Packet or Circuit Switched Voice Radio Bearers - A Capacity Evaluation for GERAN

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1 Packet or Circuit Switched Voice Radio Bearers - A Capacity Evaluation for GERAN Mats Arvedson, Magnus Edlund, Ola Eriksson, Andreas Nordin and Anders Furuskär Radio Communications Systems, S3, Royal Institute of Technology SE Stockholm, SWEDEN {i97_ans; i98_eds; i98_ena; i98_nos}@lector.kth.se; anders.furuskar@era.ericsson.se ABSTRACT GERAN, including the EDGE radio interface, constitutes the 3G evolution of GSM. The use of packet switched voice radio bearers has been discussed in the GERAN standardization, but has as yet not been adopted. This paper presents a thorough capacity comparison of circuit and packet switched voice bearers, applied to GERAN. Thus the standardization decision is verified, but also a framework for future similar investigations is provided. Several parameters affect the relative performance of circuit and packet switching. This report evaluates the effect of four of these parameters, expected to have major impact on the relative performance: reuse factor, resource pool size, voice activity, and additional overhead in the packet switched case. Varying the above parameters, it is seen that in certain blocking-limited scenarios packet switching may increase capacity significantly. The highest overall capacity is however achieved in interference-limited scenarios, for which packet switching provides no capacity gain. 1 INTRODUCTION Voice traffic in wireless communication has so far mostly been circuit switched. Accordingly, previous evaluations of communication networks have mainly considered packet switched solutions for data traffic. Lately however, on the fixed Internet packet switched voice over IP has grown in popularity and is expected to be a significant means for carrying voice traffic in the future. The characteristics of radio networks differ from those of fixed networks, and it is not obvious that packet switched voice bearers are beneficial in this case. Consequently, the GERAN standard supports only circuit switched voice radio bearers. If sufficient capacity gains can be achieved, it may however be argued that the complexity increase for making the packet switched GPRS data bearers voice capable may be worthwhile. This paper evaluates the potential gains of packet switched voice bearers, and determines in which scenarios they can be achieved. Although the focus of the paper is on GERAN, the analysis employed is not strictly limited to this scope, but could be reused for similar investigations of other systems. A general GERAN overview may be found in e.g. [1]. Capacity evaluations of voice radio bearers in GERAN have earlier been presented in e.g. [1] and [2]. However, in [1] the comparison was limited to a few specific scenarios and realizations of packet switched bearers. In [1], capacity is evaluated for GERAN with half rate channels using 8-PSK as the modulation format. A comparison of a GSM system and a packet switched GERAN system

2 concerning voice traffic is performed in [2]. A performance analysis of voice over EGPRS can be found in [8]. In [6] a capacity evaluation is done concerning a www-traffic model in EDGE. A performance analysis of voice over EGPRS can be found in [8]. This paper adds to the above by comparing circuit switched voice bearers with a set of different packet switched realizations in a number of different traffic and deployment scenarios, thus giving context to the question of packet versus circuit switching. The paper is outlined as follows. In Section 2, a brief GERAN background is provided; the radio interface in general and the radio bearers in particular. Section 3 contains the problem definition, i.e. the scope of investigation in this report. The system models chosen, and a thorough description of these, are presented in Section 4. Section 5 covers a description on how these models are simulated. The simulation results are presented in Section 6. This is followed by a discussion of the applicability of the analysis and results to other systems than GERAN in Section 7. Finally, conclusions are presented in Section 8. 2 GERAN OVERVIEW GERAN denotes a radio access network combining existing GSM and EDGE bearers and connecting to the UMTS core network [1]. With its unique ability to operate in existing 2G spectrum and reuse existing GSM base stations, GERAN efficiently parallels other 3G radio access technologies such as WCDMA. The GERAN also enables a common evolution of GSM and TDMA/136. The four different service classes defined for UMTS (conversational, streaming, interactive and background) are supported by GERAN. GERAN may be viewed as complement to UMTS for UMTS operators with high capacity demands, or as an alternative for operators without a spectrum allocation for UMTS or operators not able to make money in 3G investments. For a more detailed description of GERAN and for some performance evaluations see [1] and [2]. For a description of the EDGE concept and some protocol details see [5], [7] and [9]. Descriptions of GSM and GPRS are found in [11]. 2.1 GERAN Voice Radio Bearers With the addition of IP/UDP/RTP header compression to support voice over IP, GERAN largely reuses standard GSM circuit switched voice bearers. The 8-PSK EDGE modulation is employed for a few bearers, but typically standard GMSK modulation is used. The main characteristics of voice traffic are the delay sensitiveness and the fact that there is no time for retransmissions. The speech is encoded into frames of 20 ms. Monitoring voice activity, a DTX-transmitter that only transmits the talk spurts, can be used. This functionality will hold down the interference created within the system, since a silent user will not generate any traffic. An alternative use of the silence periods between talk-spurts is to employ packet switching, i.e. statistically multiplex other users talk-spurts in the silence periods. Packet switched voice bearers, further discussed in the next section, are not supported in the GERAN standard.

