This full text paper was peer reviewed at the direction of IEEE Communications Society subject matter experts

for publication in the WCNC 2009 proceedings.

Principle and Performance of TTI Bundling For VoIP in LTE FDD Mode
Jing Han
Nokia Device R&D/Wireless System Research Nokia (China) Investment Corporation Limited, 100176 Beijing, China ext-jing.hang@nokia.com
Abstract Limited transmission power and short TTI length impose a bottleneck on LTE UL performance in a coverage-limited scenario. This paper presents an effective coverage enhancement mechanism called TTI bundling to boost uplink VoIP performance in LTE FDD mode. Performance evaluation for TTI bundling with VoIP traffic is carried out by semi-static system simulation, and the impact from different bundling options with different bundle size and packet delay budget is investigated. The simulation analysis proves that with proper bundle size, TTI bundling can enhance the coverage performance for LTE FDD effectively. Keywords TTI Bundling, Long Term Evolution (LTE), Voice Over Internet Protocol (VoIP) , Coverage Problem

Haiming Wang
Nokia Device R&D/Wireless System Research Nokia (China) Investment Corporation Limited, 100176 Beijing, China haiming.wang@nokia.com II. COVERAGE PROBLEMS AND BASIC SOLUTION Due to the power limitation of the mobile terminals and the inherent short transmission time interval (1ms TTI) in LTE system, mobile terminals often suffer from coverage problems at cell edge. With the large inter-site distance (ISD) (i.e. 1.732km), even small packets like VoIP traffic with 12.2kbps AMR codec still can not maintain its Quality of Service(QoS) at cell edge. So in a coverage-limited scenario, VoIP packet has to be segmented into even smaller packets to fulfill the required QoS. Such operation takes place in RLC layer thus called L2 Segmentation, which is already supported in LTE specifications as a basic coverage enhancement approach. Together with HARQ procedure, L2 segmentation will allow one normal packet be segmented and transmitted separately in several continuous subframe while maintaining a specified air interface delay, thus improve the coverage performance [5] to some extent. However, the potential bottleneck of L2 packet segmentation is requiring other advanced coverage solution. To elaborate the problem for L2 segmentation, the discussion on L2 segmentation is included firstly below. A. Basic of L2 Segmentation Power control, Adaptive Modulating and Coding rate (AMC) and HARQ are three main link adaptation methods used in LTE system. However, power-limited mobile terminals at cell border can not increase its transmission power and moreover, those users can not increase transmission times with the specified packet delay budget at air interface by current HARQ configuration in LTE uplink. Thus AMC is the best way to fulfill the QoS requirements. However, when performing AMC, the transmission block size can not be too large so that the payload size in transmission block has to be reduced in order to maintain power density. When the required payload size is smaller than a single VoIP packet, the segmentation at RLC layer has to be carried out and the normal packet will be divided into several sub-packets (Refer to figure1 below). Then these sub-packets will be sent separately in different HARQ processes. B. Analysis of L2 Segmentation L2 segmentation can provide significant gain from the link

I. INTRODUCTION While the data revenues are increasing, the voice service still makes majority of the operator revenues. Therefore, the long-term evolution (LTE) of the 3GPP radio access technology is designed to support not only data services efficiently, but also voice service with high efficiency [1]. Voice over Internet Protocol (VoIP) is regarded as the major voice service in LTE system because of its IP based essential and easily integrated within LTE packet switching framework. Many research work and investigation about VoIP capacity in LTE system indicate that VoIP service can obtain better frequency bandwidth efficiency than CS(Circuit Switched) voice service in interference-limited scenario (3GPP Macro Scenario1) [2-4]. However, in a coverage-limited scenario (3GPP Macro Scenario3) mobile terminals at cell edge often hit the maximal transmission power which makes the uplink performance becomes the system bottleneck. Thus, designing an effective scheme to solve uplink coverage problem to improve the uplink VoIP performance is a very critical issue. The main structure of this paper is organized as follows: Section II illustrates the uplink coverage problem and basic solution in LTE system; Section III presents the principle of TTI bundling and related HARQ design; numerical results are given and analyzed in Section IV before conclusions are drawn in Section V.

