VoIP Dimensioning FDD RECOMMENDATION 12/100 56-HSC 105 50/1-T1 Uen M Copyright © Ericsson AB 2011-21. All rights reserved. No part of this document may be reproduced in any form without the written permission of the copyright owner. Disclaimer The contents of this document are subject to revision without notice due to continued progress in methodology, design and manufacturing. Ericsson shall have no liability for any error or damage of any kind resulting from the use of this document. Trademark List All trademarks mentioned herein are the property of their respective owners. These are shown in the document Trademark Information. 12/100 56-HSC 105 50/1-T1 Uen M | 2021-02-23 Contents Contents 1 Introduction 1 1.1 Limitation 1 1.2 Concepts 1 2 Technical Background of VoIP 3 2.1 Quality Requirements 3 2.2 VoIP Characteristics 3 2.3 Functionality Used by VoIP 5 3 VoIP Coverage 13 3.1 Introduction 13 3.2 Traffic Model 13 3.3 Link Adaptation 13 3.4 Link Performance for VoIP 13 3.5 Uplink Link Budget 15 3.6 Downlink Link Budget 17 4 Link Budget Example 19 4.1 Prerequisites 19 4.2 Uplink Coverage 19 4.3 Downlink Coverage 20 5 VoIP Capacity 23 5.1 Introduction 23 5.2 Capacity for Different Codecs 23 5.3 VoIP Capacity Limitations 25 5.4 PDCCH Allocation 26 5.5 Capacity Consideration in Mixed Scenarios 28 12/100 56-HSC 105 50/1-T1 Uen M | 2021-02-23 VoIP Dimensioning 12/100 56-HSC 105 50/1-T1 Uen M | 2021-02-23 Introduction 1 Introduction LTE is designed to support data services efficiently, and good quality voice service with high efficiency. The intention of this document is to outline the basic concepts needed to dimension LTE in terms of Voice over IP (VoIP). First, the general quality requirements for voice and the typical voice codecs are introduced. A list of some VoIP-related features are presented. This document also presents the voice uplink and downlink coverage challenges and solutions. A method is presented for basic calculation of VoIP coverage and capacity using LTE technology. The method presented in this document consists of concepts and mathematical calculations that are elements of the general dimensioning process. 1.1 Limitation The document is valid for the current release of LTE. The following topics are not covered in this guideline: — Explicit Congestion Notification (ECN) — Circuit Switched Fallback (CSFB) — Quality of Service (QoS) procedure handling Mobility is not covered in this document, see instead Deployment Guideline for VoIP and Conversational Video. 1.2 Concepts Concepts used in this document include the following: GBR bearer A bearer with reserved (guaranteed) bit rate resources non-GBR bearer A bearer with no reserved (guaranteed) bit rate resources Payload The amount of transmitted data excluding headers transported in a certain protocol unit VoIP A technology that allows telephone calls to be made over computer networks using Internet Protocol (IP) 12/100 56-HSC 105 50/1-T1 Uen M | 2021-02-23 1 VoIP Dimensioning 2 12/100 56-HSC 105 50/1-T1 Uen M | 2021-02-23 Technical Background of VoIP 2 Technical Background of VoIP With the emergence of Internet Protocol (IP) as the protocol of choice for carrying all types of traffic, LTE provides support for IP-based traffic with end-to-end Quality of Service (QoS). Voice traffic will be supported as VoIP enabling better integration with other multimedia services. This section takes a closer look at service requirements, service mapping, and how the VoIP service is realized for the LTE RAN. 2.1 Quality Requirements The performance in terms of quality for VoIP is determined mainly by three factors: — Codec bit rate — Delay caused by scheduling and retransmissions on the air interface — Frame Error Ratio (FER) for RLC frames Using a lower codec bit rate gives higher VoIP capacity, especially in large cells. User satisfaction can be measured in terms of mouth-to-ear delay. According to 3GPP, 200 ms mouth-to-ear delay corresponds to 50 ms radio interface delay. Table 1 lists the impact of delay in accordance with ITU-T G.114. Table 1 2.2 Impact of Delay According to ITU Mouth-to-ear delay User acceptance Below 200 ms Users very satisfied Between 200 ms and 285 ms Users satisfied Between 285 ms and 390 ms Some users dissatisfied VoIP Characteristics In LTE, voice is coded to VoIP packets every 20 ms with Adaptive Multi-Rate (AMR) codec with one of eight different rates for narrow-band and nine different rates for wide-band according to 3GPP standard. Table 2 lists the size of the payload for some typical AMR codecs and for Enhanced Variable Rate Codec (EVRC). Table 3 lists the size of the protocol headers at layer 2. Table 2 Codec Payload Protocol Payload AMR wide-band 23.85 kbps 61 octets AMR wide-band 12.65 kbps 33 octets 12/100 56-HSC 105 50/1-T1 Uen M | 2021-02-23 3 VoIP Dimensioning Protocol Payload AMR wide-band 8.85 kbps 24 octets AMR wide-band 6.6 kbps 18 octets AMR 12.2 kbps 32 octets AMR 7.95 kbps 22 octets AMR 5.9 kbps 16 octets AMR 4.75 kbps 14 octets EVRC 8.8 kbps 22 octets (1) SID 7 octets (1) Silence Insertion Description