3 3 PROBLEM STATEMENT The main purpose of this paper is to evaluate whether the GERAN voice radio bearers should be circuit or packet switched, for maximum capacity. More precise, which of the two techniques yields higher system capacity, given a specific bearer design, traffic and operational scenario. The major advantage of the PS technique is the possibility to utilize the available spectrum more efficiently. With a CS technique, each user has a dedicated channel, which is occupied during the call, irrespectively of whether the user is speaking or not. The PS technique can utilize the empty slots efficiently, by assigning channels only to packets transmitted, and not to specific users. The main drawback with packet switching is the additional overhead needed for addressing information, and the need for buffers. Since a user is never guaranteed a free channel instantaneously, sometimes packets will be buffered. Because of the delay sensitiveness of voice traffic, packets delayed too long will be discarded, which will decrease the voice quality. Because of this delay dropping and the additional overhead, PS cannot outperform CS in an interference-limited system. In such a system, there is no need for a more efficient usage of the channels. However, for a blocking-limited system the packet switching can potentially perform better. To make the investigation more general and thorough, the main problem will be evaluated for different network layouts and parameters. This results in a number of operational scenarios, specified in the next subsection. 3.1 Operational Scenarios The following design parameters will be varied: Reuse factor Voice activity Additional overhead in PS Size of channel resource pool Two different reuse patterns will be evaluated: 1/3 and 3/9. This variation is interesting since the packet switching will be more advantageous in a blocking-limited system than in an interferencelimited system. A system in which the capacity is blocking-limited could benefit from packet switching, since this technique makes utilization of nonoccupied time slots possible. The impact of the voice activity is expected to be twofold. First, a low voice activity will reduce interference, hence making the system less interference-limited. Second, the voice activity will determine the ratio of unutilized channels, indicating the potential capacity gain of the packet switching. A drawback with the packet switched radio bearers is that the additional overhead introduced will lower the power per information bit, resulting in a lower C/I. By investigating the main problem for different amounts of overhead, the impact of the additional overhead can be concluded. Finally, the size of the channel resource pool, into which a packet can be multiplexed, is varied. If a packet is limited to be transmitted on a subset of the available channels in the cell, this will clearly affect the performance of the packet switching technique. The probability that a channel within this subset is available is less than the probability that any channel in the cell is available. Therefore,

4 a small channel resource pool will increase the number of packets in the buffer, and consequently the delay dropping will increase. By evaluating the performance for different scenarios, the relative impacts of the different parameters are revealed. Furthermore, possible bottlenecks in existing technology could be identified. 3.2 Delimitations To isolate the problem, only the radio interface of GERAN will be considered, i.e. no further network performance degradations are assumed. Furthermore, only the downlink performance is evaluated. The radio interface performance is supposed to be limited by the downlink quality, since the uplink quality more easily can be enhanced by means of diversity. However, when it comes to the packet switched case, the channel allocation in the uplink becomes more complex than in the downlink, since the allocation process has to be distributed. This information signaling will also introduce further delay to the speech. The downlink evaluation could therefore be said to be a "best-case" study. If the downlink fails to show that packet switched bearers are superior to circuit switched, this will also be the case for the uplink. If the performance of the downlink is better with packed switching, and the reverse holds for the uplink, maybe the uplink should be circuit switched and the downlink packet switched. 4 SYSTEM MODELING 4.1 Modeling Approach Due to the complexity of the problem, a simulation approach is adopted. The MATLAB environment RUNE 1 [10] constitutes the frame in modeling the system in focus. 