978-1-4244-2948-6/09/$25.00 2009 IEEE

This full text paper was peer reviewed at the direction of IEEE Communications Society subject matter experts for publication in the WCNC 2009 proceedings.

budget point of view. However, the drawback of this technique is obvious: firstly, the header overhead is too large because each sub-packet will consume the RLC/MAC/CRC header. Take AMR 12.2kbps VoIP traffic for example, the RLC SDU is roughly to be 288 bits in PDCP layer. If the RLC SDU is segmented, each sub-packet will attach with 1 byte RLC header, 1 byte MAC header and 3 bytes CRC overhead by L1 layer. The extra overhead ratio can be defined as: ( N seg 1) * ( Rb + M b + Cb ) Rextra = (1) Pb + Rb + M b + Cb

channel condition, one PDCCH channel will use up more control channel resource, which will exhaust the system control channel and limit the chance for scheduling other users; Furthermore, HARQ feedback error is increased because the initial transmission and retransmission for each sub-packet are followed by an ACK/NACK feedback. Assuming error rate for HARQ feedback channel is 1%, if there are 10 ACK/NACK for a single VoIP packet, the error rate of feedback channel will reach about 10%, which can not meet the quality requirement for VoIP traffic. III. TTI BUNDLING SOLUTION Although L2 segmentation is easy to implement, low transmission efficiency due to additional header overhead and excessive consumption of L1/L2 control channel limits the performance of this solution. Thus, long effective TTI length concept has been proposed [5-7] and finally converges to TTI bundling concept [7].
A. Principle of TTI Bundling TTI bundling uses automatic retransmissions in several consecutive uplink TTIs without waiting for HARQ feedback to form a longer effective TTI length. Thus it increases the amount of accumulated energy within a defined packet delay budget for a single VoIP packet. The consecutive uplink TTIs is so called a bundle while the number of consecutive uplink TTIs is the bundled size. The same MCS and frequency bandwidth will be utilized among subframes in a bundle, and only one uplink grant and one HARQ feedback channel is transmitted for a bundle.

Figure 1 L2 segmentation process (for AMR 12.2kbps VoIP traffic)

Here, N seg is the number of segmentation. Pb is the payload size in bits. Rb , M b and Cb is the protocol header in bits for RLC, MAC and CRC respectively. So when the RLC SDU is segmented to 2 sub-packets, the extra overhead due to segmentation is about 12.2% due to one more 4 bytes header; if segmented to 4 sub-packets, this segmentation overhead is reach to 36.6%, and the total header overhead comes to 56%. Those cases are even worse for VoIP traffic with 7.95kbps AMR codec because smaller payload size is used. Table 1 lists the numerical results for header overhead, which is unacceptable when the number of segmentation is large, i.e. 8 segmentations.
Table 1 L2-Segmentation overhead analysis for VoIP traffic Number of Extra overhead due Header segments to segmentation overhead 12.2kbps Codec, RLC SDU size = 36 bytes 1 0% 14% 2 12% 28% 4 37% 56% 8 85% 111% 7.95kbps Codec, RLC SDU size = 26 bytes 1 0% 19% 2 16% 38% 4 48% 77% 8 113% 154%

Figure 2 Example of TTI bundling process for AMR 12.2kbps VoIP traffic

In addition, a large number of L1/L2 control channel will be consumed. When scheduling and transmitting every sub-packet, an uplink scheduling grant is needed, thus it consumes one PDCCH channel. Assuming RLC SDU is segmented to 2 sub-packets, and then the consumed number of PDCCH channel is doubled. Considering power-limit users are always under a bad

Figure2 illustrates the process of TTI bundling operation, where the packet is autonomous retransmitted in a bundle with different Redundancy Version (RV). So TTI bundling method avoids the extra header overhead introduced by L2 segmentation, and also avoids additional L1/L2 control channel consumption and increased HARQ feedback channel error.
B. HARQ timing design for TTI bundling Since there is only one HARQ feedback for several subframes in one bundle, then this HARQ feedback can use either the first subframe in the bundle or another subframe in the bundle after

