- Research Article
- Open Access
Efficient Control Channel Resource Allocation for VoIP in OFDMA-Based Packet Radio Networks
© Yong Fan and Mikko Valkama. 2011
- Received: 15 November 2010
- Accepted: 28 February 2011
- Published: 13 March 2011
We propose an efficient control channel resource allocation approach to enhance the performance of voice-over-IP (VoIP) in orthogonal frequency division multiple access- (OFDMA-) based next generation mobile communication systems. As the long-term evolution (LTE) of universal terrestrial radio access network (UTRAN), evolved UTRAN (E-UTRAN) is the first OFDMA-based packet radio network and thus selected in this paper as an application example. Our proposed physical downlink control channel (PDCCH) resource allocation approach for E-UTRAN is composed of bidirectional power control, inner loop link adaptation (ILLA), and outer loop link adaptation (OLLA) algorithms. Its effectiveness is validated through large-scale radio system level simulations, and simulation results confirm that VoIP capacity with dynamic scheduling can be further enhanced with this PDCCH resource allocation approach. Moreover, the VoIP performance requirements for international mobile telecommunications-advanced (IMT-Advanced) radio interface technologies can be met with dynamic scheduling together with proposed PDCCH resource allocation. Besides E-UTRAN, this approach can be introduced to other OFDMA-based mobile communication systems for VoIP performance enhancement as well.
- Power Control
- User Equipment
- Orthogonal Frequency Division Multiple Access
- OFDM Symbol
- Transmission Time Interval
Mobile communication systems are gradually transforming from the systems mainly oriented to voice service to ones that can handle more high data rate services. Although data services are getting more momentum during this transformation, voice service still remains as the main source of revenue for mobile network operators . For savings in capital expenditures and operating expenses, there is also a clear trend that all services are gradually being converged into packet switched (PS) domain. Related work [2–4] on supporting voice service through VoIP protocol in Third Generation cellular network, for example, high speed packet access (HSPA), have already been conducted. The future mobile networks will be completely IP-based and traditional circuit switched (CS) domain will not exist any more in them. Therefore, efficient VoIP support is a fundamental requirement for any emerging new systems.
The next generation mobile communication systems, for example, universal terrestrial radio access network (UTRAN) long-term evolution (LTE) and LTE-Advanced, worldwide interoperability for microwave access (WiMax), and ultra mobile broadband (UMB), all rely on orthogonal frequency division multiple access (OFDMA) as multiple access scheme. Conveying voice service through VoIP protocol in OFDMA-based mobile communication systems inevitably faces challenges caused by inherent VoIP traffic characteristics and stringent quality of service (QoS) criteria. VoIP traffic is low and constant bit-rate traffic, implying small size VoIP packets are transmitted at regular time intervals. The resource allocation for packet transmission is relatively small, thus necessitating multiuser sharing in frequency domain to effectively utilize available wide frequency bandwidth. Moreover, VoIP packets have to be scheduled frequently in order to satisfy the strict packet delay and loss based QoS criteria. In a typical packet radio system, such as evolved UTRAN (E-UTRAN), the transmission of each packet is done with dynamic scheduling by default  and evolved NodeB (eNB) needs to signal scheduled user equipment (UE) the resource allocation information through physical downlink control channel (PDCCH) every scheduling period, for example, one transmission time interval (TTI). As a result, the simultaneous scheduling of multiple users in a frequent manner demands lots of physical layer control signaling resources. The situation on control signaling consumption is further deteriorated by high VoIP capacity demand in PS radio network, which is supposed to accommodate a large amount of voice capacity currently handled in traditional CS mode.
Since the future mobile communication systems also target for high data rate transmission, it is preferable to allocate more radio resources for data transmission, thus leaving few resources for control signaling. Therefore, deploying VoIP in OFDMA-based packet radio systems encounters one unique challenge, that is, to satisfy high VoIP capacity demand with only limited physical layer control signaling resources. In E-UTRAN, PDCCH carries scheduling grants for both downlink and uplink, and its overall design was addressed in . During the E-UTRAN standardization in Third Generation Partnership Project (3GPP), different solutions and schemes [7–11] have already been studied to overcome the limitations of PDCCH resources on VoIP capacity. All these aim at reducing the consumption of PDCCH resources and the possibility of conducting effective PDCCH resource allocation is not considered. The existing UTRAN LTE system level simulation results [9, 12] using simplified PDCCH modeling show that VoIP capacity with dynamic scheduling is limited by available PDCCH resources. The studies [13, 14] using realistic PDCCH modeling further confirm such limiting effect. Compared with delay-tolerant full buffer traffic, the impact of PDCCH limitation on delay-sensitive VoIP traffic is much more severe. While PDCCH limitation causes only marginal loss of cell throughput with full buffer traffic and there is also no evident relationship between cell throughput and the number of users scheduled per TTI , VoIP capacity in E-UTRAN is directly proportional to the number of users scheduled simultaneously in each TTI [13, 14]. On the other hand, it also implies great potential to significantly enhance VoIP capacity if effective PDCCH resource allocation is in place. However, the related study is obviously missing in available open literature.
