Network Architecture's Impacts on

OTT Service Experience and Air

Interface bandwidth Requirements
（2014H1）
White Paper on Network Architecture's Impacts on OTT Service
Experience and Air Interface Rate Requirements（2014H1） Contents

Contents

1 Introduction.................................................................................................................................... 1
2 Context ............................................................................................................................................ 2
3 Relationships Between User Experience and TCP Throughput Rates ............................... 3
3.1 Analysis of Transmission Mechanisms of Mainstream OTT Services .......................................................................... 3
3.2 Relationships Between TCP Throughput Rates and RTT ............................................................................................. 4
3.2.1 Relationships Between TCP Throughput Rates and RTT .......................................................................................... 4
3.2.2 User Experience Formula for Services Based on TCP .............................................................................................. 5
3.2.3 TCP Characteristics Analysis for Web Services ......................................................................................................... 6
3.2.4 TCP Characteristics Analysis for Video Services ...................................................................................................... 8

4 Comparison of User Experience in Typical Network Scenarios ........................................ 11

4.1 E2E RTT Analysis in Typical Network Scenarios ...................................................................................................... 11
4.2 Comparison of User Experience Between UMTS and LTE........................................................................................ 12
4.2.1 Comparison of Web Service Experience Between UMTS and LTE ........................................................................ 12
4.2.2 Comparison of Video Service Experience Between UMTS and LTE ...................................................................... 13
4.3 Comparison of User Experience Between Gi Cache and Non-Gi Cache Scenarios ................................................... 14
4.3.1 Comparison of Web Service Experience Between Gi Cache and Non-Gi Cache Scenarios for LTE ...................... 14
4.3.2 Comparison of Web Service Experience Between Gi Cache and Non-Gi Cache Scenarios for UMTS .................. 14
4.3.3 Comparison of Video Service Experience Between Gi Cache and Non-Gi Cache Scenarios for LTE .................... 15
4.3.4 Comparison of Video Service Experience Between Gi Cache and Non-Gi Cache Scenarios for UMTS ................ 15
4.4 Comparison Between National and International User Experience ............................................................................ 16
4.4.1 Comparison of User Experience in Accessing Top30 National and International Web Pages ................................. 16
4.4.2 Comparison Between National and International Video Service Experience .......................................................... 16
4.5 Comparison of User Experience Between Eastern and Western China ...................................................................... 17
4.5.1 Comparison of Web Service Experience Between Eastern and Western China ....................................................... 17
4.5.2 Comparison of Video Service Experience Between Eastern and Western China..................................................... 18
4.6 E2E RTT and User Experience Summary in Typical Network Scenarios .................................................................. 18

5 Summary ....................................................................................................................................... 20
5.1 Service Inflow Determining Air Interface Requirements ........................................................................................... 20
5.2 Air Interface Bandwidth and RTT Together Determine OTT Service Rates on the Wireless Network ...................... 21
5.3 Factors Affecting E2E RTT ........................................................................................................................................ 21
5.4 Air Interface Optimization and Architecture Optimization Together Achieve Optimal User Experience .................. 22
White Paper on Network Architecture's Impacts on OTT Service
Experience and Air Interface Rate Requirements（2014H1） Contents

5.5 Routine Work Introduction for mLAB User Experience Study .................................................................................. 22
5.5.1 Objective .................................................................................................................................................................. 22
5.5.2 Working Mode ......................................................................................................................................................... 22
5.5.3 Contact ..................................................................................................................................................................... 23
5.5.4 Disclaimer ................................................................................................................................................................ 23

A Terms ............................................................................................................................................ 24
B References .................................................................................................................................... 26
White Paper on Network Architecture's Impacts on OTT Service
Experience and Air Interface Rate Requirements（2014H1） Figures

Figures

Figure 2-1 Comparison between LTE and UMTS when the air interface bandwidth is not restricted .................. 2

Figure 3-1 Transmission mechanisms of mainstream OTT services ..................................................................... 3

Figure 3-2 Mapping between the TCP throughput rate and RTT in the slow start phase ...................................... 4
Figure 3-3 Relationships between TCP throughput rates and E2E RTT in the TCP slow start phase ................... 5

Figure 3-4 TCP service experience formula .......................................................................................................... 5

Figure 3-5 Statistics on the number and sizes of embedded objects on web pages ............................................... 6

Figure 3-6 Statistics on TCP characteristics of web services ................................................................................ 7

Figure 3-7 Relationship Between the Page Display Latency and Number of Embedded Objects (LTE) .............. 8

Figure 3-8 Statistics on TCP characteristics of video services .............................................................................. 8

Figure 3-9 Statistics on TCP characteristics of video services ............................................................................ 10