This full text paper was peer reviewed at the direction of IEEE Communications Society subject matter experts for publication in the WCNC 2009 proceedings.

receiving. According to the HARQ feedback timing, two alternatives of TTI bundling are proposed: Alternative 1 In this alternative, the HARQ feedback timing is according to the last subframe in a bundle. Assuming subframes of the bundle are transmitted in subframe k , k + 1 , k + 2 and k + 3 ; then the ACK/NACK feedback timing for the last subframe is k + 7 . And if NACK is received, then the earliest time to retransmit this bundle is at k + 11 in LTE system. However, synchronous HARQ scheme is configured for LTE FDD uplink and there are totally 8 HARQ processes, which is to say, the retransmission of this bundle has to be delayed to k + 16 . So the HARQ RTT (Round Trip Time) for TTI bundling in alternative1 is equal to 16 TTIs, which is doubled compared with original HARQ RTT. The advantage of alternative1 is that the HARQ feedback is mature and actually reflect the status of entire bundle that received by NodeB. So no unnecessary retransmissions will take place and there is no resource wasting as well. And the disadvantage is that the bundled HARQ RTT is increased to double and the energy accumulation speed is slower than that of normal HARQ RTT.

bundled HARQ process is defined by: Original _ HARQ _ No Bundled _ HARQ _ RTT (2) Bundled _ HARQ _ No = Bundled _ size Original _ HARQ _ RTT In equation (1), Original _ HARQ _ No and Original _ HARQ _ RTT is fixed for LTE uplink system, so assuming 4 bundled size, the number of bundled HARQ process is 4 for alternative1 and 2 for alternative2 respectively (refer to Figure 5).
Initial HARQ Process Bundled HARQ Process
0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7 0 1 2 3 0 1 2 3 0

Bundled HARQ process for Alternative 1

Initial HARQ Process Bundled HARQ Process

0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7 0 1 2 3 0 1 0 1 0

Bundled HARQ process for Alternative 2

Figure 5 HARQ process design for TTI bundling

IV. PERFORMANCE AND ANALYSIS In this section, the performance of VoIP with different TTI bundling alternatives and bundle sizes is investigated in details. In terms of path-loss information, users in the cells are classified into two user groups: normal users and cell edge users. Only cell edge users will use TTI bundling scheme. A. System Model We use a quasi-static system level simulator based on the 3GPP LTE system simulation scenario [8]. Our simulation platform models interference from 57 sectors (19 cells with 3-sector per cell) in a wrap-around model and users are dropped uniformly in entire sector. Inter-site distance is 1.732 km and the shadowing standard deviation is 8dB. 2 long blocks per TTI are reserved for pilots and 2 out of 25 physical resource blocks (PRBs) for uplink control channels, so there are 23 PRBs left for data. Synchronous adaptive HARQ is used and full IR combining is assumed. The maximum number of retransmissions is determined by the VoIP delay budget (DB) and number of HARQ processes. Given 50ms DB and 8 HARQ processes, then the maximum retransmission number is 6. An SNR target based power control is used to make up for the overall path loss and to control the inter-cell interference. Link to system interface used is AVI assuming practical FDE receiver and realistic channel estimation. The other parameters are listed in Table 2.
Table 2. Simulation parameters Parameter Propagation Model (dB) Sector antenna pattern Shadowing correlation between cells / sectors Value 128.1 + 37.6 Log10R, R in kilometers 70 deg (-3dB) with 20dB front-to-back ratio 0.5 / 1.0

Figure 3 TTI bundling alternative1

Alternative 2 Unlike alternative1, the HARQ feedback timing is according to the first subframe of the bundle in alternative2 (refer to figure4 below). So with TTI bundling, HARQ RTT is still the same as the original HARQ RTT. However, the HARQ feedback here is premature and there could be the case that the feedback is NACK but eNodeB correctly decodes the packet after receiving the entire bundle. So unnecessary retransmissions maybe occur and the system resources are therefore wasted. However compared with alternative1, alternative2 can quickly collect the energy for the packet decoding and this is critical to users with severe channel condition. In the following performance evaluation part, VoIP capacity with different alternative and different bundle size are given out to compare these two alternatives from the system simulation.