To meet the ambitious performance target for international mobile telecommunication-advanced (IMT-Advanced) candidate radio interface technologies (RITs) , LTE is evolving into LTE-Advanced, and VoIP performance also needs to be improved. In the LTE-Advanced self-evaluation inside 3GPP, both dynamic scheduling and semipersistent scheduling are considered. While all existing solutions and schemes, such as dynamic packet bundling  and semipersistent scheduling [10, 11], try to reduce consumption of control channel resources, we consider a new paradigm in this paper to overcome limitation of control channel resources and further enhance VoIP performance, that is, to allocate limited control channel resources more effectively so that we can schedule more VoIP users dynamically in each TTI. Our proposed PDCCH resource allocation approach includes bidirectional power control, inner loop link adaptation (ILLA), and outer loop link adaptation (OLLA) algorithms. They are simple yet efficient and can be easily implemented requiring no changes to the existing 3GPP technical specifications.
The remainder of this paper is structured as follows. Section 2 gives a brief introduction of evolved universal terrestrial radio access (E-UTRA), the air interface of UTRAN LTE downlink as well as VoIP QoS and capacity criteria. Section 3 presents the design of proposed PDCCH resource allocation approach and related wideband channel quality indicator ( ) measurement model and dynamic scheduling algorithm. Section 4 describes the overall simulation environment for VoIP on E-UTRA and used VoIP traffic model. The system simulation results in both 3GPP LTE and ITU IMT-Advanced scenarios are examined and analyzed in Section 5 before we draw final conclusions in Section 6.
2.1. E-UTRA Overview
2.2. VoIP QoS and Capacity Criteria
For voice service, the one way end-to-end delay should be below 250 ms to guarantee relatively good voice quality rating . The packet delay includes delay in evolved packet core (EPC) and E-UTRAN. Since the focus of this study is on E-UTRAN part, we deduct packet processing delay and propagation delay of approximately 100 ms in EPC, the delay budget left for E-UTRAN is around 150 ms. We further assume that both end-users are E-UTRAN users, the tolerable delay for medium access control (MAC) queuing and scheduling as well as physical (PHY) transmission should be strictly within 80 ms. To further improve voice quality and better account for variability in network, 3GPP uses 50 ms air interface delay bound (one-way delay from eNB to UE) in LTE system performance evaluation . According to satisfied user criterion (SUC) also specified in , a user is regarded as satisfied if 98% of packets for the user are received successfully within 50 ms air interface delay bound when monitored over the duration of the whole voice call. The VoIP system capacity is further defined as the number of users per cell when more than 95% of the users in the cell are satisfied.
3.1. PDCCH Overview
PDCCH carries downlink scheduling assignments and uplink scheduling grants. The information fields in the scheduling assignments and scheduling grants include resource allocation information, transport format information, and HARQ information. Since there is no HARQ mechanism in PDCCH and the failure of PDCCH transmission will directly lead to failure of subsequent PDSCH transmission, the BLER target is set to 1% for the purpose of reliable PDCCH transmission. The MCS for transmission of PDCCH payload is QPSK modulation with four different effective coding rates of 2/3, 1/3, 1/6, and 1/12, corresponding to aggregation level of 1, 2, 4, 8 CCEs, respectively. A user in favorable radio channel conditions may require just 1 CCE with QPSK 2/3. Adaptive coding can be used to aggregate 2, 4, or even 8 CCEs with lower effective coding rate, thus improving the PDCCH coverage for users in worse radio channel conditions.
3.2. Measurement Model
where round( ) is the operation of rounding to the nearest integer. Finally, the quantized wideband s are reported to eNB with specified reporting schemes subject to certain feedback delay.
