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Voice Service On LTE PDF
Voice Service On LTE PDF
Prepared by:
Dr. Irina Cotanis, Anders Hedlund
Date:
October 2012
Document:
NT12-13122., Rev. 1.0
Ascom (2012)
All rights reserved. TEMS is a trademark of Ascom. All other trademarks are the property of their respective holders.
Contents
Ascom (2012)
2.1
2.2
2.3
3.1
3.1.1
3.1.2
3.1.3
3.1.4
3.2
4.1
4.2
4.2.1
4.2.2
4.3
4.3.1
4.3.2
5.1
5.1.1
5.1.2
5.2
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Conclusions ............................................................. 27
References................................................................ 28
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This paper discusses the different LTE voice service solutions as well as
aspects of the key performance evaluation metrics that must be considered
when implementing them. It takes an in-depth look into the challenges that
accompany the delivery of high quality of experience (QoE) LTE voice
services, as well as what is required to cope with these challenges. The
paper concludes with several examples of LTE voice service
troubleshooting that can help carriers efficiently provide voice service at
exigent QoE levels, consequently easing the all-IP migration for the voice
service that still accounts for more than 70% of their revenue.
For several years, the 3GPP and other wireless industry forums [4], [5], [6]
evaluated various voice service solutions that could optimally meet the
requirements imposed by the integration of voice within a data-oriented
network, such as LTE. Two 3GPP standardized solutions proved to be
feasible: Circuit Switch Fallback (CSFB) [7] and Voice over Internet
Protocol (VoIP) over IMS (or One Voice or Voice over LTE VoLTE) [8].
CSFB is seen as an interim and transitional solution until IMS technology is
fully deployed for wireless capabilities so it can then reliably offer complete
mobile voice support.
In addition, the deployment of packet switched (PS) voice (VoIP), and the
evolution of smartphones and broadband services, made it possible for
third-party Over the Top (OTT) voice solutions (e.g., Skype and Viber) to be
offered wirelessly over LTE, as well as 3G.
2.1
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Iu-ps
UTRAN
SGSN
Gs
Gb
Uu
GERAN
S3
Iu-cs
MSC
Server
Um
SGs
LTE-Uu
UE
S1-MME
E-UTRAN
MME
Figure 1
By design, the CSFB solution does not allow LTE functionality during voice
calls and generates interruption of an ongoing data connection. In addition,
it has weak support for multilayer networks (e.g., femto cells). The CSFB
solution does have minimal flexibility to be integrated with broadband voice
and multimedia services (e.g., presence, instant messaging, content
sharing) defined by the GSMA in the Rich Communication Suite Enhanced
(RCSe) [9] for LTE offerings.
As one would expect, because of the extensive signaling required to set up
the call, the CSFB solution comes with longer call setup times that could
significantly degrade the user experience. The call setup time also
increases with a change to another network. Estimated values show an
increase of 1.5 seconds in call setup time, regardless of call origination.
Some results of a live CSFB scenario are presented in section 5.2.
Therefore, the evaluation of the CSFB solutions performance requires
testing related to registration (e.g., MME translation of the Tracking Area
Indicator [TAI] of the LTE domain to the MSC Local Area Indicator [LAI] of
the 2G/3G domain) as well as the call setup; the latter potentially having a
significant negative impact on the QoE of the voice service. In addition,
evaluation of how much an incoming CSFB call impacts an ongoing user
data session is important in understanding the overall QoE.
2.2
The VoLTE solution initiated by the GSMA [8] is based on IMS technology
as defined by the 3GPP. The high-level architecture is presented in Figure
2. LTE radio access does not support direct connectivity to the circuitswitched core network and services, but rather radio is connected to an
Evolved Packet Core (EPC) that provides IP connectivity for the end user
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GERAN
S3
S1-MME
S6a
MME
PCRF
S11
S10
LTE-Uu
UE
S12
Serving
Gateway
E-UTRAN
S5
Rx
Gx
S4
PDN
Gateway
S1-U
SGi
Operator's IP
Services
(e.g. IMS, PSS)
Figure 2
VoLTE also defines a set of new interfaces (e.g., between the user's
equipment and the operators network, the Home and Visited Network
during roaming, and the networks of the two parties making a call).
On the network side, besides the new infrastructure and interfaces, VoLTE
standardization needs to address a series of functionalities required by the
integration within the LTE and the 2G/3G legacy networks. The subscribers
requirement for a seamless, anytime and anywhere call makes mobility and
handover to a non-LTE radio access technology (RAT, e.g., GSM, CDMA,
WCDMA) one of the most important functionalities. This is achieved by
using the Single Radio Voice Call Continuity (SRVCC) [10] function. Other
functionalities address optimal routing of bearers for voice calls when
customers are roaming, commercial frameworks and provisioning
capabilities for roaming and interconnect, as well as security and fraud
threat audit to prevent hacking and unauthorized entry into any area within
the network.