4.2 Voice Traffic Modeling Calls arrive to the system according to a Poissonprocess, and terminals are uniformly distributed in the service area. The call lengths, as well as the talk spurts, are exponentially distributed. The system is evaluated for voice activity factors of 60 %, 50 %, and 40 %. 4.3 Channel Assignment Modeling In the system, a static channel allocation to the base stations is employed. The circuit switched channel assignment scheme assigns channels to arriving calls if there is any channel available. A call occupies the assigned channel during the whole holding time, even when there are no frames transmitted (silence). The number of channels available does not alone limit the capacity of the packet switched system. Since the assignment of channels is performed for every time slot, only calls transmitting a frame during that time slot will participate in the channel assignment process. This implies that the maximum capacity of the packet switched system is determined by the number of channels and the 1 RUNE, Rudimentary Network Emulator. In the simulation, an extended and modified version was used

5 voice activity, in contrast to the circuit switched case where a silent call still occupies its dedicated channel Queuing Modeling The packet switched case introduces a queue in the system, allowing voice frames to be slightly delayed. Speech is sensitive to delay and the maximum queuing time is an important variable to consider. Packets are buffered base station-wise, and ordered according to their accumulated queuing time. Delayed packets are prioritized in the channel assignment decision, in order not to cause delay dropping. Packets with the same delay are assigned channels in a random manner. More exactly, random channels are assigned to random packets. Packets buffered for more than the maximum allowed queuing time are discarded. The queuing model can be described by a number of M/D/n queuing systems, one for each base station. Packets are served in a First Come First Serve manner. A large pool implies higher efficiency due to more possible allocation solutions. As part of the packet switched system, the size of this pool is modeled and evaluated. The concept of channel resource pools introduces a modified queuing model. Now, the buffer of each base station will consist of N different M/D/m systems, where N is the number of combinations of resource pools that a terminal can receive on, and m is the size of one resource pool. 4.4 Frequency Hopping Modeling To efficiently model the slow frequency hopping employed in GERAN, the two main characteristics of slow frequency hopping are considered, namely the interference averaging and the frequency diversity. The interference averaging is done for each terminal by calculating the interference level on all available channels, and using the average interference in the C/I ratio. The frequency diversity is included in the FER in Figure Frame Erasure Rate vs C/I Channel Resource Pools A packet can be multiplexed onto a set of different channels; a specific subset of the channels in the system. The size of this channel pool is limited by hardware in the terminal. At present, a GSM terminal can only listen to one frequency and one time slot at a time. If the terminal is equipped with a more advanced receiver it can receive on multiple frequencies and time slots. In this model, the size of the channel resource pool is varied from 8 to 24 channels, representing reception on all 8 time slots for one to three frequencies. FER C/I (db) Figure 1 FER vs. C/I 4.5 Channel Modeling The relation between the Frame Erasure Rate (FER) and the Carrier-to-Interference Ratio (C/I)

6 is determined by the speech and channel coding, the modulation format, and the fast fading, multipath channel. As argued in [10], a simple mapping of C/I to FER can successfully represent this rather complex dependency. For GERAN this relation is previously described and plotted in [3], and reproduced below (Figure 1). The speech-coding format employed is EFR (Enhanced Full Rate), the modulation format is GMSK, the propagation model is TU3, the terminal velocity is 3 km/h, and ideal frequency hopping is assumed. For the packet switched radio bearers, the curve in Figure 1 will be shifted to the right, due to the additional overhead 2. This report considers three levels of degradation, introducing a rightward shift of 0 db, 0.4 db and 1.0 db respectively. This roughly corresponds to additional overheads of 0 %, 10 %, and 25 % (neglecting the increase in code rate). 