ON THE EFFECT OF TOKEN BUCKET PERFORMANCE IN VOICE QUALITY OVER INTERNET

Transcription

1 ON THE EFFECT OF TOKEN BUCKET PERFORMANCE IN VOICE QUALITY OVER INTERNET F. D. Trujillo *, A. J. Yuste **, E. Casilari *, A. Díaz Estrella * and F. Sandoval * * Dpto. Tecnología Electrónica. E. T. S. I. Telecomunicación Universidad de Málaga, Campus Universitario de Teatinos, s/n Málaga (Spain) Phone: Fax: trujillo@dte.uma.es ** Dpto. de Electrónica. E. U. P. de Linares Universidad de Jaén, C/Alfonso X El Sabio, Linares (Jaén, Spain) Phone: Fax: ajyuste@ujaen.es Abstract Interest in modern Internet applications is constantly growing among service providers and potential customers. Some existing and emerging services, like voice over Internet or videoconference, require a high level of quality and impose great demands on the network. It is clear the need to maintain this quality, so it is necessary the use of protocols and architectures that support quality of service, like IntServ or DiffServ. Both schemes use the token bucket algorithm to control the packets, policing them or shaping them. We present in this paper a study about voice over Internet (Vo) and how the correct dimensioning of the token bucket can achieve a suitable voice quality: non-discarding voice packets and an adequate delay in the sequence of voice packets. Key Words Quality of service, token bucket, voice over, communication networks 1. Introduction Nowadays, the matter about voice source multiplexation is widely investigated. This issue is deeply studied and analysed in the case of data networks and environments (, Internet Protocol). On the current public Internet networks only a best-effort transport service is offered. Therefore, the network performance depends on the instantaneous load and no guarantees can be given [1] [2]. Furhtermore, the new broadband services and applications over -networks demand a certain quality, named Quality of Service (QoS). When real-time services are being deployed on networks there is a clear need for QoS; i.e., guarantees on the network performance. Telephony is an interactive real-time application which puts stringent requirements on the network. Delay and packet loss have a strong impact on the voice quality, depending on the type of codec used and on the fact whether a Packet Loss Concealment (PLC) is implemented or not [3]. Introducing QoS in -environments implies that extra functionality has to be added so as to be able to guarantee a bound on the delay and the packet loss. This extra functionality includes policers, shapers, markers, flow admission control, schedulers, etc. The IETF (Internet Engineering Task Force) has introduced two architectures for QoS, the Integrated Services (Intserv) model [4] and the Differentiated Service (DiffServ) model [5]. In the IntServ model, the resources are reserved per flow by, for example, the resource ReSerVation Protocol (RSVP) [6]. The traffic specification is carried in the RSVP PATH message and each router on the path must admit the reservation, carried in the RSVP message, so that the call is accepted. In the DiffServ scheme, the routers in the network do not have per-flow awareness. The customer has an SLS (Service Level Specification) with the provider in which the traffic profile of the aggregate traffic is specified. Both schemes use the token bucket algorithm [7]. Thus, in the RSVP PATH message and in the SLS, the traffic profile is specified in terms of token bucket parameters. Choosing the appropriate traffic descriptors is an important issue. Overdimensioning the traffic descriptors will result in a waste of resources and possibly higher costs while underdimensioning will result in packet loss (when excess traffic is dropped) and, hence, degrade voice quality. When the reservation is admitted, the traffic profile is enforced at the edge of the network. Discarding out-of-profile packets is called policing. The fraction of discarded packets is the Packet Discard Ratio (PDR).