On the terminal side, the phone needs to have VoIP client software loaded
to provide the VoLTE functionality, which can be implemented at the
application layer of the phones protocol stack, in the form of an app. In this
case, the clients features such as time scaling of the voice signal which
regards the jitter-buffer handling can be controlled and tuned for improved
voice service quality by the phone vendors. The VoLTE functionality also
can be embedded in the phones chipset, in which case the modem-based
clients features are set by the chipset vendor (allowing less flexibility for
control and tuning). Details on the importance of the time-scaling feature on
the QoE of the voice service are presented in section 3.1.3.
Adopting the IMS-based specifications allows the VoLTE solution to be
integrated with the suite of applications that will become available on LTE
through the IMS core. A variety of services can run seamlessly, rather than
having several disparate applications operating concurrently. The GSMA
defined the multimedia communication suite (RCS) to run both over LTE
and other networks such as 2G/3G. It covers multimedia services in three
areas: rich address book, rich messaging, and rich call. As part of the rich
call, RCS includes the voice service, regardless of whether it is realized on
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2.3
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(GBR) bearer with something like a. Quality Class Indicator (QCI) =7 [11]. It
will more likely be a dynamic scheduler scheme that ensures optimal radio
resources for each transmission depending on the radio conditions and
load instead of an optimized VoIP scheduler (e.g., semi-persistent) and
dedicated bearer (e.g. QCI=1 [11]). Third-party voice service providers have
no control over these QoS aspects in the wireless network, and thus they
cannot ensure a good QoE under all load situations. This issue could
possibly be resolved by installing logic in the network that would ensure
QoS for data streams that are recognized as belonging to an external voice
service to which the user is subscribed. However, standardized work is
needed on this topic.
Second, third-party OTT calls cannot be handed over to a circuit-switched
2G/3G network when a user leaves the LTE coverage area since the
external applications cannot easily be tied into the wireless network
infrastructure.
Therefore, testing OTT solution performance requires a careful analysis of
the aforementioned QoS aspects that could possibly generate poor or even
unacceptable QoE. In addition, the understanding of the OTT voice QoE
requires evaluation of the behavior and performance of OTT clients that
embed proprietary error concealment schemes and adaptive buffering
techniques on the subscribers device.
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3.1
3.1.1
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exhibit a larger range of delays and/or attenuations than the ones the echo
canceller was designed to intervene with and compensate for.
3.1.3
Commonly used phones impact the voice quality due to time-variant linear
distortions, such as spectral shaping, and/ or non-linear distortions like
microphone and transducer interfaces and reverberations caused by
hands-free set-ups at acoustical interfaces. Todays smartphones designed
for HD voice, and with technologically advanced acoustical interfaces, are
expected to have less impact on voice quality. However, LTE requirements
for high bandwidth efficiency (to support a multitude of data and multimedia
services while coexisting with voice delivered on PS IMS support) drove
the necessity of adaptive buffering schemes. These buffering schemes can
use various time-scaling or speech-frequency re-sampling algorithms to
cope with challenging network behaviours affected by packet loss and
delays such as inter-RAT handovers and IP congestions. Time scaling can
be either stretching (under sampling) or compressing (over sampling) the
speech signal, depending on the rate with which it comes from the buffer
and/or if the buffer is over-run or empty [19].
There are two main categories of time-scaling algorithms: with speechpitch preservation or without ([14]); each exhibiting different trade-offs
between performance and speech processing complexity. The impact of
the algorithms performance on the overall speech quality is determined by
the distribution of the time scaling and its frequency of occurrence within
the speech sample, as well as its length. All these characteristics are given
by the network behavior (e.g., packet loss, variable delay), which requires
different levels and distributions of algorithmic error compensation.
The time-scaling algorithms are not standardized, leaving open the
possibility of various performances and, therefore, of different QoE trends.
The algorithms are implemented in the VoIP client that supports the voice
service in the network. The clients can be either software clients as
applications on the device, or modem-based implemented in the phones
chip. Therefore, the clients performance of coping with network behaviors
is client- and chipset vendor-dependent and phone-dependent.
