IP Telephony Performance Evaluation and Simulation

IP telephony presents some big advantages over traditional circuit-switched telephony especially in the sense of increased utilization of expensive backbones and significantly reduced cost. However, in order to win the competition, IP telephony must provide the same quality of voice service as the traditional telephony, which has been proven as a major challenge. In today’s VoIP market, multiple vendors have developed different IP telephony products based on some well-known IP telephony Standards such as ITU-T H.323 and IETF Session Initiation Protocol (SIP). In Part I of this paper, we present some metrics to help evaluate IP Telephony Performance, discuss a new architecture proposed by Pawan and etc.[1], Distributed Open Signaling Architecture (DOSA), as an enhancement to H.323 and SIP. Then in Part II, we conduct a simulation experiment to demonstrate the backbone-scheduling requirements for this robust IP telephony architecture. Part I: Critical Paper Review and Summary Introduction High-quality telephony service must meet the stringent bounds on end-to-end voice signal delay, jitter and loss. This requires adequate capacity allocation along the end-to-end path for a voice flow. This requirement is met quite well in traditional telephony in PSTN by using per-flow hop-by-hop signaling and providing service guarantee with a temporary dedicated circuit. In IP Telephony, however, in order to improve the system utilization efficiency, no dedicated circuit will be associated with each voice flow. Instead, all voice flows are statistically multiplexed together with all other data and video traffics in every hop of Internet. This makes a VoIP call much cheaper than a PSTN phone call. However, how to provide comparable voice service quality as the PSTN phone poses a major challenge for IP Telephony designer. Since originally IP provides only best-effort service for all traffics, unavoidably, this makes the end-to-end delay, jitter, and packet loss for voice traffic vary dramatically between different Internet traffic situations. In order to solve these problems, tremendous research effort has been given to VoIP architecture design, which also gave the birth of two well-known IP telephony Standards: H.323 and SIP. Although IP Telephony has come to people’s life for several years, there is still much to be improved. Even in the two IP telephony standards, no specific and optimized end-to-end Quality of Service (QoS) Management Module has been finalized and implemented. In this paper, we will discuss a new architecture proposed by Pawan and etc. [1], Distributed Open Signaling Architecture (DOSA), as an enhancement to current IP standards. Before the discussion of detailed architecture, let’s see how IP telephony performance is measured. Performance Measure Metrics 1.End-to-END Delay Metrics. ITU-T has carried out an extensive testing over three decades, and then gave the Recommendation G.114 to provide guidelines on the tolerable delay for a normal telephone conversation. According to the recommendation, the maximum one-way delay acceptable for most applications is 150ms. We can use a delay budget to allocate this 150-ms one-way delay limit among several different sources. The biggest delay component is backbone propagation delay. The worst-case one-way delay in the continental United States has been measured at 95ms. This leaves a remaining delay budget of only 55ms in one-way for all other sources. As a general example, we can allocate 10ms to the speech coder, 10ms to speech enhancement and silence suppression, and 5 ms to processing, propagation, and interleaving in the local area network, and 15 ms to play-out buffer at the receiver. Finally, we have 15 ms budget for jitter tolerate during the whole transit. Among this 15 ms, 5ms can be given to the local area network, and this leaves 10ms for queuing delay in the backbone network. 2. Packet Loss Metrics. Packet Loss requirement for encoded speech is typically 1 percent or less. Although we can use loss-concealment algorithms to reproduce intelligible speech even with higher loss rates, this is not desired if we want it work as a circuit-switched replacement. 3. Post-Dial Delay Metrics. We want the delay between the user dialing the last digit and receiving positive confirmation from the network not to be perceptibly different from post-dial delay in the circuit-switched telephony. 4. Post-Pickup Delay Metrics. We want the delay between a user picking up a ringing phone and the voice path being cut through to be short enough that the initial “hello” is not clipped. Distributed Open Signaling Architecture (DOSA) Signaling architecture and resource management framework are two important components of the whole IP telephony system. Since user expectations of quality impose stringent requirements on both network and signaling performance, we believe that network-layer QoS guarantee is essential to robust IP telephony service. As a result, the network must support resource reservations and admission control, and the signaling architecture must ensure the authorization and correct charge for the resource usage. Figure 1 shows all the key components in the DOSA architecture. From this diagram we can see there are totally four key components in the system: Customer Premises Equipment (CPE), Edge Router, Gate Controller, and PSTN Gateway. Every conventional telephone or multimedia PC accesses a local access network through a Customer Premises Equipment, which may be a multimedia terminal adaptor. Every local access network accesses a Managed IP backbone through an Edge Router. An important management component in IP telephony system is the Gate Controller. If an IP telephone wants to talk to a PSTN telephone, a PSTN Gateway can provide the interface between the IP network and PSTN. Figure 1: Distributed Open Signaling Architecture [1] For robust telephony service, a service provider must manage access to the network-layer resources, and the Edge Routers are in charge of Admission Control at the boundary to the backbone network. ERs implement a function called a gate to control the access. Once a gate is opened, the ER starts to provide enhanced QoS service. However, before opening a gate, the ER requires authorization CPE ER pc telephone Gate Controller PSTN Gateway Gate Controller ER CPE Managed IP Backbone