A Case for Packet-Switched Telecommunication Infrastructure for Low-

Density Traffic Rural Areas Kennedy Ifeh

University of the Witwatersrand, Johannesburg, South Africa Kennedy.Ifeh@students.wits.ac.za

Abstract – This paper proposes a low cost voice over IP (VoIP) infrastructure for the provisioning of integrated telecommunication services to low density rural areas. A related circuit switched model, the Digital Concentrator Ring (DCR-300), was first designed by Telkom, South Africa, to supply plain old telephone service (POTS) in the rural areas. Subsequently, a research paper proposed the remodeling of the DCR-300 and showed that the redesigned system can sustain more voice traffic. This paper builds on the remodeled architecture by modifying it further into a packet switched platform, to suit the trend of telecommunications network evolution. Simulation result shows that the packet switched system can sustenance even more voice traffic, without negative impact on the mean service time of voice packets. Furthermore, the proposed design can guarantee both bandwidth and cost efficiency.

Keywords—DCR-300; VoIP; Teletraffic; Packet switched;

I. INTRODUCTION

The expansion and penetration of telecommunication services beyond major cities to low density areas is a challenge in Africa. The reason is that low density areas are characterised by large geographical spaces and long subscriber-to-subscriber distances, which makes investments unattractive due to the high cost in terms of capital expenditure and operational expenses. More so, in such areas telecommunication operators have had to contend with theft of copper cables, batteries and solar panels.

Against this backdrop, Telkom Development Laboratory (TDL), South Africa, developed the Digital Concentrator Ring (DCR-300) as a low-cost telecommunication solution to provide plain old telephone service (POTS) in rural areas, where little or no infrastructure exist. Issues such as copper cables theft, solar panel theft, systems maintenance and installation cost are the primary factors that were considered for the system design [1]. The DCR-300 is a ring-structured access network based on short haul 2Mbit/s PCM links.

In [2], Z. Milkos and H. E. Hanrahan modeled the DCR-300 and simulated the system using generic components to demonstrate its efficiency in terms of

teletraffic capacity utilization and link failure protection.

This paper builds on the access network design proposed by [2] to make a case for packet switched telecommunication infrastructure in low density rural areas. The approach is to make a comparison between the reference model and our packet based model, even though both systems have similar design. As will be shown, the DCR-300 solution will perform more efficiently if redesigned into a packet switched platform.

The justification for revisiting this project is based on the fact that the fixed line network (in Africa) is on the verge of its evolution towards the convergence of the traditional circuit switched telephony with the packet switched telephony [3]. The entire telecommunication network is also on the verge of migrating to a packet switched multi service network. In this context, the 3rd Generation Partnership Project (3GPP) had introduced the IP multimedia Subsystem (IMS) in release 5 and 6 of its technical specification, as the service engine for next generation IP networks. In subsequent releases, the IMS is positioned to support all types of fixed and mobile accesses. The Telecoms and Internet converged Services and Protocols for Advanced Networks (TISPAN) in turn adopted the 3GPP IMS standards as part of its architecture evolution [4]. This trend will see the convergence of fixed and mobile network functions, in the context of fixed mobile convergence (FMC). This will open up traditional wireline telephony for competition in the area of Voice over IP (VoIP) over broadband access [5]. Overtime, Multimedia Telephony (MMTel) based on IMS, which offers multimedia capabilities, will become the dominant communication service.

The DCR-300 was initially developed to support legacy circuit switched networks. The packet switch DCR-300 proposed in this paper will simplify the migration to convergent all-IP multiservice network while protecting legacy services at the same time. More so, it will have the capability to provide integrated high quality voice, full rate fax, video and data connectivity.

The paper is arranged as follows, Section II details the reference model (DCR-300) and its teletraffic derivatives in perspective. Section III presents our

model. Section IV presents the concept of voice packetisation. Section V gives the teletraffic performance evaluation of our model. Section VI makes a comparison between the two approaches. Section VII presents a perspective of voice packet delay follow by simulation result in VIII and conclusion in Section IX.