Figure 4-1 E2E RTT analysis in typical network scenarios ................................................................................. 11

Figure 4-2 Comparison of web service experience between UMTS and LTE .................................................... 12

Figure 4-3 Comparison of video service experience between UMTS and LTE .................................................. 13

Figure 4-4 Comparison of web service experience between Gi cache and non-Gi cache scenarios for LTE ...... 14

Figure 4-5 Comparison of web service experience between Gi cache and non-Gi cache scenarios for UMTS .. 14
Figure 4-6 Comparison of video service experience between Gi cache and non-Gi cache scenarios for LTE .... 15

Figure 4-7 Comparison of video service experience between Gi cache and non-Gi cache scenarios for UMTS 15

Figure 4-8 Comparison between national and international web service experience .......................................... 16

Figure 4-9 Comparison between national and international video service experience ........................................ 16

Figure 4-10 Comparison of web service experience between eastern and western China .................................. 17

Figure 4-11 Comparison of video service experience between eastern and western China ................................ 18
Figure 5-1 Relationship between air interface requirements, and RTT and number of concurrent TCP
connections........................................................................................................................................................... 20
Figure 5-2 Relationship between air interface requirements and RTT ................................................................ 21

Figure 5-3 Factors affecting E2E RTT ................................................................................................................ 21

Figure 5-4 Air interface optimization and architecture optimization together improving user experience ......... 22
White Paper on Network Architecture's Impacts on OTT Service
Experience and Air Interface Rate Requirements（2014H1） Tables

Tables

Table 3-1 Video initial buffer and segment statistics ............................................................................................. 9

Table 4-1 E2E RTT analysis in typical network scenarios................................................................................... 12

Table 4-2 E2E RTT and user experience in typical network scenarios ................................................................ 18
White Paper on Network Architecture's Impacts on OTT Service
Experience and Air Interface Rate Requirements（2014H1） 1 Introduction

1 Introduction

Based on the study of OTT service experience and network requirements, mLAB finds that
user experience for LTE is better than that for UMTS when the air interface bandwidth is not
restricted (for example, MBR is set to the maximum value for a tested subscriber). To study
the preceding phenomenon as well as the best experience supported by radio network
capabilities, this document describes network architecture's impacts on user experience based
on the study of the transmission mechanisms of mainstream OTT services (E2E RTT is used
to indicate network architecture differences in this document).
Key findings of the study are as follows:
 When the air interface bandwidth is not restricted,
 Available user experience and service rates are determined by the size of service contents
and the service TCP throughput rate.
 Smaller RTT indicates that the TCP throughput rate will reach or nearly reach the
maximum available air interface bandwidth within a shorter period of time.
 Smaller RTT indicates a higher service TCP throughput rate.
 Network architecture is the most important factor that affects E2E RTT. In the same
scenarios, different network architectures cause different RTT and throughput rates at the
TCP layer. To achieve optimal user experience, networks need to be planned based on air
interface bandwidth requirements and network architectures need to be improved
continuously.
White Paper on Network Architecture's Impacts on OTT Service
Experience and Air Interface Rate Requirements（2014H1） 2 Context

2 Context

UMTS and LTE networks connected to the same core network (CN) with the same Gi
interface are used to test and compare user experience. Figure 2-1 shows the test and
comparison when the same top web pages are accessed, and the UMTS and LTE air interface
bandwidth is not restricted.

Figure 2-1 Comparison between LTE and UMTS when the air interface bandwidth is not
restricted

As shown in Figure 2-1, the percentage of page display latency which is smaller than 5s for
LTE is 20% higher than that for UMTS and the average TCP throughput rate for web services
on the LTE network is 10% higher than that on the UMTS network, that is, the LTE service
inflow is higher than the UMTS service inflow.
Air interfaces of the tested UMTS and LTE networks do not bear any other service and air
interface bandwidths are not restricted, and therefore the difference between UMTS and LTE
is that RTT of UMTS RAN is different from RTT for LTE RAN. Currently, mLAB has
studied and released air interface bandwidths' impacts on user experience. For details, see
OTT Service KQI and Mobile Network Requirements Reference V2.5 2013H2 released by
mLAB (this document is updated annually). To study the best user experience supported by
radio network capabilities, the author of this document focuses on analysis of the TCP
throughput rate which may become an experience bottleneck and network architecture's
impacts on the TCP throughput rate (E2E RTT is used in this document to analyze
architecture differences).
White Paper on Network Architecture's Impacts on OTT Service 3 Relationships Between User
Experience and Air Interface Rate Requirements（2014H1） Experience and TCP Throughput Rates

