Providing QoS for VoIP

Teresa Tung
(Professor Jean Walrand)

There is great importance in developing networks that provide some guarantee of QoS in order to accommodate real-time (EF) applications such as VoIP and video. These latency sensitive applications are not worthwhile to use over a network if the provided quality is poor. For instance, telephony type traffic can be carried over the Internet in the form of VoIP, but it can also be carried over a public switched telephone network at a guaranteed quality. In order for VoIP to be a competitive alternative there needs to be similar guarantees in expected quality. Poor quality of these real-time applications in terms of delay is a result of congestion in the network.

My research investigates a scheme that provides QoS for such EF traffic by means of adding packet marking capability within the router. Within a network, one can have certain routers that have the capability of marking a packet at the onset of congestion. By means of such packet marking, network resources can be shared by both EF flows and BE flows. The EF flows are governed by admission control in response to the number of packets marked by the network. As such, an EF flow will only be admitted if the routers the flow will use have the available resources to provide the QoS that the EF flow needs. A BE flow is governed by TCP with ECN wherein a marked packet is treated as a dropped packet. As such, a BE flow will not saturate the network with its transmission, leaving adequate resources to service the EF flows.

Past work includes theortical analysis and simulation results via ns. Current work lies in creating an implementation using laptops as wireless routers wherein more extensive tests can be run on a small network of such routers.

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