“…The second limitation of TFRC mechanisms is that they are originally designed for applications that use fixed packet size, and vary their sending rate in packets per second in response to congestion. Hence, they should not be used for applications that vary their packet size instead of their packet rate in response to congestion [12]. Varying the packet size during the time interval between two estimations of the sending rate distorts packet-based measurement of loss event.…”
Section: Rate and Loss Control For Multimedia Applicationsmentioning
Abstract. In this paper, we present an adaptive architecture for the transport of VoIP traffic over heterogeneous wired/wireless Internet environments. This architecture uses a VoIP gateway associated with an 802.11e QoS enhanced access point (QAP) to transcode voice flows before their transmissions over the wireless channel. The instantaneous bit rate is determined by a control mechanism based on the estimation of channel congestion state. Our mechanism dynamically adapts audio codec bit rate using a congestion avoidance technique so as to preserve acceptable levels of quality. A case study presenting the results relative to an adaptive system transmitting at bit rates typical of G.711 PCM (64 kbit/s) and G.726 ADPCM (40, 32, 24 and 16 kbit/s) speech coding standards illustrates the performance of the proposed framework. We perform extensive simulations to compare the performance between our adaptive audio rate control and TFRC mechanism. The results show that the proposed mechanism achieves better voice transmission performance, especially when the number of stations is fairly large.
“…The second limitation of TFRC mechanisms is that they are originally designed for applications that use fixed packet size, and vary their sending rate in packets per second in response to congestion. Hence, they should not be used for applications that vary their packet size instead of their packet rate in response to congestion [12]. Varying the packet size during the time interval between two estimations of the sending rate distorts packet-based measurement of loss event.…”
Section: Rate and Loss Control For Multimedia Applicationsmentioning
Abstract. In this paper, we present an adaptive architecture for the transport of VoIP traffic over heterogeneous wired/wireless Internet environments. This architecture uses a VoIP gateway associated with an 802.11e QoS enhanced access point (QAP) to transcode voice flows before their transmissions over the wireless channel. The instantaneous bit rate is determined by a control mechanism based on the estimation of channel congestion state. Our mechanism dynamically adapts audio codec bit rate using a congestion avoidance technique so as to preserve acceptable levels of quality. A case study presenting the results relative to an adaptive system transmitting at bit rates typical of G.711 PCM (64 kbit/s) and G.726 ADPCM (40, 32, 24 and 16 kbit/s) speech coding standards illustrates the performance of the proposed framework. We perform extensive simulations to compare the performance between our adaptive audio rate control and TFRC mechanism. The results show that the proposed mechanism achieves better voice transmission performance, especially when the number of stations is fairly large.
“…TFRC algorithm was first specified in RFC 3448 [22]. S. Floyd, E. Kohler [23] proposed TFRC-SP (Small-Packet TFRC) a variant of TFRC which supports fixed sending rate by using variable sized small packets.…”
To accomplish increasing real time requirements, user applications have to send different kinds of data with different speeds over the internet. To effectuate the aims of the computer networks, several protocols have been added to TCP/IP protocol suite. Transport layer has to implement emerging techniques to transfer huge amount of data like multimedia streaming. To transmit multimedia applications, one of the suitable congestion control mechanisms in transport layer is TCP Friendly Rate Control Protocol (TFRC). It controls congestion based on its equation. To get more smoothed throughput, intermediate nodes (like Routers. etc.) have to use suitable procedures in all real time situations. To eradicate the level of congestion in the network, we introduce enhanced Holt-Winters equations to RED queue management algorithm and applied to TFRC. The simulation results have shown that this strategy reduces packet loss and increases throughput.
“…With the advent of audiovisual streaming on the Internet, congestion control mechanisms, such as TCP friendly rate control (TFRC) [8], will be mandatory for multimedia traffic. Since these algorithms emulate the behavior of TCP during the start-up phase, performance degradation similar to that of TCP is observed on DAMA links for these protocols.…”
SUMMARYOne of the aims of the SatNEx (Satellite Network of Excellence, http://www.satnex.de) European project is to provide the scientific community that operates on satellite communications with measurement campaign results and evaluation tools, in order to support research activities that investigate problems in satellite communications at different levels of the OSI stack. In fact, the research activities developed in SatNEx address the most relevant aspects of satellite communications, such as channel modulation and coding schemes, access strategies, advanced networking, end-to-end quality of service, transport performance, delay tolerant networks, and new applications. Even if SatNEx is mainly devoted to satellite communications, integration of satellite and terrestrial wireless segments has also been taken into account. This paper presents the results obtained in the specific research activity devoted to trials and testbeds developed during the SatNEx project. The high number of authors of this paper is an indication of the strong collaborative work sustained, the numerous measurement campaigns, and the complexity in developing integrated testbeds.
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