This paper describes migration strategies in optical networks. It is expected that the conversion of current transport networks (IP/MPLS over SDH) into more packet oriented (IP/MPLS over Ethernet) technologies will result in significant cost savings for network operators. The state space of a migrating network can be modeled as a directed graph. To find the optimal migration path within this state oriented graph, several approaches, such as the Incremental Migration and the All-Period Migration can be used. Current research results regarding cost development and resource utilization in layer 1 and layer 3 are presented in this work. The effects of several planning techniques within such uncertain environment, like routing and grooming, on the total cost development are discussed, too.
PIM-SM (Protocol Independent Multicast -Sparse Mode) is a multicast routing protocol that uses one central node (which we refer to with the Rendezvous Point "RP") for all transmitters in a multicast group. It builds source-specific trees to the transmitters whose data rates exceed a predefined threshold. Investigations have been done to improve and provide an efficient mechanism for switching between the shared-based mode and source-based mode in order to limit some Quality of Service (QoS) such as end-to-end delay, bandwidth, etc. In this paper, we improve the switching mechanism proposed by [FD02]. To evaluate the performance of our improvement, two PIM-SM models (without a switching mechanism and with a rate based switching mechanism) are used as a reference for comparison. Our improved switching mechanism results in a lower network load compared to the standard switching mechanism and keeps the end-to-end delay within a predefined threshold. The additional control load overload is presented.
SUMMARYThis paper presents a new numerical method to analyse steady-state TCP throughputs in DiffServ networks. Different TCP models are briefly reviewed, and a Markov chain is used to capture the behaviour of a multirandom early detection (RED) queue. The established closed network model couples models for TCP, marking device and multi-RED queue. We assess accuracy and limits of the technique, and apply it for a discusion of parameter settings for marking devices.
In recent years new challenges have emerged in the telecommunications market resulting from the increase of network traffic and strong competition. Because of that, service providers feel constrained to replace expensive and complex IP-routers with a cheap and simple solution which guarantees the requested quality of services (QoS) with low cost. One of these solutions is to use the Ethernet technology as a switching layer, which results in using the cheap Ethernet services (E-Line, E-LAN and E-Tree) and to replace the expensive IP-routers. To achieve this migration step, new algorithms that support the available as well as the future services have to be developed. In this paper, we investigate the multicast protection issue. Three multicast protection algorithms based on the shared capacity between primary and backup solutions are proposed and evaluated. The blocking probability is used to evaluate the performance of the proposed algorithms. The sub-path algorithm resulted in a low blocking probability compared with the other algorithms.
Delay is one of the most important parameters of VoIP which can significantly impact the voice quality. When crossing an ADSL downlink, a VoIP application has to contend for the bandwidth with other applications, e.g., TCP. With the assumption that the DSLAM only has a FCFS mechanism employed in the downlink direction, the VoIP packets may suffer from serious queueing delay problem in the DSLAM buffer. This problem can be solved by introducing TCP packet dropping at the receiver. But the drop ratio should be adjusted depending on the network conditions such as network load. In this paper, an adaptive packet dropping algorithm based on the RED mechanism is proposed to guarantee the VoIP quality in ADSL downlink under different network conditions. Simulations are conducted to compare our approach with the fixed drop ratio approach considering different network traffic. The results show that our algorithm can successfully adapt to the network conditions. Moreover, compared with the fixed drop ratio one, our approach can achieve better VoIP quality with fewer TCP packet dropping.
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