3.3. PDCCH Power Control and ILLA Algorithms
Previous study  on PDCCH resource allocation for full buffer traffic proposed a power boost algorithm. It artificially lowers CCE aggregation level by adding an initial power offset on user's wideband before determining the required number of CCE. However, using high power offset will quickly run out of power while on the contrary CCE will easily be depleted if low power offset is applied. Therefore, it is hardly optimal in both power domain and CCE domain. In this study, we come up with a bidirectional power control algorithm to effectively balance the allocation of PDCCH power and CCEs, thus maximizing the number of dynamically scheduled users per TTI. The proposed PDCCH resource allocation can be divided into three steps and functions as follows.
Estimate CCE aggregation level: the estimated CCE AL is determined by choosing the minimum required number of CCEs satisfying equation , where SINRWB is user's wideband taking into account measurement error and quantization effect as well as feedback delay, is the required to reach the PDCCH target BLER of 1% with .
where is the required number of REs for user, is the number of OFDM symbols reserved for PDCCH per TTI, and is the size of one CCE in terms of the number of REs.
is the maximum amount of power allowed to be reduced for each user and is the maximum amount of power allowed to be boosted for each user. Their values are set within the dynamics of eNB power amplifier and the absolute value of is larger than that of to enable conservative power boost and aggressive power reduction.
Balance allocation of CCE and power: depending on whether there is still CCE or power left, the algorithm enters into the final step to balance CCE and power allocation. If there is power left, the algorithm will try to boost the power of user who originally cannot reach CCE in Step 2 due to the limitation from . If there is CCE left, the algorithm will try to reduce the power of user who is originally boosted to CCE in Step 2.
3.4. PDCCH OLLA Algorithm
Unlike PDSCH which can directly apply such type of OLLA, PDCCH cannot do so due to the lack of acknowledgment mechanism in PDCCH. Therefore, we rely on PDSCH OLLA to indirectly control the average BLER for PDCCH. The compensation offset factor generated from PDSCH OLLA is then scaled by a factor to compensate the difference between BLER targets for PDSCH and PDCCH transmission. After that, the sum of these two factors is fed into PDCCH ILLA algorithm as for CCE aggregation level estimation.
3.5. Dynamic Scheduling Algorithm
A VoIP optimized dynamic scheduling algorithm  is used together with proposed PDCCH resource allocation method. The algorithm is separated into two parts: selecting users and assigning PRBs to users. In the user selection, we introduce a two-step scheduling candidate set (SCS) approach to efficiently control the interuser fairness as well as directly address the specific requirements for VoIP traffic scheduling.
For interuser fairness control, all the schedulable users on the basis of eNB buffer status are first sorted according to a relative wideband metric and then organized into primary SCS. Instead of using traditional proportional fair metric based on user throughput, we come up with this new relative wideband metric for interuser fairness control. The relative wideband is defined as , where is instantaneous wideband and represents the average wideband in the past. is calculated with traditional recursive method  by replacing user throughput with value. Relative wideband reflects user's instantaneous channel condition relative to the average and therefore indicates a favorable time instant for scheduling.
users with pending retransmission,
users whose delay of the oldest unsent packet in the eNB buffer is close to given threshold, and
remaining users other than the above two groups.
In PDSCH, there is no power control mechanism and each PRB is assigned with the same power level. The number of PRB allocated to each user is determined by adaptive approach based on user's channel condition. As for assigning PRBs to selected users, we give priority to first transmission users in choosing PRBs with relatively favorable channel quality and grant the retransmission users the priority to reserve the same amount of PRBs as in the original transmissions. By doing so, a high successful rate of first transmission can be achieved and then packet transmission delay over the air interface is reduced. Besides, retransmission users can anyway benefit from HARQ combining gain for correct reception, although they are given less freedom to choose suitable PRBs.
4.1. Overall Simulation Environment
Main simulation parameters and values.
3GPP LTE: Macro Case 1 and 3
ITU IMT-A: indoor hotspot, urban microcell,
urban macrocell, and rural macrocell
Asynchronous adaptive with chase combining
Num. of stop-and-wait processes: 8
Max. number of retransmissions: 3
Measurement time duration: 2 ms
Measurement subband bandwidth: 360 KHz
Narrowband error standard deviation: 1 dB
Wideband error standard deviation: 0.2 dB
Quantization step: 1 dB
Feedback delay: 2 ms
Measurement period: 5 ms
Transmission power: 12 W (Macro scenario)
Num. of data OFDM symbols per TTI: 11
Num. of PRBs: 25
Num. of subcarriers: 300
BLER target: 10%
OLLA step up: 0.5 dB
OLLA step down: 0.05 dB
Max. OLLA step up: 3 dB
Max. OLLA step down: −1 dB
Transmission power: 8 W (Macro scenario)
Num. of control OFDM symbols per TTI: 3
Num. of CCEs (REs): 10 (360)
BLER target: 1%
Max. power increase: +2 dB
Max. power decrease: −4 dB
4.2. VoIP Traffic Model
VoIP traffic characteristics.