4.6 Performance Measures Two performance measures are used in this evaluation, one Quality-of-Service measure (user s perspective), and one capacity measure (operator s perspective). The capacity is measured as the number of calls per site, divided by the allocated bandwidth [Calls/Site/MHz]. The Quality-of- Service requirements are: Maximum 1 % average FER Maximum 2 % blocking probability A satisfied user is defined as a user who is not blocked and experiences an average FER of maximum 1 %. The maximum capacity of an operational scenario is defined as the capacity at which at least 90 % of the users are satisfied, under the constraint that no more than 2 % of the arriving calls are blocked Model Delimitations No admission control is modeled in the packet switched system. Accordingly, no call attempt is blocked by the system. No power control is employed in either of the PS or the CS case. 4.8 Model Limitations The voice activity factor and the distributions of talk spurts are likely to be different for different users depending on language, culture and the nature of the call. These different distributions of voice activity are not covered in the simulation model. Moreover, the exponentially distributed holding time employed here is questioned in [4]. Still it is still the most commonly used model. 5 SIMULATIONS The simulations are carried out using a RUNE based simulation environment. The systems are simulated in steady state for 4 minutes (real time domain). By doing this, 98 % of the call length distribution is covered, and the loss of generality in this matter is minimal. The simulation employs 2 The additional overhead will reduce the power per transmitted information bit. Furthermore, the coding rate has to be increased for the information to fit into the 20 ms frame. 3 The blocking probability is theoretically determined by the Erlang-B formula.

7 a discrete time step of 20 ms, corresponding to one speech frame. For each frame, the C/I ratio is measured and mapped onto the corresponding FER level (see Figure 1). A packet that is discarded due to a too long delay will have a FER of 100 %. Based on this, an average FER for each call is calculated. The main simulation parameters are listed in Table 1. Variations of the parameters reuse pattern, additional overhead, DTX-efficiency, and channel resource pool size constitute the different operational scenarios. Table 1 Simulation Parameters Traffic Type Real-Time Voice Time Discrete time step, 20 ms Reuse 1/3, 3/9 Site-to-Site Distance 800 m (3/9), 1350 m (1/3) Propagation Model log(r) [db] Lognormal fading, standard deviation 8 db Lognormal Correlation Distance 110 m Spectrum 1.8 MHz (72 channels) Frequency Hopping Mobile Speed Random (interference averaging) 3 km/h DTX Voice Activity 60 %, 50 %, 40 % Distribution of talk spurts/silence/call length Exponential Mean length of talk spurt 1.1 s, 0.9 s, 0.7 s Mean length of silence Mean call length Handover Handover Margin Power Control Channel Allocation Channel Resource Pool Size Terminal distribution Maximum delay 0.7 s, 0.9 s, 1.1 s 60 s Selection of strongest BS 3 db None Static 8, 16, 24 Uniform 40 ms, 2 speech frames Additional Overhead (PS) 0 %, 10 %, 25 % Admission Control None 6 SIMULATION RESULTS In this section, the results from the simulations will be presented. First, the effects of the different parameter variations are displayed and discussed ( ). Second, the results from the scenarios where packet switching outperforms circuit switching are presented (6.5). Finally, the maximum capacities of the different operational scenarios are visualized in a bar chart (6.6). 6.1 Overhead Effect The effect of a large additional overhead for the PS case is, not very surprisingly, a performance degradation. However, as seen in Figure 2 the effect of overhead is decreasing as the system load increases. The overall degradation is of course due to the decrease in C/I caused by the lower power per information bit, and consequently an increase in FER. As the load of the system increases, a greater portion of the frame erasures are caused by delay dropping, which is independent of the C/Ivalue. This explains the decreasing effect of the additional overhead. 6.2 Resource Pool Size Effects As Figure 3 suggests, the effects of variations in channel resource pool sizes for loads below 16 Calls/MHz/Site, are negligible. At this low load, hardly any packets are buffered for any size of the resource pools. As load increases, the number of delay dropped packets increases. The tendency is that the smaller the resource pool, the greater the performance degradation.