Figure 4 TTI bundling alternative2

C. HARQ process design for TTI bundling As to the aspect of TTI bundling timing and HARQ process number after TTI bundling, permanent TTI bundling scheme is preferred by us [7]. That is to say, the starting point of bundling is fixed to time and the number of bundled HARQ processes is fixed according to bundled size and alternatives. Thus the number of

This full text paper was peer reviewed at the direction of IEEE Communications Society subject matter experts for publication in the WCNC 2009 proceedings.

Penetration loss Channel model UE speed UE maximum power Carrier frequency Bandwidth TTI length Channel update eNB receiver UE transmitter antenna eNB/UE antenna gain Thermal noise density Frequency re-use Control channel modeling VoIP Codec

20dB SCM-C 3km/h 24dBm 2000MHz 5MHz 1ms per slot (0.5 ms) 2 antennas with MRC (maximum ratio combining) 1 antennas 14dBi / 0dBi -174dBm/Hz 1 Realistic PDCCH modeling, 10CCEs reserved for UL grant per DL subframe AMR 12.2kbps

frequency selective scheduling is used for UL VoIP considering the huge number of sounding pilots it need). To fit the block size of different packets, QPSK 2/3 with 2 PRBs allocation is for voice packet of 12.2kbps AMR codec while QPSK 1/4 with 2 PRBs allocation is for SID packet. For 3 PRBs allocation, QPSK 1/2 and BPSK 1/3 are for voice packet and SID packet, respectively.
C. QoS based Outage Criterion Two main QoS requirements of VoIP are the packet delay and packet loss rate. The packet delay in UL we modeled consists of the queuing delay in the scheduler at eNB and the packet transmission delay which mainly depends on number of retransmission. According to [8], the radio interface delay budget is 50ms and for bundled users, we also investigate the impact of longer delay budget which is 60ms or 70ms. A VoIP user is in outage (unsatisfied) if more than 2% of its packets in a 60s call can not be correctly received within 50ms delay budget. Furthermore, VoIP capacity is defined as the maximum number of per sector VoIP users that can be supported without exceeding a 5% outage level. That is, N (5) k = MAX i { FER > 2% < 5%} N Total

B. Traffic Model Voice traffic is modeled based on a 2-state voice activity model [8]. 2-state voice activity model is shown in figure6:

Where NTotal is the total number of users in the system, N FER > 2% the number of users for whom the average FER exceeds 2%, k capacity (Num of user /cell), and i the simulated number of users per cell.
Figure 6 2-state voice activity models

In the model, the probability of transitioning from state 1 (the active state) to state 0 (the inactive or silent state) while in state 1 is equal to a , while the probability of transitioning from state 0 to state 1 while in state 0 is c . The model is assumed updated at the speech encoder frame rate R = 1/ T , where T is the encoder frame duration (typically, 20ms). The steady-state equilibrium of the model requires that a c P0 = , P (3) 1 = a+c a+c Where P0 and P 1 are respectively the probability of being in state 0 and state 1. The Voice Activity Factor (VAF) is given by c =P (4) 1 = a+c 50% VAF is assumed to model half probability of active and inactive periods ( a = c = 0.01 ). The duration of both active and inactive periods is negative exponentially distributed with an average of 2 seconds. Including the compressed RTP/UDP/IP headers and other overhead, there are totally 40 bytes per voice packet for 12.2kbps AMR codec and 15 bytes for an SID (Salience Indicator) packet. Adaptive Transmission Bandwidth (ATB) is available by allocating 2 or 3 PRBs within one TTI to transmit one packet according to users path loss (no time or

