On the quality of VoIP with DCCP for satellite communications

We present experimental results for the performance of selected voice codecs using DCCP with CCID4 congestion control over a satellite link. We evaluate the performance of both constant and variable data rate speech codecs for a number of simultaneous calls using the ITU E-model. We analyse the sources of packet losses and additionally analyse the effect of jitter which is one of the crucial parameters contributing to VoIP quality and has, to the best of our knowledge, not been considered previously in the published DCCP performance results. We propose modifications to the CCID4 algorithm and demonstrate how these improve the VoIP performance, without the need for additional link information other than what is already monitored by CCID4. We also demonstrate the fairness of the proposed modifications to other flows. Although the recently adopted changes to TFRC specification alleviate some of the performance issues for VoIP on satellite links, we argue that the characteristics of commercial satellite links necessitate consideration of further improvements. We identify the additional benefit of DCCP when used in VoIP admission control mechanisms and draw conclusions about the advantages and disadvantages of the proposed DCCP/CCID4 congestion control mechanism for use with VoIP applications

Performance of VoIP with DCCP for satellite links

We present experimental results for the performance of selected voice codecs using the Datagram Congestion Control Protocol (DCCP) with TCP-Friendly Rate Control (TFRC) congestion control mechanism over a satellite link. We evaluate the performance of both constant and variable data rate speech codecs (G.729, G.711 and Speex) for a number of simultaneous calls, using the ITU E-model and identify problem areas and potential for improvement. Our experiments are done on a commercial satellite service using a data stream generated by a VoIP application, configured with selected voice codecs and using the DCCP/CCID4 Linux implementation. We analyse the sources of packet losses which are a main contributor to reduced voice quality when using CCID4 and additionally analyse the effect of jitter which is one of the crucial parameters contributing to VoIP quality and has, to the best of our knowledge, not been considered previously in the published DCCP performance results. We propose modifications to the CCID4 algorithm and demonstrate how these improve the VoIP performance, without the need for additional link information other than what is already monitored by CCID4 (which is the case for Quick-Start). We also demonstrate the fairness of the proposed modifications to other flows. We identify the additional benefit of DCCP when used in VoIP admission control mechanisms and draw conclusions about the advantages and disadvantages of the proposed DCCP/ CCID4 congestion control mechanism for use with VoIP applications

Performance modelling of a multiple threshold RED mechanism for bursty and correlated Internet traffic with MMPP arrival process

Access to the large web content hosted all over the world by users of the Internet engage many hosts, routers/switches and faster links. They challenge the internet backbone to operate at its capacity to assure e±cient content access. This may result in congestion and raises concerns over various Quality of Service (QoS) issues like high delays, high packet loss and low throughput of the system for various Internet applications. Thus, there is a need to develop effective congestion control mechanisms in order to meet various Quality of Service (QoS) related performance parameters. In this paper, our emphasis is on the Active Queue Management (AQM) mechanisms, particularly Random Early Detection (RED). We propose a threshold based novel analytical model based on standard RED mechanism. Various numerical examples are presented for Internet traffic scenarios containing both the burstiness and correlation properties of the network traffic

VoIP: Making Secure Calls and Maintaining High Call Quality

Modern multimedia communication tools must have high security, high availability and high quality of service (QoS). Any security implementation will directly impact on QoS. This paper will investigate how end-to-end security impacts on QoS in Voice over Internet Protocol (VoIP). The QoS is measured in terms of lost packet ratio, latency and jitter using different encryption algorithms, no security and just the use of IP firewalls in Local and Wide Area Networks (LAN and WAN). The results of laboratory tests indicate that the impact on the overall performance of VoIP depends upon the bandwidth availability and encryption algorithm used. The implementation of any encryption algorithm in low bandwidth environments degrades the voice quality due to increased loss packets and packet latency, but as bandwidth increases encrypted VoIP calls provided better service compared to an unsecured environment.Les eines modernes de comunicació multimèdia han de tenir alta seguretat, alta disponibilitat i alta qualitat de servei (QoS). Cap tipus d¿implementació de seguretat tindrà un impacte directe en la qualitat de servei. En aquest article s¿investiga com la seguretat d'extrem a extrem impacta en la qualitat de servei de veu sobre el Protocol d'Internet (VoIP). La qualitat de servei es mesura en termes de pèrdua de proporció de paquets, latència i jitter utilitzant diferents algoritmes d¿encriptació, sense seguretat i només amb l'ús de tallafocs IP en local i en xarxes d'àrea àmplia (LAN i WAN). Els resultats de les proves de laboratori indiquen que l'impacte general sobre el rendiment de VoIP depèn de la disponibilitat d'ample de banda i l'algorisme de xifrat que s'utilitza. La implementació de qualsevol algorisme de xifrat en entorns de baix ample de banda degrada la veu a causa de l'augment de la pèrdua de paquets i latència dels paquets de qualitat, però quan l'ample de banda augmenta les trucades de VoIP xifrades proporcionen un millor servei en comparació amb un entorn sense seguretat.Las herramientas modernas de comunicación multimedia deben tener alta seguridad, alta disponibilidad y alta calidad de servicio (QoS). Ningún tipo de implementación de seguridad tendrá un impacto directo en la calidad de servicio. En este artículo se investiga como la seguridad de extremo a extremo impacta en la calidad de servicio de voz sobre el Protocolo de Internet (VoIP). La calidad de servicio se mide en términos de pérdida de proporción de paquetes, latencia y jitter utilizando diferentes algoritmos de encriptación, sin seguridad y sólo con el uso de cortafuegos IP en local y en redes de área amplia (LAN y WAN). Los resultados de las pruebas de laboratorio indican que el impacto general sobre el rendimiento de VoIP depende de la disponibilidad de ancho de banda y el algoritmo de cifrado que se utiliza. La implementación de cualquier algoritmo de cifrado en entornos de bajo ancho de banda degrada la voz debido al aumento de la pérdida de paquetes y latencia de los paquetes de calidad, pero cuando el ancho de banda aumenta las llamadas de VoIP cifradas proporcionan un mejor servicio en comparación con un entorno sin seguridad

Evaluation Study for Delay and Link Utilization with the New-Additive Increase Multiplicative Decrease Congestion Avoidance and Control Algorithm

As the Internet becomes increasingly heterogeneous, the issue of congestion avoidance and control becomes ever more important. And the queue length, end-to-end delays and link utilization is some of the important things in term of congestion avoidance and control mechanisms. In this work we continue to study the performances of the New-AIMD (Additive Increase Multiplicative Decrease) mechanism as one of the core protocols for TCP congestion avoidance and control algorithm, we want to evaluate the effect of using the AIMD algorithm after developing it to find a new approach, as we called it the New-AIMD algorithm to measure the Queue length, delay and bottleneck link utilization, and use the NCTUns simulator to get the results after make the modification for the mechanism. And we will use the Droptail mechanism as the active queue management mechanism (AQM) in the bottleneck router. After implementation of our new approach with different number of flows, we expect the delay will less when we measure the delay dependent on the throughput for all the system, and also we expect to get end-to-end delay less. And we will measure the second type of delay a (queuing delay), as we shown in the figure 1 bellow. Also we will measure the bottleneck link utilization, and we expect to get high utilization for bottleneck link with using this mechanism, and avoid the collisions in the link