8 Fraction of Satisfied Users [served] Overhead Effect (Reuse 1/3, Voice Activity 60%, Resource Pool 8) CS PS, Overhead: 0% PS, Overhead: 10% PS, Overhead: 25% Fraction of Satisfied Users [served] Pool Size Effect (No Overhead, Reuse 1/3, Voice Activity 60%) CS Resource Pool: 24 Resource Pool: 16 Resource Pool: Capacity [Calls/MHz/Site] Capacity [Calls/MHz/Site] Figure 2 Overhead Effect Figure 3 Pool Size Effect At offered loads around the size of the resource pool, the different resource pool buffers are virtually empty, since the available channels can handle almost all arriving packets. At moderate loads, the offered traffic can get unevenly distributed to the different buffers, introducing a trunking loss to the buffering system. This is causing the system with resource pool size 8 to perform worse than a system in which a packet can be multiplexed into all available channels, since in this case three queuing systems of eight servers (channels) each, will cause more delay dropping than one system of 24 servers. At high loads, all the buffers are heavily loaded, and most of the packets that are delay dropped would be discarded irrespectively of the resource pool size. Therefore, the performance degradation decreases as the load approaches maximum capacity. 6.3 Voice Activity Effects The voice activity, or rather the combination of the voice activity and the DTX efficiency, is a parameter that affects the performance of both the packet and circuit switching techniques, contrary to the case in the two previous sections. Since the DTX transmitter only transmits the talk spurts, and is quiet when there is no speech, the voice activity clearly affects the interference in the system. However, in terms of capacity, i.e. the number of calls that the system handles, the effects on the two switching techniques differ somewhat. For the CS case, a lower voice activity will result in lower interference and hence, higher quality of service. The system gets less interference-limited. This means that a system that is blocking-limited, will not benefit from a lower voice activity, in terms of maximum capacity as defined in (4.6). For the PS case, the effect of a lower voice activity is simply a scaling of the capacity axis, resulting in a rightward shift of the curve, as depicted in Figure 4. This is quite logical, since the call quality depends on the number of packets transmitted, and not on the number of calls in the system, which is the capacity measure. For a fixed fraction of satisfied users, a system with a lower voice activity can handle more calls than a system with higher voice activity. In fact, this relation should be linear.