Table 3 Codec Headers Protocol Header (1) 2 octets RLC UM header 1 octet PDCP MAC BSR (2) (3) PHR (4) RTP/UDP/IP header 1 octet (basic header) 2 octets (header in MAC if present) 2 octets 1–60 octets (1) Packet Data Convergence Protocol (2) Medium Access Control (3) Buffer Status Report (4) Power Headroom Report The header sizes in Table 3 reflect the lengths of the PDCP and RLC Sequence Numbers (SNs) as specified in 3GPP TS 36.322: Radio Link Control (RLC) protocol specification and 3GPP TS 36.323: Packet Data Convergence Protocol (PDCP) specification. A PDCP SN length of 12 bits requires a header size of 2 octets. Robust Header Compression (ROHC) is a method for compressing the headers described in Table 3 and Section 2.3.2 on page 6. The IP header with ROHC will be between 1 and 60 octets, with the vast majority of the IP packets at 3 octets. The IP header without ROHC will be 40 octets (IPv4) or 60 octets (IPv6). Cyclic Redundancy Check (CRC) bits in layer 1 are considered by the receiver during decoding. This means that there are 24 more information bits than the actual size of the transport block. As described in Table 2, one VoIP packet with an AMR 12.2 codec with ROHC will have the Transport Block Size (TBS) as calculated in Example 1. Find the number of information bits for a VoIP packet using AMR 12.2 and ROHC. Example 1 Answer: 4 12/100 56-HSC 105 50/1-T1 Uen M | 2021-02-23 Technical Background of VoIP Payload and headers are given by Table 2 and Table 3. [32(AMR 12.2) +3(ROHC) + 8(PDCP, RLC, MAC, BSR, PHR header)]*8 = 344 bits. The desired transport block size of 344 is rounded up till 376 as listed in 3GPP TS 36.213: Physical layer procedures, provided that two Resource Blocks (RBs) are used. The total number of information bits including CRC: 376 + 24 = 400 bits. Find the effective layer 1 bit rate for a VoIP packet using AMR 12.2 and ROHC. Example 2 Answer: The total number of information bits including CRC: 376 + 24 = 400 bits. A VoIP packet is transmitted every 20 ms. The effective layer 1 bit rate: 400/20 = 20 kbps. 2.3 Functionality Used by VoIP The following VoIP related features are considered in the dimensioning method. — VoIP Bearer — ROHC — Segmentation — TTI Bundling — Frequency Hopping — PDCCH Coverage Extension — Radio Link Control (RLC) Unacknowledged Mode (UM) — Delay Based Scheduler (DBS) — Grant Estimation These features are included in the current release. 2.3.1 VoIP Bearer The ‘‘bearer’’ is a central element of the Evolved Packet System (EPS) QoS concept. The network-initiated QoS control paradigm specified in EPS is a set of signalling procedures for managing bearers and controlling the QoS assigned by the network. The EPS QoS concept is class based, where each bearer is assigned one and only one QoS Class Identification (QCI) by the network. QCI is a scalar 12/100 56-HSC 105 50/1-T1 Uen M | 2021-02-23 5 VoIP Dimensioning that is used within the RAN as a reference to node-specific parameters that control packet forwarding treatment. Two types of bearers exist: — Guaranteed Bit Rate (GBR) — Non-guaranteed Bit Rate (non-GBR) Table 4 shows an extract of the 3GPP table that defines the 3GPP TS 23.203: Policy and charging control architecture QoS classes. Table 4 QCI 1 7 QCI Characteristics for Voice Services Standardized in 3GPP Resource Type GBR Non-GBR Priority 2 7 (1) PELR 100 ms 10-2 Conversational Voice 100 ms 10-3 Voice, Video (Live Streaming) Interactive Gaming PDB (2) Example Services (1) Packet Delay Budget (2) Packet Error Loss Rate The PDB includes an average of 20 ms for the delay between a Policy and Charging Enforcement Function (PCEF), an entity in the Packet Data Network Gateway (PGW), and a radio base station. Hence the PDB that applies to the radio interface will be 80 ms for the services in Table 4. VoIP is mapped to a GBR data radio bearer with specific QoS, which means that it is been given priority over other services like for example TCP-based traffic. 2.3.2 ROHC With ROHC, the RTP/UDP/IP headers are compressed from 40 bytes (IPv4) or 60 bytes (IPv6) to between 60 and 1 bytes as shown in Figure 1. The most important factor for high VoIP capacity is ROHC functionality, especially in the uplink. ROHC is also increasing the coverage. 24 bytes IP header Payload IP header Payload 3 bytes L0000414A Figure 1 6 ROHC Reduces the Size of the VoIP Packets 12/100 56-HSC 105 50/1-T1 Uen M | 2021-02-23 Technical Background of VoIP 2.3.3 Delay In the LTE network, the Transmission Time Interval (TTI) is short. This can lead to a specific access problem: if the UE is power limited, it cannot accumulate enough energy during one TTI. To solve this problem, many Hybrid Automatic Repeat Request (HARQ) retransmissions are required. The HARQ process must wait for a reply to the first transmission (negative acknowledgement) before sending additional retransmissions. Many retransmissions lead to long delays, which can be intolerable for delay-sensitive applications such as VoIP. Give an example of the number of retransmissions and the delay incurred when close to a