3.1.4
Network-Centric Factors
These factors affecting the LTE voice service quality emerge from various
networks radio frequency (RF) and non-RF characteristics described in
more detail in section 4. Those that have immediate impact generate
interruptions or loss and delay. Not cancelled packet loss could generate
perceived interruptions, whether caused by reasons such as non-ACK RLC
mode of the VoIP dedicated bearer or IP congestion, and uncompensated
handover (inter-RAT). Delay especially when it is variable length and
randomly distributed during speech, rather than at the beginning of speech
is very annoying to subscribers. These scenarios could be caused by
uncompensated handover delays (LTE intra-RAT) or uncompensated RLC
retransmissions in scenarios for which the VoIP bearer is deployed using
ACK RLC mode. VoLTE might also show the delays and interruptions due
to the IMS technology solutions that cope with mobility and unreliable radio
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3.2
LTEs signature on voice service quality also comes with the human
perception dimension; that is, subscribers comparing the experience of the
VoIP quality and additionally VoLTE (VoIP over IMS) quality against the
circuit switched (CS) voice quality they have experienced for the past 20
years.
PSTN/CS voice service with a dedicated 56k (or 64k) time slice allocated
for each channel/circuit is governed by highly optimized 2G/2.5G/3G
networks. In addition, CS voice service benefits from well established and
standardized codecs with highly efficient error concealment and rate
adaptation techniques encoding both NB and WB, as well as enhanced
speech processing procedures. All these factors contribute to raise voice
service quality to levels that are known to satisfy subscribers, and this is
categorized as providing a mean opinion score (MOS) of 4.2MOS to
4.4MOS for the entire calls length.
VoIP service is supported by packet switching that was not originally
designed for real-time sessions, such as voice and video traffic, and/or
mobility. VoIP requires the call to go through various transformations, such
as encoding/decoding at low and adaptive bit rates, changes in routing
during the call, packets out of sequence or lost, delays, and buffering/jitter
delays. To compensate for all of these challenges while sustaining the
customer experience, new protocols such as Real-time Transport Protocol
(RTP), Real-Time Transport Control Protocol (RTCP), and Session
Initiation Protocol (SIP), as well as new QoS strategies and policies such as
Multiprotocol Label Switching (MPLS) have been developed. In addition,
new codecs, with a large variety of rates and even variable rates, and
multiple bandwidths, from narrowband to super wideband with complex
error concealment techniques, create the foundation for a high quality voice
service, if provided at pre-established service level agreements (SLAs).
Therefore, given a dedicated bandwidth, minimum delay, HD voice and
efficient QoS policies such as MPLS, it is expected that the quality of VoIP
service will soon meet, and actually surpass, the voice service QoE that
subscribers are already used to and expect.
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4.1
4.2
Network-Centric View
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Figure 3
Figure 4
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The highest layer in the RAN part of the stack, the RLC layer, can be set up
in two different modes for voice; acknowledged (ACK) and unacknowledged (U-ACK). The third existing mode is transparent and it is
used for signaling broadcast-like system information.
The RLC acknowledge mode ensures an error-free radio interface since the
erroneous blocks from the Medium Access Control layer are retransmitted
by the RLC layer. However, the price paid for an error-free air interface can
result in delays caused by the retransmissions.
In LTE, the MAC layer handles retransmissions with a very short delay
(within 10ms in most cases). In this case, it is better to leave the eventual
remaining block errors (residual errors from the MAC layer) to be handled
by the error concealment mechanism in the UE-based VoIP client, rather
than introducing additional delays in the RLC layer.
The delay values larger than the jitter-buffer in the voice client will leave the
client with no speech to decode resulting in degraded speech quality. The
jitter-buffer consists of 20ms speech frames delivered from the RTP layer.
The jitter-buffer is typically around 80 to 100ms and could be dynamically
adjusted to the level of variations measured on the received packets from
the RTP layer.
Therefore, for a real-time, delay-sensitive service such as VoLTE, the RLC
layer should be set up in unacknowledged mode. A sequence numbering
mechanism is generally used to ease the RLC packet handling and ensure
an efficient, more reliable unacknowledged mode.
4.2.1.2
Scheduling
A VoLTE service can be scheduled with higher priority than other non-realtime services and prioritization between different services is possible if the
scheduler knows which type of services each user runs. This is referred to
as a QoS-aware scheduler, which is a feature of the scheduler, rather
than a configuration of the MAC protocol itself.
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4.2.1.4
Semi-Persistent Scheduling
The MAC layer allows for the possibility of persistent scheduling. However,
the VoLTE service follows a known pattern (typically 20ms blocks with a
limited size) and each radio block does not have to be assigned uniquely.
Therefore, it is possible to reserve and dedicate a part of the resources to a
particular end-user service. Called a Semi-Persistent Scheduling (SPS)
configuration, it reduces the signaling overhead significantly and thereby
the load on the Physical Dedicated Control Channel (PDCCH).
4.2.1.5
DRX Configuration
MAC Retransmissions
Whenever RLC runs in U-ACK mode, it is likely that more than the network
default setting of four retransmissions is needed. The LTE MAC can
configure the Hybrid Automatic Repeat reQuest (HARQ) retransmission
handling mechanism so that it performs better for voice services.