9 Fraction of Satisfied Users [served] Voice Activity Effect (25% Overhead, Reuse 1/3, Resource Pool 8) CS, Voice Activity: 40% PS, Voice Activity: 40% CS, Voice Activity: 60% PS, Voice Activity: 60% Fraction of Satisfied Users [served] Reuse Effect (Voice Activity 60%, Resource Pool 8, 25% Overhead) CS, Reuse 1/3 PS, Reuse 1/3 CS, Reuse 3/9 PS, Reuse 3/ Capacity [Calls/MHz/Site] Capacity [Calls/MHz/Site] Figure 4 Voice Activity Effect Figure 5 Reuse Effect A PS system in which 20 users talk on average 60 % of the time, should hold the same quality as a system with 30 users talking 40 % of the time. Figure 4 supports this reasoning. 6.4 Reuse Effects As seen in Figure 5, a large reuse will mean lower capacity for a fixed amount of bandwidth. Another drawback is the increased blocking probability (for CS), due to so-called trunking losses because of a smaller number of channels allocated to each site. The solid vertical lines in figure 5 displays the 2 % blocking probability limits for the two reuses (The left line corresponds to reuse 3/9, and the right line to reuse 1/3). 6.5 Scenarios where PS outperforms CS By varying the different parameters, and plotting the capacity curves, the scenarios for which the packet switching technique outperforms the circuit switching are determined. The different scenarios are identified for one reuse factor at the time, starting with reuse 3/ Reuse 3/9 In the 1/3 reuse system with the parameter setting in Figure 5 above, the CS technique yields higher maximum capacity (as defined in 4.6) than the PS technique. The CS system is interference-limited, since the system reaches 90 % satisfied users before the 2 % blocking line. Considering the system with reuse 3/9, reproduced in Figure 6, the situation will be the other way around. The maximum capacities are marked with a circle in the figure. This system is blockinglimited, and the PS technique yields the higher maximum capacity. With this result concluded, some further statements can be made. For reuse 3/9 the PS technique will in fact outperform the circuit switching for all voice activities below 60 %, for all sizes of channel resource pools larger than 8, and for all overheads below 25 % (i.e. for all combination of these). A decrease in voice activity will further increase the blocking-limitation of the CS system, whereas the voice quality of the PS system can be maintained for a higher number of users (according to the discussion in 6.3).

10 1 Reuse Effect (Voice Activity 60%, Resource Pool 8, 25% Overhead) 1 Reuse Effect (Voice Activity 40%, Resource Pool 16, 25% Overhead) 0.98 Fraction of Satisfied Users [served] CS PS Fraction of Satisfied Users [served] CS PS Capacity [Calls/MHz/Site] Capacity [Calls/MHz/Site] Figure 6 Reuse 3/9 Figure 7 Reuse 1/3 A larger number of channels onto which a packet can be multiplexed will enhance the PS performance due to trunking gains (see 6.2 and 4.3). A decrease in additional overhead will be of unilateral benefit to the PS technique Reuse 1/3 For the 1/3 reuse system, the packet switching technique does not perform better than circuit switching in general. For the scenarios with a voice activity of 60 % and 50 %, the circuit switching yields higher capacity for all other settings of the overhead and pool size parameters. For a voice activity of 40 % however, no such general conclusion can be drawn, but the other parameters has to be varied as well. As figure 7 depicts, PS will do better than CS for 40 % voice activity, if the overhead is 25 % and the size of the resource pool is 16. With the same reasoning as in the previous subsection (6.5.1), it can be concluded that with a voice activity lower or equal to 40 %, PS will outper form CS for all overheads lower than 25 %, and for all resource pool sizes larger than Summary of Results The effects of the variations in additional overhead are small, compared to the variations of the other parameters. Therefore, only the operational scenarios where the additional overhead equals 25 % are presented in the bar chart on the next page (Figure 8). Then, the impact of a reduction in additional overhead will only be of benefit to the PS system. The general tendency is that if the packet switching performs better than circuit switching, it is a result of the blocking-limitation of the CS system. The blocking-limited systems are marked with a * in Table 2. For the reuse 1/3 systems, the circuit switching yields equal or higher capacity than packet switching. The reader is referred to sections 6.2 and 6.3 for a thorough reasoning about the impact of voice activity and resource pool size. When it comes to the scenarios employing a reuse 3/9, all CS systems are blocking-limited, and the packet switching yields a higher maximum capacity for all levels of voice activity considered.

