videoconferencia

Multimedia congestion control: circuit breakers for unicast RTP sessions

The Real-time Transport Protocol (RTP) is widely used in telephony, video conferencing, and telepresence applications. Such applications are often run on best-effort UDP/IP networks. If congestion control is not implemented in these applications, then network congestion can lead to uncontrolled packet loss and a resulting deterioration of the user's multimedia experience. The congestion control algorithm acts as a safety measure by stopping RTP flows from using excessive resources and protecting the network from overload. At the time of this writing, however, while there are several proprietary solutions, there is no standard algorithm for congestion control of interactive RTP flows. This document does not propose a congestion control algorithm. It instead defines a minimal set of RTP circuit breakers: conditions under which an RTP sender needs to stop transmitting media data to protect the network from excessive congestion. It is expected that, in the absence of long-lived excessive congestion, RTP applications running on best-effort IP networks will be able to operate without triggering these circuit breakers. To avoid triggering the RTP circuit breaker, any Standards Track congestion control algorithms defined for RTP will need to operate within the envelope set by these RTP circuit breaker algorithms

Covert Channel for Improving VoIP Security

Abstract. In this paper a new way of exchanging data for Voice over Internet Protocol (VoIP) service is presented. With use of audio watermarking and network steganography techniques we achieve a covert channel which can be used for different purposes e.g. to improve IP Telephony signaling protocol's security or to alternate existing protocols like RTCP (Real-Time Control Protocol). In this paper we focus on improving VoIP security. The main advantage of this solution is that it is lightweight (it does not consume any transmission bandwidth) and the data sent is inseparably bound to the voice content