realistic delay budget of 80 ms. Example 3 Answer: Assume that the HARQ Round Trip Time (RTT) is 8 ms. A single transport block can suffer ten retransmissions, 8*10 ms = 80 ms before the delay budget is exceeded. The following are two solutions for reducing the number of retransmissions and thus reduce the delay: — Segmentation of large packets into smaller. Smaller transport blocks include less information bits and, due to stronger coding, they can be correctly decoded with smaller amount of retransmissions, which reduces the delay — TTI Bundling. Each bundle requires only one collective acknowledgement, which reduces the overhead and thus the delay (in current release only available for FDD) 2.3.4 Segmentation Segmentation is one of the methods for reducing the number of retransmissions. RLC SDUs are segmented at the RLC layer and the resulting segments transmitted in subsequent TTIs. When a RLC SDU is divided into many segments, each transport block has its own RLC/MAC header. The transport blocks are transmitted in consecutive TTIs using different HARQ processes. The size of each RLC/MAC header is at least 8 bits but can be more. In addition, layer 1 adds 24 bits of CRC to each transport block. This means that segmentation increases the number of bits used for MAC and RLC headers as well as for CRC. Figure 2 shows an example of segmentation of an AMR 12.2 kbps VoIP packet. 12/100 56-HSC 105 50/1-T1 Uen M | 2021-02-23 7 VoIP Dimensioning 288 bits payload from PDCP RLC layer segmentation RLC/MAC/BSR/PHR header CRC attachment RLC BSR header header 8 8 16 16 MAC PHR header header RLC header 144 bits 24 bits 8 CRC MAC header 8 144 bits 24 bits CRC L0000861A Figure 2 Segmentation of an AMR 12.2 Kbps Packet in a Compressed IP Packet Size of 288 Bits With AMR 12.2, the RLC SDU is roughly 288 bits in the PDCP layer. If the RLC SDU is segmented, each sub-packet will attach a 1 byte RLC header, a 1 byte MAC header, and a 3 byte CRC overhead. When the RLC SDU is segmented to two sub-packets, the extra overhead due to segmentation is about 11.1% due to 2 more header bytes and 3 bytes CRC. If segmented to 4 sub-packets, this segmentation overhead is 33.3%. This overhead percentage will be even higher for VoIP traffic with AMR 7.95 kbps because of the smaller payload size used. When scheduling and transmitting an uplink sub-packet, an uplink scheduling grant is required, thus consuming one Physical Downlink Control Channel (PDCCH). Assuming that a RLC SDU is segmented to two sub-packets, the number of consumed PDCCH channels is doubled. Considering that power limited users are often under poor channel conditions, one PDCCH channel will use up more control channel resources as described in Control Channel Guideline, which will use up the system control channel resources and limit the opportunity to schedule other users. Segmentation is built into the system in the link adaptation procedure. The link adaptation will search for the number of segments and TBS that maximizes the robustness of the transmission. 2.3.5 TTI Bundling The second method to reduce the delay is TTI bundling. TTI bundling uses automatic retransmissions in several consecutive TTIs without waiting for HARQ feedback, resulting in a longer effective TTI length. The set of consecutive uplink TTIs is called a bundle. The number of consecutive uplink TTIs is the bundle size. Bundling must be activated by Radio Resource Control (RRC) signalling. The 8 12/100 56-HSC 105 50/1-T1 Uen M | 2021-02-23 Technical Background of VoIP bundle size is always equal to four according to the 3GPP standard. Only one uplink grant and one HARQ feedback channel is transmitted for a bundle. In TTI bundling, the packet is autonomously retransmitted. All retransmissions are sent together in a bundle without waiting for an Acknowledgement (ACK). Each retransmission has a different Redundancy Version (RV). Figure 3 shows the process of TTI bundling operation for AMR 12.2 kbps. 288 bits payload from PDCP RLC/MAC/BSR/PHR header CRC attachment RLC BSR header header 8 8 16 16 288 bits 24 bits MAC PHR header header CRC Channel Coding HARQ funcionality N RBSs with RV0 initial transmission K N RBSs with RV2 auto-retransmission RV3 RV1 K+1 K+2 K+3 L0000862A Figure 3 Example of TTI Bundling Process for AMR 12.2 kbps and Transport Block Size of 288 Bits TTI Bundling assumes that the bundle size is fixed and that the receiver waits the total transmission period (bundle of TTIs) before sending feedback. The benefit of this approach is that the transmitter knows when the feedback will arrive. Scheduling is quite straightforward since the number of retransmissions in a bundle is known. In LTE, due to physical and processing delays, it is assumed that the delay from the transmission of one RV until the start of the next retransmission, also known as HARQ RTT, is 8 ms without using TTI bundling. If many retransmissions are bundled together, a new bundle is not possible until 8 ms from the last retransmission of the previous bundle. Thus the total RTT of bundling is longer than 8 ms. In