II. THE DCR-300 MODEL AND ITS TELETRAFFIC DERIVATIVES IN PERSPECTIVE

Figure 1: The DCR-300 System

The DCR-300 system is designed to provide POTS to different terrains in rural areas where little or no infrastructure exists. The system is modeled in a ringed pattern as shown in Figure 1. The original plan [1] indicates that the system makes use of wireless or wired connection (radio, optical fibre, optical laser etc) up to 300 meters from customer after which copper is used to the customer location. The customers are connected from concentration terminal (CT), each of which can serve up to 128 customers. The CTs are then connected together in two unidirectional loops (in opposite directions), to ensure restoration should the primary link fail.

In [2] some modifications were introduced to the original system design to allow for capacity expansion. The unidirectional links in the ring was replaced with bidirectional links to increase the number of customers in the ring. The R2 bottleneck interface at the exchange was replaced with two interfaces to allow originating and terminating traffic to compete for a single pool of slots at the exchange. By so doing, 15 channels resource initially available at the ring was increased to 30 channels, and hence, network traffic was increased from 14.22 Erl to 18.405 Erl with the same grade of service (GoS). Seven CT ringed network was modeled, as outlined in Figure 2.

Figure 2 Seven-CT ring network

Traffic measurements from the simulated system were compared with results from theoretical calculations using Erlang B formula. Firstly, the network was examined with 1000 users generating only telephone traffic, and secondly, with 500 users generating mixed traffic. Simulation runs for 1000 voice users gave an average GoS of 0.25% and 43.25 mErl per user. The result is based on average holding time of 180 seconds and slot occupancy of the first and last link in the network was 21.63 and 21.53 slots respectively, out of 30 possible slots for each link. Simulation runs with Erlang B theory shows that 1000 users at 43.25mErl per user (originating and terminating sessions) competing for 60 slots gives a GoS of 0.29%. The closeness in the values of the two results show that the routing strategy chosen divided the traffic almost equally among the network links thereby minimizing blocking [2]. Similarly, simulations results showed that the system can handle half (500) the amount of users when it supports mixed traffic. However, variable traffic erlang per user were considered for telephone, payphone and internet dial-up.

III. PACKET SWITCHED DCR-300 SYSTEM MODELLING

This new concept of IP-based DCR-300 is inspired by the original system first developed in [1]. However, our system is purely packet-based. Beginning from the exchange, we proceed to modify the original network and its components to suit specifications for a VoIP network.

We assume that three interfaces are provisioned at the network edge to connect the DCR-300 ring to the internet, Public Switched Telephone Network (PSTN), and the cellular network, for global telecommunication connectivity. The interface into the DCR-300 ring should have transcoder functions to packetize incoming Pulse Code Modulation (PCM) channels into the ring. Nowadays a number of vendors offer converged gateway equipments that can perform multiple functions. The gateway should be a router with enough digital signal processing (DSP) resources performing the functions of a transcoder which can terminate the incoming G.703/704 links and packetize the speech into the ring. Since two G.703 PCM 2.048 Mbps links were used to evaluate the system in [2], we shall use the same link specification here amounting to a total of 4.096 Mbps link. We also propose that call managers be

integrated with the router to control call features at the customer edge.

For the purpose of analyses in this paper, we assume the ring is radio, optical fibre, optical laser etc and IP enabled through DSL technologies like SDSL which offers symmetrical data rates. Layer-3 addressing will be used for routing of packetised voice through the ring to the CTs. Connection to the customers will be through a small router with an X.21 interface, used to terminate the ring on each CT. The router is then connected to an Ethernet-based IP switch which connects to customer-end analogue phone gateway. Analogue voice Gateways (such as Cisco VG224/ VG248) can be used to integrate the customer copper line to the IP-based DCR-300 system. Cisco VG200 series gateway can service analogue devices up to 1km away. We propose that the CTs should house multiple of such gateways depending on the amount of anticipated users.

