Narrow Results

The Stream Control Transmission Protocol (SCTP) establishes a transport layer framework for multihomed communication where the standard failover mechanism improves the transmission resilience. A delay-centric handover mechanism for SCTP have been proposed to improve the end-to-end delay of real-time multimedia transmissions. In this paper, we explore the coexistence of delay-centric handover and standard SCTP failover to identify whether they work together to select a communication path that preserves voice transmission quality. The study was performed using computer simulations, which allowed us to evaluate users' perceived quality and network stability for several conditions of traffic load and packet loss ratio. Although voice quality is generally improved, conflicting decision about interface selection resulted in performance degradation under certain conditions. The global stability of the delay-centric operation when multiple independent voice transmissions compete for the available bandwidth was also evaluated. Simulations showed that under high link utilization, instabilities due to frequent path switching may lead to lower voice quality.

This research developed a novel algorithm to evaluate internet-based services such as VoIP, Video Conferencing, HTTP and FTP, of different IEEE 802.11 technologies in order to identify the optimum network architecture among Basic Service Set (BSS), Extended Service Set (ESS) and the Independent Basic Service Set (IBSS). The proposed algorithm will yield the rank order of different IEEE 802.11 technologies. By selecting the optimum network architecture and technology, the best overall network performance that provides good voice, video and data quality is guaranteed. Furthermore, it meets the acceptance threshold values for the VoIP, Video Conferencing, HTTP and FTP quality metrics. This algorithm was applied to various room sizes ranging from $2\times 3$m to $10\times 14$m and the number of nodes ranged from 1 to 65. The spatial distributions considered were circular, uniform and random. The Quality of Service (QoS) metrics used were delay, jitter, throughput and packet loss.

Real time Internet traffic such as Voice over IP (VoIP) is difficult to estimate and simulate. Since the process is non-ergodic we must use ensemble averages rather than time averages, and for combinations of sources the simulation time can be excessive. We give some analytical results for Constant Bit Rate (CBR) traffic for the n*D/D/I queue, and for a CBR queue with varying source rates we present an expression for the Interval of Significance (IoS). This is the minimum number of time slots necessary to simulate the traffic, important because of the considerable computation time often involved in simulating this queue. Our results help in dimensioning networks for CBR services.

The purpose of this paper is to evaluate the performance of H.323 handoff methods that use ad hoc multipoint conference, transfer, and third party rerouting signaling respectively. This performance is evaluated in terms of inter-operability and signaling delay. The inter-operability specification includes multipoint conference messages, H.450.2 messages, and null TerminalCapabilitySet message. The delay of each signaling is analyzed into number of TCP connections and number of logical channel setup procedures. The results of our experiment show that third party rerouting signaling has better result than other signalings in inter-operating with terminals of four vendors, since it uses only H.323 mandatory messages. Also it requires fewer number of TCP connections and logical channel setup procedures than the others do, so it can reduce the signaling delay by 1.4 sec in average.

This paper presents a comparative analysis of call set up times, MOS (Mean Opinion Score), packet end to end delay and packet loss for the H.323 and SIP (Session Initiation Protocol) signaling schemes used in Voice Over IP (Voice over Internet Protocol) calls. We compare voice performance parameters like MOS, delay, jitter and call set up times and establish the differences in performance of these two VoIP protocols through an opnet simulation. This detailed simulation study is performed over H.323 and SIP networks where we have shown that the H.323 network performed better in terms of call set up time and packet end to end delay for the signaling messages. However, MOS for the voice packets in both the networks was expectedly the same as the same network topology was used for both the signaling protocols. There has been an influx of a large number of signaling and transport protocols over IP networks, giving rise to unacceptably large delays, most unsuitable for real time applications. SIP and H.323 are used for establishment and release of Voice aver IP calls as well as Video and Media calls. Both these protocols play a very important role in terms of optimizing the call set up time and call reliability and flexibility over IP networks for real time applications like voice and video.