TTI bundling, each new bundle is started after 16 ms from the previous one leading to a pattern that is synchronous with the 8 ms HARQ RTT. The delay of 16 TTI is specified by 3GPP. The time slots in between can be used for other bundled HARQ processes from the given User Equipment (UE) or other UE. 12/100 56-HSC 105 50/1-T1 Uen M | 2021-02-23 9 VoIP Dimensioning The benefit of TTI bundling is increased coverage, because the MAC/RLC layer overhead, as well as the number of different control messages, is decreased, leading to improved performance of the uplink. Also, the higher layer PDU can be decoded even if one TTI fails. The increased coverage is typically between 2 and 5 dB, depending on channel conditions, compared to segmentation. TTI Bundling is a licensed feature that is dynamically activated when the SINR value measured by the UE falls below a threshold. 2.3.6 VoLTE Frequency Hopping With VoLTE Frequency Hopping, support for frequency hopping is introduced on the PUSCH to improve the VoIP coverage in uplink. Frequency hopping is only used when TTI bundling is triggered, at low SINR close to the cell border. VoLTE Frequency Hopping is a feature that is licensed per node. 2.3.7 PDCCH Coverage Extension With PDCCH Coverage Extension, the coverage of the PDCCH will be improved with up to 1.7 dB depending on bandwidth. This leads to improved VoIP coverage in downlink. PDCCH Coverage Extension is part of the LTE basic package. 2.3.8 RLC Unacknowledged Mode RLC Unacknowledged Mode (UM) does not perform any error recovery. It supports concatenation and segmentation and attempts to establish the original order of PDUs and SDUs. Depending on the traffic characteristics, if the number of transmissions per second is limited for example by PDCCH resources, RLC UM significantly increases the system capacity. RLC UM saves 8 bits overhead in each transport block compared with AM. RLC UM is a feature that is licensed per node. In order to activate the feature, the parameter rlcMode is set to UM. 2.3.9 DBS and Grant Estimation DBS is a scheduler that is particularly suitable for a service such as VoIP, where the experienced quality depends on the delay of the IP-packets. The weight of the DBS is based on the delay of the packet. For the LTE uplink, the scheduler is located at the eNodeB and the buffers are in the terminal. Therefore, the scheduler has no direct knowledge of the amount of data available in the terminal. A possibility is that the terminal can include a BSR (Buffer Status Report) together with a data transmission on PUSCH to inform the scheduler of the size of each priority queue in the terminal. However, Grant 10 12/100 56-HSC 105 50/1-T1 Uen M | 2021-02-23 Technical Background of VoIP Estimation together with BSR is increasing the VoIP capacity by doing a better estimation of the UE buffer than can be done with BSR only. Grant Estimation estimates the amount of data in the UE buffer by using the VoIP characteristics. Furthermore, it estimates the age of the data in the UE buffer, which is required for the DBS. There is one license controlling DBS and Grant Estimation. 12/100 56-HSC 105 50/1-T1 Uen M | 2021-02-23 11 VoIP Dimensioning 12 12/100 56-HSC 105 50/1-T1 Uen M | 2021-02-23 VoIP Coverage 3 VoIP Coverage This section describes VoIP coverage. 3.1 Introduction The method to calculate VoIP coverage presented here is based on Layer 1 link simulations. These simulations have been verified against lab measurements. 3.2 Traffic Model The coverage dimensioning method for VoIP is developed for a traffic model consisting of only one VoIP user, for which the coverage is calculated. The interference is created by a background traffic consisting of mobile broadband traffic. 3.3 Link Adaptation The different LTE transport formats (modulation, coding scheme, number of RBs, and TBS) have different Signal to Interference and Noise Ratio (SINR) requirements. The link adaptation in the RBS determines the transport format based on the estimated SINR. The RLC protocol will be given a transport block with the given size. If the transport block selected is smaller than the VoIP packet waiting in the buffer, the packet will be segmented into smaller parts. In uplink, the maximum power in the UE is limited. For a user at the cell border, the maximum transmit power will be used. The link adaptation will use a minimum allocation of two RBs in order to maximize the power spectral density. For dimensioning purposes, it is assumed that the user is at the cell edge and will use the maximum transmit power and the minimum allocation of two RBs. 3.4 Link Performance for VoIP The VoIP link performance is given as the corresponding minimum SINR requirement for a UE on the cell border. The following settings were used in the simulations to determine the minimum SINR requirement: — Bandwidth 10 MHz — 100 ms buffer size in the UE — FER equal to 1% 12/100 56-HSC 105 50/1-T1 Uen M | 2021-02-23 13 VoIP Dimensioning — 2 receive antennas — With and without TTI bundling of 4 subframes — 4 HARQ transmissions per bundle (16 ms