Figure 3 details our perspective of the proposed IP based DCR-300 system. This architecture enables the converged router at the exchange to communicate directly with each router at the CTs. As such, QoS can be introduced into the network to prioritize voice traffic. Access to the network can be via wireline and/or wireless access points. For the purpose of comparison with the reference architecture, we assume wireline access. At the customer end, we propose that plain old analogue phones be installed and connected to the switch via copper cables. As against what obtains with the referenced system where it is required that solar panel be installed at the customer location, we propose that the customer phones be power over Ethernet [6]. This will save the cost of providing means of power to subscriber location. Rather than subscriber location, we suggest that high capacity solar panels be installed at each CT site and adequate security provided.

Figure 3: IP Based DCR-300 System

IV. VOICE PACKET MODEL

The modeling of voice traffic is largely attributed to the seminal work conducted by Paul T. Bradly, investigating the statistical nature of telephone conversation [7]. The key observation is that voice conversation can be represented by ON/OFF patterns. During talk-spurts the state is ON and at idle period, the state is OFF.

VoIP sources are modeled based on such ON/OFF characterizations. The VoIP source produces packet streams when in ON state and no packets are produced in the OFF period. The packet size and spacing depend on the codec used and particular protocol stack, typically RTP/UDP/IP [8][9].

Voice over IP utilizes different codecs for converting from PCM signals to IP packets and vice versa. Digital signal processing modules vary in terms of the bit rate they utilize, compression ratio and frame size [10], Table 1 [9] lists the variations for different codec types.

Coder

Type

Rate [kbps]

Packetization period [ms]

Frame size [ms]

Algorithmic delay [ms]

Codec delay [ms]

G.711 PCM 64 20 0,125 0 0,125 G.723.1 MPC-MLQ 5,33 30 30 7,5 37,5 G.723.1 ACELP 6,4 30 30 7,5 37,5

G.726 ADPCM 32 20 10 0 10 G.728 LD-CELP 16 30 0,625 0 0,625

G.729A CS-ACELP 8 20 10 5 15

Table 1: Variations for Different Codec Types

Pulse-Code Modulation; MPC-MLQ –Multipulse LPC with Maximum Likelihood Quantization; ACELP – Algebraic Code Excited Linear Prediction; ADPCM – Adaptive Differential Pulse Code Modulation; CS-ACELP – Conjugate Structure Algebraic Code Excited Linear Prediction.

Table 2 [9] list different protocol and their corresponding header and tail sizes.

Technology/ Protocol Header and tail size [B]

Ethernet 14 B Frame Relay 4 B

PPP 6 B IP+UDP+RTP 40 B IP+UDP+Crtp 2 B

Table 2: Header and Tail Size

In this paper, we make some basic assumptions for the purpose of analyses. We assume that the network offers only voice traffic. We assume that G.729 encoder is used for all voice traffic, producing a voice stream at the rate of 8 Kbps, with parketization period of 20 ms.

Voice Packet Calculation:

• (8x1000) bits/s x (20/1000)s=160 bits=20bytes. • Total size of header=40bytes (IP header, 20

bytes; UDP header, 12 bytes and RTP header, 8 bytes). However, RTP compression is able to reduce the header size to 4 bytes.

• Total packet size=30 bytes (voice payload, 20 bytes; compressed IP/UDP/RTP header, 4 bytes; Multilink Protocol, 6 bytes).

• Total packet size (bits)=30 bytes*8 bits=240 bits.

• PPS=(8kbps codec bit rate)/160 bits = 50pps Note: 160 bits=20 bytes (default voice payload)*8 bits per byte.

• Bandwidth per call=voice packet size (240 bits)*50pps=12 kbps.

Thus, instead of using the basic digital signaling rate of 64 kbit/s, corresponding to the capacity of one voice-frequency-equivalent channel, only 12 kbps is required per call.