D. Scheduling metric Fully dynamic scheduling is used among all users, thus each initial transmission and retransmission are allocated by one physical downlink control channel (PDCCH). Bundled user group will have higher priority than normal user group for all initial transmissions and retransmissions. In each user group, retransmission is always having the highest scheduling priority. E. Downlink control channel model Physical downlink control channel, which carries uplink scheduling grant and used for time-frequency radio resource allocation for uplink transmissions, is modeled during the investigation. That is to say, the number of Control Channel Elements (CCEs) that consumed by one PDCCH is explicitly mapped according to the G-factor value of specific user. 2, 4 and 8 CCEs can be aggregated to support one PDCCH and users that with low G-factor will consume more CCEs. When scheduling a user in UL, we will take the number of required CCEs for consideration: If there are enough CCEs available for this user, then continue scheduling. If there are not enough CCEs available for this user, try to schedule next user from the scheduling priority list. F. HARQ IR model Based on HARQ Chase Combining (CC) scheme, HARQ

This full text paper was peer reviewed at the direction of IEEE Communications Society subject matter experts for publication in the WCNC 2009 proceedings.

Incremental Redundancy (IR) is modeled by adding a coefficient factor which is the gain of IR to CC. The coefficient factor dependeds on the used MCS and the transmission number of the packet. HARQ IR model is described as: n E E (6) ( s )C ,n = n 1. ( ECR, M )n . ( s )k N0 k =1 N 0 Here, ECR is the code rate of first (initial) transmission, M is the modulation order, (Es/N0)C,n is the combined Es/No after N transmissions, is chase combining efficiency (e.g. 0.95) and (ECR) is the gain of incremental redundancy over chase combining (the value of equals 1 when chase combining is used). IR gain depends on the modulation order, coding rate and retransmission number.

Figure 7 Example of transmission number for bundled users

For the uses with TTI bundling, because the subframes that after the first subframe in a bundle are all automatic retransmissions, HARQ IR is implemented both for intra-bundle and inter-bundle. The total transmission number is incremented by each occurrence of retransmission. Take 4 bundled size for example, if there are two transmissions of the bundle, then the total transmission number is 8 (Figure7).
G. Simulation Results and analysis In this part, the system simulation results are given out for VoIP traffic with AMR 12.2Kbps by semi-static system simulation. The investigation is carried out on different TTI bundling alternatives, different bundling size and different packet delay budget. Impact from different bundle size Firstly, the VoIP capacity results with different bundled size and but with the fixed packet delay budget (50ms) for both alternatives are concluded in table3 below.
Table 3. VoIP capacity for different bundled size (50ms delay budget) Bundle size (TTIs) No bundling VoIP capacity Alternative 1 per sector Alternative 2 2 <120 148 4 <120 144 135 8 140 106

In terms of the above results, for alternative1 of TTI bundling, the best performance has been achieved with 4 bundle size, which could achieve 144 users per sector. With a short delay budget at air interface (50ms), 2 bundle sizes can only achieve maximum 8 TTIs energy due to the doubled HARQ RTT, which limits the performance of 2 bundle size. On the other hand, both 4 bundle size and 8 bundle size can collect enough TTIs energy. However, the granularity of 8 bundle size is too large to some degree, which will waste resources and result in a little poorer performance.