transmission time per bundle in total) with TTI bundling — 4 HARQ transmissions without TTI bundling — ROHC — DBS — With and without frequency hopping Table 5, Table 6 and Table 7 show the resulting minimum desired SINR level in uplink for various voice codecs and for low correlation between antennas. The results are presented for the three channel model-Doppler shift combinations of the link simulations presented in Coverage and Capacity Dimensioning. Table 5 shows the resulting minimum desired SINR level in uplink for various voice codecs and without TTI bundling. Table 5 Uplink SINR Values for VoIP with ROHC and without TTI Bundling Uplink SINR [dB] Codec EPA5 EVA70 ETU300 AMR wide-band 23.85 kbps 1.8 -3.4 -3.4 AMR wide-band 12.65 kbps 1.4 -3.9 -3.7 AMR wide-band 8.85 kbps 0.9 -3.9 -3.9 AMR wide-band 6.6 kbps 0.8 -4.1 -4.0 AMR 12.2 kbps 1.4 -3.9 -3.7 AMR 7.95 kbps 0.9 -3.9 -3.9 AMR 5.9 kbps 0.8 -4.1 -4.0 AMR 4.75 kbps 0.7 -4.3 -4.2 EVRC 8.8 kbps 0.9 -3.9 -3.9 Table 6 shows the resulting minimum desired SINR level in uplink for various voice codecs and with TTI bundling. Table 6 Uplink SINR Values for VoIP with ROHC and TTI Bundling Uplink SINR [dB] Codec 14 EPA5 EVA70 ETU300 AMR wide-band 23.85 kbps -4.0 -5.8 -5.9 AMR wide-band 12.65 kbps -4.3 -8.2 -8.4 12/100 56-HSC 105 50/1-T1 Uen M | 2021-02-23 VoIP Coverage Uplink SINR [dB] Codec EPA5 EVA70 ETU300 AMR wide-band 8.85 kbps -4.4 -8.6 -9.1 AMR wide-band 6.6 kbps -4.2 -8.5 -9.1 AMR 12.2 kbps -4.3 -8.2 -8.4 AMR 7.95 kbps -4.4 -8.6 -9.1 AMR 5.9 kbps -4.4 -8.6 -9.1 AMR 4.75 kbps -4.2 -8.5 -9.1 EVRC 8.8 kbps -4.4 -8.6 -9.1 Table 7 shows the resulting minimum desired SINR level in uplink for various voice codecs and for VoLTE Frequency Hopping. Table 7 Uplink SINR Values for VoIP with VoLTE Frequency Hopping, ROHC and TTI Bundling Uplink SINR [dB] Codec 3.5 EPA5 EVA70 ETU300 AMR wide-band 23.85 kbps -6.4 -6.9 -5.9 AMR wide-band 12.65 kbps -7.4 -9.3 -8.5 AMR wide-band 8.85 kbps -7.5 -9.7 -9.2 AMR wide-band 6.6 kbps -7.6 -9.7 -9.3 AMR 12.2 kbps -7.4 -9.3 -8.5 AMR 7.95 kbps -7.5 -9.7 -9.2 AMR 5.9 kbps -7.6 -9.7 -9.3 AMR 4.75 kbps -7.4 -9.7 -9.3 EVRC 8.8 kbps -7.5 -9.7 -9.2 Uplink Link Budget Calculating a link budget for VoIP is similar to the method used for a mobile broadband service, see Coverage and Capacity Dimensioning. The main difference is how the SINR value in the RBS sensitivity is obtained. The uplink link budget, Lpmax , is given by Equation 1: Lpmax = PUE;RB 0 SRBS;RB 0 BIUL 0 BLN F 0 LBL 0 LCP L 0 LBP L + Ga 0 LJ Equation 1 Uplink Link Budget where 12/100 56-HSC 105 50/1-T1 Uen M | 2021-02-23 15 VoIP Dimensioning PUE;RB is the UE output power per resource block [dBm/RB] SRBS;RB is the RBS sensitivity per resource block [dBm/RB] BLNF is the log-normal fading margin [dB] BIUL is the uplink interference margin (‘‘noise rise’’) [dB] LBL is the body loss [dB] LCPL is the car penetration loss [dB] LBP L is the building penetration loss [dB] Ga is the antenna gain [dB] LJ is the jumper loss [dB] The RBS sensitivity SRBS;RB is calculated with Equation 2: SRBS;RB = Nt + Nf + 10 log (WRB ) + Equation 2 = NRB;UL + [dBm=RB ] RBS Sensitivity per Resource Block where Nt is the thermal noise density [dBm/Hz] Nf is the noise factor in the RBS [dB] WRB is the bandwidth of one resource block (180 kHz) NRB;UL is the thermal noise per resource block in uplink [dBm/RB] is the minimum desired SINR level for VoIP [dB] (Section 3.4 on page 13) The uplink interference margin BIUL arises from a background mobile broadband traffic service in accordance with the example in Coverage and Capacity Dimensioning. With antenna arrangements including more than two receive antennas for the uplink, the minimum desired SINR levels in Section 3.4 on page 13 are relaxed. Table 8 presents the gain values for different antenna arrangements to be used in link budget calculations for uplink. Table 8 16 Gain Values for Different Antenna Arrangements in Uplink Antenna arrangement Transmission mode Gain [dB] 1x4 SIMO 3 1x8 SIMO 6 12/100 56-HSC 105 50/1-T1 Uen M | 2021-02-23 VoIP Coverage The gain value in Table 8 is subtracted from the SINR value in Equation 2 to obtain the final RBS sensitivity per resource block SRBS;RB . 3.6 Downlink Link Budget The method to obtain the receiver sensitivity in the UE, SUE , is analogous to that of the uplink SRBS by way of downlink variants of Equation 1 and Equation 2: Lpmax = Ptx;RB 0 SUE;RB 0 BIDL 0 BLNF 0 LBL 0 LCPL 0 LBP L + Ga 0 LJ Equation 3 Downlink Link Budget where Ptx;RB is the transmitter power per resource block at the TX reference point [dBm] SU E;RB is the UE sensitivity per resource block [dBm] BIDL is the downlink interference margin [dB] SUE;RB = Nt + Nf;UE + 10 log (WRB ) + Equation 4 = NRB;DL + [dBm=RB ] UE Sensitivity per Resource Block where Nf;U E is the noise factor in the UE [dB] NRB;DL is the thermal noise per resource block in downlink [dBm/RB] The downlink interference margin BIDL arises from a background mobile broadband traffic service in accordance with the example in Coverage and Capacity Dimensioning. Table 9 presents