V. PERFORMANCE EVALUATION

We shall adopt the same parameters and assumptions used in [2] to evaluate our IP-based DCR-300 model, so that we can be able to compare both systems. In the reference paper [2], removal of R2 interface bottleneck made 4096 kbps of bandwidth available for voice calls. Given that 64 kbps is required for one voice channel, we can deduce that the DCR-300 ring only have 64 channels available for voice calls (4096 kbps / 64 kbps). However, in our IP-based DCR-300 model 12 kbps (which represents a continuous talking path in the circuit-switch model) can be accommodated 341 times in the 4096 kbps ring. If, however, standard 150 bps is to be reserved for signaling between the phone and the call manager (for every call instance), that will mean setting aside a conservative estimate of 10% of 4096 kbps. This implies that our IP-based DCR-300 model can accommodate approximately 307 voice channels (3686.4 kbps/ 12 kbps).

According to the reference paper, using Erlang B formula / !∑ / ! (1)

A total of 1000 users at 45 mErl per user competing

for 64 slots give a GoS of 0.15%. Given the same GoS or blocking probability, B, of 0.15% and number of voice channels, N, as 307, the offered traffic is 270. With each user generating 45 mErl, it means our IP-based DCR-300 system can support a total of 6000 users (270/0.045).

VI. SYSTEMS COMPARISON

Voice is transmitted on IP-based networks at lower cost than circuit switched networks which require a dedicated connection for the entire call duration. Hence in terms of efficiency, our proposed IP based DCR-300 system makes more efficient use of bandwidth than the circuit switched model.

In terms of economic considerations one may argue that our IP-based model deploys much equipment at the access. The high cost of equipments is, however, compensated for by the enhanced efficiency of the system and the reduced cost of recurrent expenditures in the long run. For example, the solution is more conservative in terms of energy consumption due to the use of power over Ethernet to power up phones [6]. Installed solar panels at the CTs will prevent solar panel theft which characterizes the circuit switched model. Moreso, the IP-based model is designed to be scalable. Wireless access points can be used to integrate a large coverage area. This way, larger coverage is guaranteed without long cable overlay to every location.

It can also be argued that the circuit switched model should offer higher voice quality due to the fact that channels are exclusively reserved for voice signals within the ring. However, our IP-based model have features for call admission and control (CAC), to prevent voice quality degradation due to the over-exhaustion of bandwidth within the ring. CAC can be implemented both in the router at the exchange to limit the number of calls within the DCR-300 ring and at the routers connected to the various CTs.

However, a shortcoming in our proposed IP-based DCR-300 model is packet delay. Thus, we turn our attention to the issue of voice packet delay, its impact on the overall usability of VoIP and how it can be overcome.

VII. END-TO-END DELAY OF VOICE PACKETS IN PERSPECTIVE

Figure 4: Dimensioning a link for voice sources over IP networks

Our aim in this section is to present an analytic model of the queuing/switching delay of voice traffic in our proposed system. Figure 4 illustrates the teletraffic problem scenario. Traffic from a number of voice sources are aggregated by the CTs and emptied into the ring. We assume that Node 0 represents a summation of the seven CTs and Node 1 represents the router at the exchange. The ring is bidirectional, and hence, voice

packets originating from the any CT will take the shortest path to Node 1. Voice traffic is prioritized at both Node 0 and Node 1. We assume that subscribers’ density at all source nodes is designed in such a way that congestion cannot occur at Node 0. We also assume that congestion cannot also occur in any of the links connecting the individual CTs because the links are bidirectional and packets can be routed through secondary routs if the primary routes are congested. However, given a limited amount of link bandwidth and considering the fact that voice traffic are aggregated from various nodes, Node 1 is prone to congestion and becomes the primary cause of queuing/buffer delays in the network.

Before we present the analytic model of queuing/switching delays in Node 1, we first distinguish the delay causing factors in a VoIP network. Delay components in a VoIP network are either fixed or variable. Fixed component delays are predictable and derives from propagation delays within the network e.g. packetization, coding, propagation delays etc. Variable delays arise from queuing delays in the egress trunk which buffers on the outgoing interface of the router e.g. queuing/switching delays - clocking frames into and out of intermediate switches.