However alternative2 has the best performance with 2 bundle size which is 148 users per sector. This is because for 2 bundle size, maximum 14 TTIs can be accumulated in 50ms delay budget and at mean while 2 bundled TTIs for alternative 2 has the lowest waste of resources. And for 4 bundled TTIs and 8 bundled TTIs, unnecessary retransmissions combined with large granularity resource allocation result in severe waste of resources and a poorer capacity performance. When TTI bundling is not used for cell edge users, only maximum 7 TTIs' energy can be accumulated within the defined 50ms delay budget. While for alternative1 with 4 bundle size and alternative2 with 2 bundle size, the maximum number of collected TTI can reach 12 and 14 respectively. So 2.34dB~3dB energy accumulation gains can be got, and VoIP capacity can be improved 20% at least by using TTI bundling compared to the case of no bundling. From table3, maximum 20% ~ 23% gain has been obtained, which is nicely fit the 20% energy accumulation gain. Figure8 and figure9 illustrates the retransmission times for alternative1 and alternative2 respectively. For alternative1, we could observe that the residual block Error rate (BLER) with 2 bundle size is highest among three bundle size, which is because the limit energy collection in defined delay budget. Both 4 bundle size and 8 bundle size can make totally 3 transmissions for bundled users. After 3 transmission times, 4 bundle size can collect maximum 12 TTIs energy while for 8 bundle size, maximum 24 TTIs energy can be achieved. Thus the residual BLER of 8 bundle size is much lower than that of 4 bundle size. On the other hand, figure8 tells that after aggregate 4 TTIs energy, about 60% packets can be correctly received, while in 8 bundle size case, this 60% packets still collect 8 TTIs energy, which results in the resource waste. Thus, on the contrary, the performance of 8 bundle size is a little poorer than 4 bundle size. When we turned to alternative2, observation becomes a little different. 2 bundled TTIs wastes fewer resources and utilizes the resources more efficiently. Take the first transmission in figure9 for example, both 2 bundled TTIs and 4 bundled TTIs have ~90% packets to retransmit, which is because the feedback is according to the first transmission in bundle for alternative2. But actually, after 2 TTIs and 4 TTIs accumulated, there are only ~67% and ~40% packets need to be retransmitted (according to alternative 1 in figure8). So the differences are ~23% and ~50%, which are the proportion of packets that wasting resource for 2 bundled TTIs and 4 bundled TTIs in alternative 2. Whats more, 2 bundle size will enable less resource waste than 4 bundled size when one packet needs to be retransmitted. Thus, 2 bundled TTIs could use resource more efficiently, which brought in the capacity gain in results.

This full text paper was peer reviewed at the direction of IEEE Communications Society subject matter experts for publication in the WCNC 2009 proceedings.

be scheduled, which further reduces the number of discarded packets that never been transmitted. The capacity of 70ms delay bound is increased about 8%~13% compared with 50ms delay bound. However for alternative2, increasing delay limit doesnt help a lot because it seems we can get enough energy accumulation already for most of packets within 50 ms delay limit.
Table 4. VoIP capacity for different delay budget Delay budget (ms) Alternative 1 VoIP capacity per sector Alternative 2 50 144 148 60 160 153 70 163 155

V. CONCLUSIONS
Figure 8 Retransmission distributions for alternative1

In this paper, we present the principle of TTI bundling and give a detail evaluation on performance of TTI bundling for VoIP traffic. The related HARQ timing and process design in LTE FDD mode is discussed as well. Semi-static system performance investigation has been carried out for different TTI bundling alternatives with different bundle size and different packet delay budget. The simulation results show that TTI bundling is always a powerful method to solve the coverage problem in LTE system. Compared to no bundling case, about 20% gain can be obtained. Of course, besides VoIP traffic, other services could also use TTI bundling method when the user is encountering with coverage problem. REFERENCES

Figure 9 Retransmission distributions for alternative2

[1] [2] [3] [4]

With a short delay limit (50ms) and 2 bundle size, alternative2 has the a little bit better performance (148 users/sector) compared with 4 bundled TTIs of alternative1 (144 users/sector). On the other hand, the average TTIs that used by correctly received packets are ~5.9 TTIs and ~5.8 TTIs for 2 bundle+alternative2 and 4 bundle + alternative1 respectively, which can be concluded from the figure8 and figure9. So the resource cost for these two cases is similar (only ~2% difference). Thus, with higher maximum accumulate TTIs and similar resource consumption, alternative2 with 2 bundles has a bit better performance with a short 50ms delay budget.
Impact from different delay budget The impact from different delay budget is shown in table4. For alternative1, with a bit loose packet delay limit, VoIP capacity of TTI bundling alternative 1 is increased greatly. This is because the larger packet delay bound will accumulate more energy, e.g. for 50ms delay limit, the packet will cumulate at most 12 TTIs energy with 4 bundled TTIs; However for 60ms delay limit, this number increased to 16 TTIs, which bring in 33% more energy, thus the packet will be correctly decoded more easily. Besides, longer delay limit means each packet has more chance to

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