the gain values for different antenna arrangements to be used in link budget calculations for downlink. Table 9 Gain Values for Different Antenna Arrangements in Downlink Antenna arrangement Transmission mode Gain [dB] 1x4 SIMO 3 2x2 TxDiv, OLSM, CLSM 0 2x4 TxDiv, OLSM, CLSM 3 4x2 TxDiv, OLSM, CLSM 0 4x4 TxDiv, OLSM, CLSM 3 12/100 56-HSC 105 50/1-T1 Uen M | 2021-02-23 17 VoIP Dimensioning The gain value in Table 9 is subtracted from the downlink SINR value in Equation 4 to obtain the final UE sensitivity per resource block, SUE;RB . In downlink the PDCCH is assumed to be the limiting link. The SINR values to use are obtained from Table 12 in Chapter 6, in Coverage and Capacity Dimensioning by considering the performance for PDCCH. The maximum downlink path loss value is used for determining whether the uplink VoIP channel or the downlink PDCCH channel is the limiting link. If the maximum path loss value for the downlink PDCCH channel is lower than the path loss value for uplink VoIP, the downlink is the limiting link. The true cell range is then determined from the maximum path loss value for downlink. 18 12/100 56-HSC 105 50/1-T1 Uen M | 2021-02-23 Link Budget Example 4 Link Budget Example This section gives an example of a link budget calculation. Realistic input criteria are used and fed into an uplink coverage calculation and a downlink coverage calculation. 4.1 Prerequisites The prerequisites are as follows: — AMR 12.2 codec — RBS capability: 40 W output power; 2x2 MIMO (OLSM) — Terminal capability: 23 dBm output power; RX diversity 1x2 — Bandwidth: 20 MHz The following additional assumptions have been made for the calculations: — Only one single VoIP user in the system (the user for which the link budget is calculated) — Background mobile broadband traffic service with noise rise in uplink and downlink in accordance with the example in Coverage and Capacity Dimensioning — A Remote Radio Unit and no feeder loss — 18 dB indoor penetration loss — 3 dB additional loss due to jumpers, body attenuation — 18.5 dB antennas — Channel model EPA at 5 Hz Doppler frequency — Frequency band: 2600 MHz — Urban area 4.2 Uplink Coverage In this section an example of uplink link budget is presented, with ROHC and with TTI bundling. 12/100 56-HSC 105 50/1-T1 Uen M | 2021-02-23 19 VoIP Dimensioning Table 10 Uplink Link Budget Example Link Budget UE output power PUE 23 Number of RBs, dimensioning service nRB 2 UE output power/RB PUE,RB 20 dBm/RB Thermal noise/RB NRB,UL -118.9 dBm/RB SINR at receiver -4.3 dB Gain for more than two Rx 0 dB RBS Sensitivity/RB SRBS,RB -123.2 dBm/RB Antenna gain Ga 18.5 dBi Jumper loss LJ 0 dB Body loss LBL 3 dB Car penetration loss LCPL 0 dB Building penetration loss LBPL 18 dB Fading margin BLNF 4.9 dB 135.8 dB Max path loss unloaded 4.3 dBm Interference margin, interfering service BIUL 2.6 dB Max path loss Lpmax 133.2 dB Range dmax 778 m Downlink Coverage In this section an example of downlink link budget is presented. The SINR value for PDCCH is used in a link budget for is used to find the maximum path loss value in the downlink, see Table 11. Table 11 Downlink Link Budget Example Link Budget Required SINR for PDCCH -6 dB Gain due to antenna arrangement 0 dB Thermal noise/RB NRB,DL -114.4 dBm/RB UE Sensitivity SUE -120.4 dBm RBS output power PRBS 40 W Number of RBs nRB 100 RBS output power/RB 20 0.4 W RBS output power/RB PRBS,RB 26.0 dBm/RB Interference margin, interfering service BIDL 8.6 dB Fading margin BLNF 4.9 dB Antenna gain Ga 18.5 dBi 12/100 56-HSC 105 50/1-T1 Uen M | 2021-02-23 Link Budget Example Link Budget Jumper loss LJ 0 dB Body loss LBL 3 dB Car penetration loss LCPL 0 dB Building penetration loss LBPL 18 dB Max path loss Lpmax 130.4 dB Range dmax 647 m The maximum path loss value according to Table 11 is lower than the maximum path loss for the uplink VoIP channel, Table 10. Hence the true cell range is determined from the new maximum path loss value for downlink. 12/100 56-HSC 105 50/1-T1 Uen M | 2021-02-23 21 VoIP Dimensioning 22 12/100 56-HSC 105 50/1-T1 Uen M | 2021-02-23 VoIP Capacity 5 VoIP Capacity Sections 5.1 to 5.4 treat capacity for VoIP only and Section 5.5 on page 28 treats capacity for mixed scenarios. 5.1 Introduction VoIP capacity in terms of number of users per cell depends on following factors: — Delay requirement: Higher allowed radio interface delay permits more users — Codec: Lower codec bit rates permits more users — Cell size: Smaller cell size permits more users — PDCCH allocation: Increased number of symbols allocated for the PDCCH gives better capacity. PDCCH allocation is dependent on Control Format Indicator (CFI) and system bandwidth; increasing the bandwidth will give a higher number of users — Enhanced PDCCH link adaptation: Gives improved capacity for the PDCCH — Header compression: ROHC increases the VoIP capacity generally but especially in cases where Physical Downlink Shared Channel (PDSCH) or Physical Uplink Shared Channel (PUSCH) capacity is the limiting factor — Unacknowledged mode: RLC UM increases the VoIP capacity in cases