2 In this paper, we present a study about the token bucket and the necessary numbers of tokens to obtain a suitable delay in the voice packets and, in this way, not to degrade the voice quality. The detailed description of the token bucket algorithm and its model is given in Section 2. Section 3 presents the voice traffic model and its features. The source is defined by two parameters: the call input rate and the call mean length. The transmission standars that can be used are also reviewed in this Section. In Section 4, the simulation set-up is described. Performance evaluation and results are presented in Section 5. Some conclusions and future works are drawn in Section QoS: token bucket algorithm The token bucket (TB) algorithm can be used for either policing (dropping non-conformant packets) or shaping (delaying non-conformant packets) [7]. The algorithm uses two parameters: Token rate (r) in bytes/s: the maximum sustainable bit rate. Maximum burst size (b) in bytes: the maximum size of a burst of packets that is compliant. The traffic profile, A(t), that is defined with TB parameters (r,b) equals: A ( u, t) = r ( t u) + b t > 0, u < t Non conforming packet T 0 =0 B(t o )=b; k=1; (1) Arrival of packet k at time t k T k =[r t k ]; T k =T k -T k-1 B(t k )=min[b,b(t k-1 )+ T k ) B(t k )-size of packet(k) 0 B(t k )=B(t k )- size of packet(k) Figure 1: Token bucket algorithm For bursty sources, two traffic envelopes have to be specified, one for the peak rate and one for the sustainable rate. The TB is depicted in figure 1. The parameter T k denotes the total number of tokens generated over the interval [0,t]. ΔT k is the number of tokens generated between the arrival instants t k and t k 1 of packets k and k 1. B(t k ) denotes the number of tokens in the bucket at the arrival time t k of packet k. If there are less tokens than the packet size in bytes, then the packet is discarded. The algorithm can also be explained as shown in figure 2. There is a token buffer, which is filled with tokens at rate r. When the buffer contains b tokens, all new tokens are discarded. When a packet arrives and there is at least a number of tokens in the buffer equal to the packet size in bytes, the packet is compliant. Otherwise, the packet is not compliant and it will be discarded when the algorithm is used for policing purposes. Token rate (r) Bucket depth (b) flow Packet conforming? No Drop or shape Figure 2: Token bucket model 3. The voice source model Yes Network The characteristics of a voice source have been widely studied. In case of using Voice Activity Detection (VAD), the behaviour of a voice source is like the on-off source model. According to [8], three states can be distinguished in the source activity: When the user is talking. The pauses between words and syllables (about 200 ms). The silence period: the user listens to the speaker. In accordance with the information presented in [8], the length of both periods (on and off) is defined like an exponential distribution. The duration of the on-period is seconds and the duration of the off-period is seconds [7]; hence, a source activity of 39 %. Kbps Voice data Length (ms) (bytes) G G G G G G G Table I: Voice transmission standards

3 The features of several standards of voice transmission (codecs) are presented in table I. In an network, voice is transported over Real Time Protocol (), which is encapsulated in User Datagram Protocol () for which is the network layer [9]. Either Point to Point Protocol (PPP) or synchronous optical network/synchronous digital hierarchy (SONET/SDH) can be used as the link layer. PPP adds a fixed amount of overhead per packet [10], whereas the SONET/SDH overhead per packet depends on the length of the packet. The Vo data build is illustrated in figure 3. For simplicity, in this paper, we take the structure drawn in figure 4, which allows only packets from a single voice call to be transported within a single packet, resulting in a significant amount of overhead. 1 Flag 4 PPP Voice 2 FCS Figure 4: Data structure for Vo packet 4. Simulation environment Flag There are some necessary elements in a Vo network; for example, telephones, PC adapters, telephonic hubs and gateways (connect the PSTN with ) (PSTN, Public Switched Telephone Network). In figure 5, the transmission scheme of a Vo application is presented. 5 ms 30 ms 20 ms 1 Gateway Network PSTN Gateway 5 ms 30 ms Figure 5: Transmission scheme of Vo PPP PPP Figure 3: The Vo packet The source transmits packets of the same size with packet interdeparture time, T pack. The size of the packets is determined by the bit rate of the codec, R cod, and the packetization delay, T pack. The packetization delay is the time needed to fill one packet. The packet size consists of a PPP/// header and a payload. It can be expressed as equation (2): Packet size ( R T cod + 68) = pack (2) In our study, we have chosen the packetization delay, T pack, equals to 40 ms, so that there is not an excessive overload in the network. With this value chosen and from equation (2), we obtain a packet size of 108 bytes for G.729 standard (table I). Despite this value of T pack, the header overload is a matter that can not be forgotten; even there are some methods of header packetization. The final structure of the Vo packet is shown in figure 4. In an environment like the drawn one at figure 5, the delay of the voice packets end-to-end transmitted can be analysed and divided into several steps and it must be considered in both sense of communication: The first 5 ms come from, basically, the propagation. The delay in the gateway (30 ms) is due to the codification and the packetization. The delay experienced by the packets while passing through the network (20 ms), and the variations in this packet transfer delay, named, jitter. Another 30 ms due to the gateway. And, finally, the transfer time from the telephone network operator (PSTN) to the telephone of the end customer. The gateway is an essential device network; it connects the Vo network with the PSTN or ISDN (Integrated Service Digital Network). In this paper, the gateway has been simulated as an infinite queue. The telephonic calls arrive the gateway and they are being transmitted depending on the contracted link rate. In literature, the gateway model is achieved taking into account the total number of voice sources; but in our study, the gateway