The different delay components affect voice packet in different ways. It is important to understand and account for delay when designing a VoIP network because the mean delay is a function of the final voice quality. While the Engineer cannot change fixed delays, it is important that VoIP networks are designed in such a way that there is less probability that voice packets are delayed in queue.

Based on standards, end-to end delay in a VoIP network should be less than 150 msec, packet loss less than 1% and jitter less than 30% [11].

Defining an analytical model for the whole delay component in a VoIP network falls outside the scope of this paper. The next subsection, however, deals with the aspect of delay that applies to queuing theory.

A. Analytical Model of Switching Delay We make the following assumptions and

derivations based on the model adopted in [12]. • We ignore the switch’s buffer size and assume

M/D/1/k (where k defines the buffer size) model is replaced with M/D/1/∞ model, to be able to get an idea of voice packet delay in the switch.

• The arrival process of the sum of voice packets from various sources, say M, to Node 1 is a Poisson process with exponentially distributed random variable. We denote the arrival rate of a source as λ.

• The service rate is denoted as µ and it is constant because in our model the same codec is used for all voice traffic.

• Service discipline in the priority queue is FIFO, first in first out.

• The length of the queue is infinite.

The output line utilization factor, , of the switch is (2)

for stability, 0≤ 1 Since all voice traffic entering the switch use the

same codec and transmission medium, arrival rate, λ, is given as . . (3)

where represents the constant bit rate of the voice traffic, represents constant voice packet size.

If the output line transfer rate and the switch’s packet processing time are known, we can define the service rate, µ, as thus

µ . (4)

where denotes line speed [kbit/s], denotes header and tail length [b] and denotes the packet processing time in the switch. Hence, by substituting (3) and (4) into (2) we get the following formulation for the system overloading . . .. (5)

The mean service time, T, is given as [s] (6)

where is the service time for one voice packet. Substituting (4) and (5) into (6) we get . . . . . . .. . . [s] (7)

Then for the probability that that there are k requirements waiting in the system, the following is valid: 1 ∑ . .! for k≥2 1 1 for k=1 (8) 1 for k=0 Substituting (2), (3) and (5) into (8) we get the following for the probability that there are exactly k requirements in the system: For k≥2 1 . . .. .

1 . . . . .. .. . . .. . . . . ..!

for k=1 1 . . .. . . . . 1 , (9)

For k=0 1 . . .. .

Therefore, expressing the service time with the help of (4) for the service rate, we can get the following for the delay probability . . for k<0,∞ (10)

VIII. SIMULATION RESULTS

Figure 5 below shows the formulation of (6) achieved using matlab to model the mean service time, T, of voice traffic with respect to a number of voice traffic sources. We assumed the following parameters; 12 Kbit/s constant bit rate, , of voice traffic; 160 bits for voice payload size, ; 80 bits for the total size of header, ; 4096 kbit/s as bandwidth of the link.

Figure 5: Mean Service Time with Respect to Number of Sources

It is observed that the voice packet mean service time, T, increases relative to the increase in the number of voice traffic load until it reaches a threshold of 2.2s. The increase in the mean service time when voice traffic increases from 80 to 100 is just 0.2s. This simple simulation attests to the fact that the increase in the number of voice traffic for packet-switched system does not have much impact on the system performance.

IX. CONCLUSION

A system that is able to reduce the amount of copper concentration for distributed telecommunication

infrastructure for low density areas has been proposed in this paper. We remodeled an initial system design by Telkom, South Africa, by enabling speech packetisation. It has been shown that our proposed packet switched model can handle 6000 voice traffic, which is six times the amount of traffic that the referenced circuit switched model can contain. Also, our proposed model is energy conservative due to the possibility of using power over Ethernet to recharge the end devices. Hence, the problem of solar panel and batteries theft at customer locations does not arise because solar panels can be installed at the CTs where adequate security can be provided.

Going further, we have presented a mathematical model of queuing/switching delay in our VoIP model using queuing theory. From the simulation result, it is observed that the effect of the addition of voice sources to the allowable threshold does not result in performance degradation for packet switched systems. However, there is the need for more simulations to access the delay probability of the system.

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