where the number of transmissions per TTI is limited — Number of Scheduling Elements (SEs) per TTI and cell: larger number gives more capacity — Delay-based scheduling: DBS and Grant Estimation reduce the PDCCH load and increase the VoIP capacity as well as the capacity for data users when VoIP users are present in the cell 5.2 Capacity for Different Codecs This section is based on information regarding FDD systems. Capacity simulations have been performed for different values of delay requirement for the radio link and codec. The simulations have been performed for two different environments, Case 1 with 500 m site-to-site distance and Case 3 with 1732 m site-to-site distance, see 3GPP TR 25.814: Physical layer aspect for evolved Universal Terrestrial Radio Access (UTRA). System bandwidth is 5 MHz and delay requirement is 80 ms. The results are shown in terms of users per cell in Table 12. 12/100 56-HSC 105 50/1-T1 Uen M | 2021-02-23 23 VoIP Dimensioning Table 12 Capacity Results for 5 MHz and Case 1 and 3 Environment Codec Capacity [users/cell] Case 1 AMR-WB 23.85 AMR-WB 12.65 AMR-WB 8.85 AMR-WB 6.6 AMR 12.2 AMR 7.95 AMR 5.9 AMR 4.75 EVRC 8.8 130 200 200 200 200 200 200 200 200 Case 3 AMR-WB 23.85 AMR-WB 12.65 AMR-WB 8.85 AMR-WB 6.6 AMR 12.2 AMR 7.95 AMR 5.9 AMR 4.75 EVRC 8.8 70 110 110 110 110 110 110 110 110 Note that the capacity analysis is based on generic simulation of the LTE system. Release-specific aspects have not been modeled. As specified by 3GPP, the requirement for VoIP capacity is at least 200 users for each cell in the 5 MHz bandwidth. For Case 1, a vast majority of the users are power unlimited and in this case reducing the bit rate provides no gain. These results apply for the pure VoIP case. It is especially challenging to reach high VoIP capacity in narrow bandwidths, 1.4 MHz and 3 MHz, because the relative control channel overhead is much larger than on the higher bandwidths. It is expected that the main issues on 1.4 MHz bandwidth will be the overhead caused by the Physical Uplink Control Channel (PUCCH) and Random Access Channel (RACH) as well as the limited amount of PDCCH resources available. Table 13 shows the simulation results for the following parameters: — Different codecs — 3GPP Case 1 and Case 3 24 12/100 56-HSC 105 50/1-T1 Uen M | 2021-02-23 VoIP Capacity Table 13 Capacity Result in Narrow Bandwidth for Case 1 and 3 with Different Codec Environment Codec Capacity 1.4 MHz [users/cell] Capacity 3 MHz [users/cell] Case 1 AMR-WB 23.85 AMR-WB 12.65 AMR-WB 8.85 AMR-WB 6.6 AMR 12.2 AMR 7.95 AMR 5.9 AMR 4.75 EVRC 8.8 22-25 35 35 35 35 35 35 35 35 45-50 70 70 70 70 70 70 70 70 Case 3 AMR-WB 23.85 AMR-WB 12.65 AMR-WB 8.85 AMR-WB 6.6 AMR 12.2 AMR 7.95 AMR 5.9 AMR 4.75 EVRC 8.8 8-10 20 20 20 20 20 20 20 20 15-20 40 40 40 40 40 40 40 40 Release-specific aspects have not been modeled in this case. 5.3 VoIP Capacity Limitations Basically, four things can limit the VoIP capacity in an LTE system: — UE power (in uplink) — Bandwidth — Number of SEs per TTI — PDCCH resources When the UE power is not limiting and 80 ms air interface delay is tolerated, PUSCH is the main limiting factor. With high cell isolation, such as in 3GPP Case 1 with tilt, the PUSCH limitation is very pronounced. When the cell isolation is lower, such as in a case without tilt, the PDSCH is close to being full. Figure 4 shows an overview of the reasons for capacity limitations for VoIP. The available number of SEs per TTI in a cell is hardware dependent. For example, in Baseband 5212 there are 16 available SEs per TTI and cell, and in Baseband 6502 there are 8 available SEs per TTI and cell in downlink. For latest SEs per TTI and cell for different hardware types, see Supported Capacity and Configurations. For narrow bandwidths, especially 1.4 MHz, the capacity is also limited by uplink control channels, see Section 5.2 on page 23. 12/100 56-HSC 105 50/1-T1 Uen M | 2021-02-23 25 VoIP Dimensioning Yes UE power Large cells Yes No PDCCH/Number of resource blocks Short delay requirement Yes No PUSCH High cell isolation No PUSCH PDSCH L0000417A Figure 4 5.4 VoIP Capacity Limitation for Larger Bandwidths PDCCH Allocation The PDCCH capacity is important since VoIP consists of many small packets. The packet intensity is composed of two parts, the activity part and the inactivity part: — For the activity part, the talk frames arrive in 20 ms interval — For the inactivity part, the SID frames arrive in 160 ms interval in steady state The same assumptions are used for both uplink and downlink. N If every packet would be transmitted individually and there is on average grants to be sent every subframe, the highest achievable capacity V oIP [users/cell] that would be obtained is given by Equation 5. N grants NV oIP = SE per TTINconsumption 32 Equation 5 Achievable Capacity The factor 2 appears because the number of grants required is the sum of the requirements for uplink and downlink. The SE per TTI consumption is for a power unlimited VoIP user given by Equation 6: V oIP packet rate 1 0 V AF Number of SEs per TTI