VoIP Dimensioning
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Copyright
© Ericsson AB 2011-21. All rights reserved. No part of this document may be
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continued progress in methodology, design and manufacturing. Ericsson shall
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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
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VoIP Dimensioning
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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)
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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
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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
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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
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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
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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.
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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
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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.
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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
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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.
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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%
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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
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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
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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
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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
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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.
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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.
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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
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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.
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VoIP Dimensioning
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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.
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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
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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.
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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 [%]
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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.
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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
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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.
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