AMR voice codec rate
Voice call length
Voice activity factor
Mean of active period duration
Mean of silence period duration
Voice/SID packet size
40 bytes/15 bytes
Voice/SID packet interarrival time
20 ms/160 ms
This section presents simulated system performance of VoIP on E-UTRA in both 3GPP LTE and ITU IMT-A deployment scenarios. The VoIP capacity is evaluated according to SUC criteria described in Section 2.2. All the figures below are plotted with system load corresponding to 95% satisfaction point.
5.1. Capacity Enhancement with Proposed PDCCH Resource Allocation
3GPP LTE deployment scenarios.
Macro case 1
Macro Case 3
5.1.1. VoIP Capacity Gain Analysis
One observation from Figures 4 and 5 is that power control gain in Case 3 is relatively higher than that in Case 1, either without bundling or with bundling. In Case 3, users experience lower on average due to larger intersite distance and thus consume more PDCCH resources than those in Case 1. Therefore, users in Case 3 benefit more from PDCCH power control.
5.1.2. PDCCH and PDSCH Resource Utilization Analysis
With proposed PDCCH power control, the resource allocation is conducted on two dimensions: CCE and power. CCE is still allocated in discrete quantities, but power is instead allocated in continuous quantities, as can be observed from Figure 9. The over-allocated power on user with CCE aggregation level of 8, 4, 2 is used to exchange for reduction of CCE aggregation level to 4, 2, 1. The saved power from users consuming 1 CCE can also be used for this purpose. As a result, not only the total number of CCEs used per TTI can be decreased, but also the assigned power per TTI can be reduced. As also observed from Figure 9, the amount of used power and CCEs with power control is well below that without power control, although the achieved capacity is 40–46% higher. The average utilization rate of CCEs and power is approximately 80%, indicating the efficient balance between CCE and power allocation can also be achieved with this algorithm.
5.1.3. VoIP Packet Delay Analysis
5.2. Capacity Evaluation for IMT-Advanced
ITU IMT-Advanced deployment scenarios.
Base coverage urban
UE speed of interest
The simulations are conducted using the same quasistatic system level simulator, according to guidelines for evaluation of radio interface technologies for IMT-Advanced  as well as additional modeling and assumptions for different ITU deployment scenarios . ITU simulation methodology is slightly different from 3GPP counterpart mainly in radio channel model and link budget setting. More details can be found in [26, 27]. Except that InH scenarios use a two-isolated cell layout, all the other three scenarios use the same simulated network layout as that used in 3GPP LTE simulations. VoIP traffic model, QoS criteria, link-to-system mapping model, and other simulation assumptions related to HARQ and are the same as those used in the simulation of 3GPP LTE scenarios.
VoIP capacity requirement in IMT-Advanced and simulated results.
Minimum required capacity
In this paper, we studied PDCCH resource allocation method oriented for VoIP optimized dynamic scheduling. A comprehensive set of system level simulation results were demonstrated, covering both 3GPP LTE scenario and ITU IMT-Advanced scenario. A thorough analysis of capacity gain, PDCCH and PDSCH utilization rate, and packet delay performance were provided. Postsimulation analysis further confirms the effectiveness of proposed PDCCH resource allocation method. The proposed bidirectional power control algorithm can efficiently balance the allocation of CCE and power in PDCCH and maximize the number of users scheduled per TTI, thus greatly improving the capacity of VoIP with dynamic scheduling. The relative capacity gain is approximately 31–46% over that without power control, depending on deployment scenario and whether packet bundling is enabled or not. Besides, PDCCH resource allocation does not impose negative impact on packet bundling level and packet delay performance. Moreover, in all four ITU IMT-Advanced deployment scenarios, enhanced VoIP capacity exceeds the target set in IMT-Advanced. This approach can also be applied in semipersistent scheduling as the retransmissions and SID transmissions in semipersistent scheduling also needs control signaling. Although we use UTRAN LTE downlink as an application example in this paper, the proposed PDCCH resource allocation method can be naturally applied to UTRAN LTE uplink as well as other OFDMA-based mobile communication systems for VoIP performance enhancement.
The authors appreciate the support from many colleagues, especially Markku Kuusela from Nokia Devices R&D, Helsinki and Petteri Lundén from Nokia Research Center, Helsinki in their study of VoIP on E-UTRA.
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