consumption = (1 + pHARQ) V AF 1000 3 V oIPbundling + SID Equation 6 Number of SE per TTI Consumption for Power Unlimited UEs where pHARQ is the probability of HARQ retransmissions that is 10 [%] V AF is the VoIP Activity Factor that is 50 [%] 26 12/100 56-HSC 105 50/1-T1 Uen M | 2021-02-23 VoIP Capacity V oIP packetrate is the VoIP packet rate that is 50pps when talk frames arrive in 20 ms interval V oIP bundling indicates how many VoIP packets that can be sent together SID is the SID interval that is 160 [ms] The number 1000 in Equation 6 arises from the fact that there are 1000 subframes of 1 ms per second. When using DBS and Grant Estimation, VoIPbundling = 2, in other case, VoIPbundling = 1. When using DRX an equivalent expression to Equation 6 is given by Equation 7: 1 0 V AF V AF Number of SEs per TTI consumption = (1 + pHARQ) DRXcycle + SID Equation 7 Number of SE per TTI Consumption for Power Unlimited UEs when Using DRX where DRXcycle is the shortDrxCycle parameter shortDrxCycle is given in number of consecutive subframes. When using DBS and Grant Estimation, shortDrxCycle is set to SF40, which means 40 subframes. In other case, shortDrxCycle is set to SF20, which means 20 subframes. Ngrants is estimated from the number of available Control Channel Elements (CCEs), NCCE as shown in Equation 8: Ngrants = NNCCE CCE;ave Equation 8 Number of Used PDCCH where NCCE is the number of allocated CCEs in the cell NCCE;ave is the average number of CCEs per PDCCH NCCE depends on the bandwidth, antenna configuration and allocation of PDCCH symbols. See Table 14 for some examples: Table 14 Number of CCEs in a Subframe for One or Two Antenna Ports Number of CCEs Bandwidth [MHz] 1 PDCCH symbol 2 PDCCH symbols 3 PDCCH symbols 10 8 25 41 15 12 37 62 20 17 50 84 See Control Channel Dimensioning for more configurations. 12/100 56-HSC 105 50/1-T1 Uen M | 2021-02-23 27 VoIP Dimensioning N CCE;ave depends on cell size and load. For a cell in Case 1 environment, where a vast majority of the UEs are power unlimited, CCE;ave is close to 1. More information can be found in Control Channel Dimensioning. N For a cell in Case 1 environment, the limit would be 232 users for each cell when using DBS and Grant Estimation and 8 CCEs . In this case, it is assumed that all users are power unlimited, and therefore the capacity number will be over-estimated compared to simulations, where some users may be power limited, see Table 12. For larger cells, for example in Case 3 environment, at least for higher frequencies not all UEs are power unlimited. Hence they can either be in segmentation or TTI bundling, which means that the SEs per TTI cost is increasing. Also CCE;ave is increasing for large cells, which means that the capacity limit will decrease. N N There is no fixed number of PDCCH or grants in reality and reliable models of PDCCH are difficult to create. For rough estimations of the upper bound of the VoIP capacity, the figures above can be used. grants may also correspond to the hardware limitations on SEs. N 5.5 Capacity Consideration in Mixed Scenarios VoIP is co-existing together with other Packet Switched (PS) data in a network. VoIP is a GBR (Guaranteed BitRate) service, which means that VoIP users are prioritized against the PS users. If there are large numbers of VoIP users, other PS services may be starved out and capacity is not fully used. PS throughput will be zero when the system reaches the PDCCH limitation. This is addressed when QoS handling is introduced. With QoS handling, VoIP is a GBR service, which means that it has priority in the scheduler before other PS services like mobile broadband. VoIP is given QCI=1 marking it as GBR. With the admission control functionality, it can then be given higher priority in the scheduler than non-GBR traffic. When VoIP users are present in the system, the following factors are limiting the performance of the PS users: — Bandwidth — Number of SEs per TTI — PDCCH resources The introduction of DBS and Grant Estimation is expected to increase the capacity in mixed scenarios: — Increased throughput for PS users for the same amount of VoIP users in a cell — Increased number of VoIP users per cell for the same PS throughput 28 12/100 56-HSC 105 50/1-T1 Uen M | 2021-02-23 VoIP Capacity The following graph in Figure 5 shows an example of the increased uplink capacity when deploying DBS and Grant Estimation in a cell with mixed traffic. Throughput RR MBB throughput (kbps) Throughput DBS+GE Number of VoIP users L0000977B Figure 5 Uplink Data Throughput with RR and DBS+Grant Estimation for Different Number of VoIP Users per Cell DBS and Grant Estimation are expected to increase the capacity in terms of number of users per cell. According to system simulations, the increase is up to 36%. For few VoIP users, the PS throughput is limited by available bandwidth. For an increasing number of VoIP users, the PS throughput instead is limited by number of available SEs per TTI or PDCCHs. 12/100 56-HSC 105 50/1-T1 Uen M | 2021-02-23 29
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