11 7 APPLICABILITY OF THE RESULTS The analysis presented above is not strictly limited to GERAN, but could in principle be reused for similar investigations of other systems. Rather similar results may further be expected, i.e. the more blocking limited the system is, the higher the potential gain with packet switching, whereas the highest capacity should still be found when blocking is avoided, i.e. under interference limited conditions. Among the studied parameters, the voice activity factor and addressing overhead are universal to all systems. The resource pool, which for GERAN is a set of time slots, of course needs to be translated in to the channel type of the system in question, e.g. spreading codes in code limited DS-CDMA system. Finally the frequency reuse parameter may of course be omitted if not applicable. 8 CONCLUSIONS The analyses carried out indicate that circuit switching yields equal or higher capacity for scenarios of reuse 1/3. For reuse 3/9 or larger on the other hand, packet switching may outperform circuit switching with up to 100%. The absolute capacities in the 3/9 case are however lower than for the 1/3 case. Whether to introduce packet switched voice bearers in the GERAN standard thus depends on if the gain seen in blocking limited scenarios is deemed worth the extra complexity, bearing in mind that the maximum absolute capacity is not improved. The extent of the complexity increase is outside the scope of this paper. On a more detailed level, the voice activity and the reuse factor have the largest impacts on the results. The capacity gain of being able to listen to up to three different frequencies is less than 15 %. Overhead has a small impact on the capacity. The capacity loss of an additional overhead of 25 % as compared to no additional overhead falls within 10 %. The employed analysis procedure may be reused for other services than voice, and in principle also for other access technologies than GERAN.

12 Maximum Capacity (25 % overhead) 30 Maximum Capacity [calls/mhz/site] PS2 PS3 PS4 CS2 PS1 40 % 50% Voice Activity 60 % CS1 Figure 8 Summary of Results Maximum Capacity [calls/cell/mhz] PS1 CS1 PS2 PS3 PS4 CS2 Reuse 3/9 Reuse 1/3 Voice Pool size CS Pool size (PS) CS Activity 8 Capacity P(Block) Capacity P(Block) 60 % 8,2 5,7* 2,0% 18,5 18,3 17,2 23,0 0,5% 50 % 9,7 5,7* 2,0% 22,8 21,3 20,8 26,7* 2,0% 40 % 12,0 5,7* 2,0% 27,0 26,9 24,0 26,7* 2,0% Table 2 Summary of Results 9 REFERENCES [1] M. Eriksson et al., The GSM/EDGE Radio Access Network GERAN; System Overview and Performance Evaluation, in Vehicular Technology Conference Proceedings, VTC Spring Tokyo IEEE 51st Volume: 3, 2000, Page(s): vol.3 [2] Jianjun Wu et al., Transmission of Voice in an EDGE Network, in Vehicular Technology Conference Proceedings, VTC 2000-Spring Tokyo IEEE 51st Volume: 1, 2000, Page(s): vol.1 [3] A. Furuskär et al., Managing Mixed Services with Controlled QoS in GERAN the GSM/EDGE Radio Access Network, Conference paper in Second International Conference on 3G Mobile Communication Technologies, March 2001 London UK. [4] T. Westholm and B. Olin, A Model for GSM Speech, in Proceedings of the 2000 Symposium on Performance Evaluation of Computer and Telecommunication Systems, p , [5] M. Eriksson et al., System overview and performance evaluation of GERAN the GSM/EDGE Radio Access Network, in Wireless Communications and Networking Conference, WCNC IEEE Volume: 2, 2000, Page(s): [6] A. Furuskär et al., Capacity Evaluation of the EDGE Concept for Enhanced Data Rates in GSM and TDMA/136 in Vehicular Technology Conference, 1999 IEEE 49th Volume: 2, 1999, Page(s): vol.2 [7] A. Furuskär et al., Enhanced Data Rates for GSM and TDMA/136 Evolution, in IEEE Personal Communications Volume: 6 3, June 1999, Page(s): [8] R. Järvelä et al., Capacity of voice over EGPRS service under different operational scenarios, in research report by Nokia Research center. < ers/p13.pdf> [9] K. Balachandran et al., EDGE Phase 2 Evolution of EGPRS for Supporting 3G Real-time Services, in research report by Lucent Technologies < [10] Jens Zander, Seong-Lyun Kim, Radio Resource Management for Wireless Networks, Artech House Publishers, [11] Yi-Bing Lin, Imrich Chlamtac, Wireless and Mobile Network Architectures, Wiley, 2000.