4.2.2
4.2.2.1
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Table 1
4.2.2.2
Frequency Hopping
Utilizing frequency diversity by using optimal parts of the spectrum over the
time domain is possible in LTE. In the downlink, the scheduler can use the
sub-band CQI feedback from the UE. For the uplink, methods for both interTTI and intra-TTI hopping exists, and the sounding signals from the UE can
be used to detect optimal parts of the spectrum for specific UEs.
This also helps in optimizing the VoLTE service.
4.2.2.3
The protocols above the radio layer are controlled mainly by the EPS
Session Management and the EPS Evolved Packet System, which sets up
a bearer context describing the characteristics of the service quality
requirements. The requirements of VoLTE service on delay, bandwidth,
and priority make the use of the QCI configuration 1 suitable for the
appropriate performance to be provided [7].
The Packet Data Convergence Protocol includes functions such as
encryption, header compression, and sequence numbering. The header
compression is very important in order to keep the protocol overhead at a
minimum level. The Robust Header Compression (RoHC) profile (e.g.,
0x0001 or 0x0101) is recommended for VoLTE services [15].
4.3
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4.3.1
On-Device Testing
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5.1
VoLTE Troubleshooting
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ESM configuration
QCI (PDN Connection)
RoHC (PDCP protocol)
RAN
Configuration
Cell bandwidth
UE category
TDD UL/DL Config
MTU Size
Protocol stack configuration
POLQA
Dynamic parameters
MOS
Speech
related
Client
related
RF Measurements
RSRP, RSSI
CINR
CQI
RI
PMI
Analyze
centric reasons
Network
relatedU
pper
Network centric
reasons
Client Information
Codec
Buffer overrun, underrun
Time scaling
Packet loss
Echo
Volume
Figure 5
5.1.1
5.1.1.1
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Figure 6
5.1.1.2
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Figure 7
5.1.1.3
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Figure 8
5.1.1.4
Cell Load
Cell load can have a significant impact on the performance of the VoLTE
service. Three types of measurements should be monitored: RSRQ
(Reference Symbol Received Quality), CFI (Channel Format Indicator) and
resource block scheduling rate. RSRQ is a measure of the relationship
between signal strength (RSRP) measured on the reference symbols and
the total received signal strength received on all symbols (RSSI). It gives an
indication of the load in the cell, but can be difficult to use since small
variations in RSRQ can be caused by large differences in load. The CFI is
the control format indication and provides the number of symbols used for
PDCCH (Physical Dedicated Control Channel), which is the control channel
for the downlink. The number of active users in the cell can be monitored
by checking the distribution of the CFI values (0, 1, 2) (Figure 9);
distribution of the CFI toward higher values indicates more users. Last, the
scheduling can be checked via the PDSCH (Physical Downlink Shared
Channel) and PUSCH (Physical Uplink Shared Channel) resource blocks
allocation. These values show how many of the total available resource
blocks are assigned to the UE. Even though the typical usage of those
measurements is to troubleshoot high-bandwidth services, they can also
provide an indication of how many users there are in the cell.
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Figure 9
5.1.2
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QoE value on the downlink of one of the devices can be the result of a
handover on the uplink of the other device involved in the call.
A demo screenshot of some of these measurements, collected in drive
testing, is presented in Figure 10.
Figure 10
5.2
As already mentioned in section 2.1, call setup time as well as the impact
on data sessions running in parallel are significant for Circuit Switched
Fallback (CSFB) QoE troubleshooting.
Drive test results on these measurements are discussed in this section
(Figures 11, 12, 13). It can be seen that the call setup time shows an
average of about 6 seconds, with deviations of about +/- 2 seconds, which
can produce an annoying experience for the customer. In addition, the data
sessions have been interrupted for about 2.5 seconds on average with
minimums of 1 second and maximums reaching 3.5 seconds. However,
these values are not likely to annoy users since data sessions are
performed (and perceived) as background activities. The time needed to
move back to LTE once the CSFB call ends, which consistently shows
values larger than 25 seconds, can be far more troubling. Although this
doesnt have a direct impact on customer experience, it shows the
deficiency of continuous use of increased LTE data capacity.
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11 13 15 17 19 21
Call Attempt
Figure 11
Data Session
Interruption Time (s)
Figure 12
Figure 13
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Conclusions
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7
[1].
[2]
3GPP One Voice; Voice over IMS profile V1.0.0, Nov. 2009
[3]
[4]
[5]
[6]
[7]
GSMA Forum
3GPP TS 23.272 Circuit Switched (CS) fallback in Evolved Packet
System (EPS); Stage 2
[8]
[9].
GSMA RCS-e
[10]
[11]
[12]
[13]
[14]
[15]
[16]
[17]
[18]
[19].
[20].
[21].
[22].
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References
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