4 has been modelled like a public telephone exchange with a call input rate defined by means of a Poisson distribution with a mean rate λ. The maximum allowed delay that the voice packets can tolerate is about 30 ms. There are three kinds of Vo calls: PC-PC calls PC-Telephone calls Telephone-Telephone calls The process caused by a Vo call consists of several steps: at first, the digitalization of the voice signal by means of PCM (Pulse Code Modulation) techniques. Then, the codec analyses the generated signal and it removes echoes and silent periods and it compresses the signal. Afterwards, the communication software builds the Vo packet, just as it has been described in the previous section. Finally, the gateway transmits the voice packet through the network towards the end customer. Before reaching this customer, the packet is decoded and sent through the PSTN towards the final customer. input rates that obtain a maximum delay under 30 ms are shown in table II. Link/Token rate Input rate, λ (1/s) 128 Kbps Kbps Kbps 1.7 Table II: Maximum allowed input rate With these values in mind, it is necessary to calculate the number of tokens that the TB must generate to be used in the traffic policing. In the following figures, the parameter b is replaced for a new one normalised, b n, given by equation (3): b n = b packet size (3) The packet loss for a link rate of 128 Kbps is presented in figure 7, with an input rate, λ, of 0.2 tasks/s. As we have just described, all of the simulations make use of voice sources generated by codec G.729 (table I), with a call mean length of 60 s. 5. Performance analysis Our first simulation results are drawn in figure 6. The maximum input rate, λ, is calculated for several typical link rates. This maximum input rate must allow the voice transmission; that is to say the maximum delay in the voice packets must be equal to 30 ms. This maximum allowed delay is indicated in figure 6 by means of a horizontal line. Figure 7: Packet loss (Link rate: 128, λ = 0.2 tasks/s) 128 Kbps 256 Kbps As we can see in figure 7, the parameter b n must be equal to 4, because with a value below 4, there are some packet losses. The figure 8 shows the same simulation (link rate of 128 kbps) with λ = 1 task/s (notice that for this value of λ, the delay introduced by the gateway is higher than 30 ms, as you can see in table II). 512 Kbps Figure 6: Maximum input rate calculation Figure 6 shows the lower the link rate is, the lower the allowed input rate is. In the other hand, the higher the link rate is, the higher the allowed input rate is. The maximum According to the results drawn in figure 8, with the same value of b n, 4, the packet discard ratio is very significant (about 5 %). Moreover, the voice packets would bear a very high delay. Therefore, when the thresholds (table II), determined by the chosen parameters, are exceeded, as we could imagine, there are important losses of packets; and this issue damages all the calls on line. To prove this question, figure 9 measures the packet discard ratio, in function of λ, with b n = 4 and a link rate of 128 Kbps.

5 6. Conclusions In this paper, the maximum allowed input rate for several link rates has been calculated in case of Vo sources. We have been demonstrated that the allowed input rate will be different depending on the parameter b (number of tokens) of the TB; so the parameter of the TB will determine the maximum allowed input rate. This parameter is very important for the service providers and network operators in order to guarantee the QoS to the customers, depending on the input rate the customers contract. Figure 8: Overdimensioning the input call rate (λ = 1 task/s) When the calculated thresholds are exceeded, the packet losses and the delay experienced by voice packets grows excessively and this increase will cause the network performance and QoS to degrade. As future works, the features of the network elements must be studied. Moreover, the parameters of the TB must be calculated in case of several call lengths and dealing with combining several codecs for the same voice source. Finally, another types of traffic sources must be also studied, like videoconference or multiplexation of voice sources in the same packet. Acknowledgements This work has been partially supported by the Spanish Comisión Interministerial de Ciencia y Tecnología (CICYT), project No. TEL Figure 9: Packet discard ratio for several λ We can notice in figure 9 that values of λ higher than the optimum value (0 2 tasks/s in the case of 128 Kbps link rate) lead to a packet loss growing dramatically. Finally, table III shows the value of allowed number of tokens, b n, depending on the link rate (for 256 Kbps and 512 Kbps) and the maximum input rate, λ. Token rate Maximum input rate b n 128 Kbps Kbps Kbps Tabla III: Number of allowed tokens for several link rates Obviously, the higher the token rate is, the higher number of established calls is; and, of course, the number of allowed tokens must increase. References [1] J. Gozdecki, A. Jajszczyk and R. Stankiewicz, Quality of Service Terminology in Networks, IEEE Communications Magazine, 41(3), 2003, [2] X. Xiao and L. Ni, Internet QoS: A Big Picture, IEEE Network, 13(2), 1999, [3] J. Janssen, D. De Vleeschauwer and G. H. Petit, Delay and Distortion Bounds for Packetized Voice Calls of Traditional PSTN Quality, Proc. 1st -Telephony workshop (TEL 2000), GMD report 95, Berlin (Germany), April 2000, [4] R. Braden, D. Clark and S. Shenker, Integrated Services in the Internet Architecture: an Overview, IETF Request for Comment 1633, June [5] S. Blake, D. Black, M. Carlson, E. Davies, Z. Wang and W. Weiss, An Architecture for Differentiated Service, IETF Request for Comment 2475, December 1998.

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