3 Relationships Between User Experience

and TCP Throughput Rates

3.1 Analysis of Transmission Mechanisms of Mainstream

OTT Services
Figure 3-1 Transmission mechanisms of mainstream OTT services

 Service throughput rates are determined by the number of concurrent TC connections

and the single TCP throughput rate. By default, 10 concurrent TCP connections can be
set up for web services of one host (there may be multiple hosts for a web page), whereas
a TCP connection is set up for each segment of videos for most OTTs. Segments are
downloaded one by one. (Buffering mechanisms are used for videos, and therefore
segment download is discontinuous.)
 The single TCP throughput rate is determined by RTT and the send window (initial size
of the send window and maximum size of the send window in the stable phase).
White Paper on Network Architecture's Impacts on OTT Service 3 Relationships Between User
Experience and Air Interface Rate Requirements（2014H1） Experience and TCP Throughput Rates

 When link bandwidths (such as air interface bandwidth) are not restricted, total TCP
throughput rates are determined by the single TCP throughput rate and number of
concurrent TCP connections.

3.2 Relationships Between TCP Throughput Rates and

RTT
3.2.1 Relationships Between TCP Throughput Rates and RTT
1. TCP throughput rate in the slow start phase
− Average TCP throughput rate within the first N RTT in the slow start phase (packet
loss is not considered):
TCP throughput rate = (2N – 1) x MSS/((N + 1.5) x RTT)
− Average TCP throughput rate within the first N RTT in the slow start phase (packet
loss is not considered):
TCP throughput rate = (2N-1) x MSS/RTT
− The send window size for RTT N is calculated by multiplying 2N-1 by MSS.
− If PMMS is used to indicate the payload of a TCP packet, the payload of transmitted
data blocks within RTT N is calculated as follows: TCP-PayLoad = (2N– 1) x PMSS

Figure 3-2 Mapping between the TCP throughput rate and RTT in the slow start phase
MSS ＝ 1500Bytes
Average Throughput Rate Throughput Rate at Nth Data
RTT(ms) Total Data
（kbps） RTT（kbps） Transmitted
N WIN Transmitted
at Nth RTT
RTT=50 RTT=100 RTT=200 RTT=50 RTT=100 RTT=200 RTT=50 RTT=100 RTT=200 （KB）
(KB)
0 1 0 0 0 240 120 60 75 150 300 0 1
1 2 80 40 20 480 240 120 125 250 500 1 3
2 4 180 90 45 960 480 240 175 350 700 4 6
3 8 336 168 84 1920 960 480 225 450 900 10 11
4 16 600 300 150 3840 1920 960 275 550 1100 21 23
5 32 1063 531 266 7680 3840 1920 325 650 1300 44 46
6 64 1890 945 473 15360 7680 3840 375 750 1500 90 91
7 128 3387 1693 847 30720 15360 7680 425 850 1700 181 183
8 256 6120 3060 1530 61440 30720 15360 475 950 1900 364 365
9 512 11149 5575 2787 122880 61440 30720 525 1050 2100 729 730
10 1024 20460 10230 5115 245760 122880 61440 575 1150 2300 1459 1460
11 2048 37791 18895 9448 491520 245760 122880 625 1250 2500 2919 2920
12 4096 70200 35100 17550 983040 491520 245760 675 1350 2700 5839 5840
13 8192 131056 65528 32764 1966080 983040 491520 725 1450 2900 11679 11680
14 16384 245745 122873 61436 3932160 1966080 983040 775 1550 3100 23359 23360
15 32768 462593 231296 115648 7864320 3932160 1966080 825 1650 3300 46719 46720
16 65535 873800 436900 218450 15728400 7864200 3932100 875 1750 3500 93439 93439

For easy description, the window size in this document is indicated by the number of packets, In TCP,
the window size needs to be multiplied by MSS. MSS is equal to 1500 bytes in the preceding table.
When the OTT TCP transmission mechanism is fixed (initial send window, slow start
mechanism, and server load control), smaller RTT indicates higher requirements of
services on the air interface bandwidth.
− Figure 3-2 shows that the objects whose sizes are smaller than 90 KB can be
transmitted within a maximum of 5.5 RTT (excluding 1.5 RTT for TCP connection
setup).
White Paper on Network Architecture's Impacts on OTT Service 3 Relationships Between User
Experience and Air Interface Rate Requirements（2014H1） Experience and TCP Throughput Rates

− When the size of the send window is 64 (MSS = 1500 bytes) and the value of RTT is
50 ms, the TCP throughput rate is 15.36 Mbit/s. When the value of RTT is 100 ms,
the TCP throughput rate is 7.68 Mbit/s. In this case, the requirements on the air
interface bandwidth are decreased by 50%.
− Therefore, smaller RTT indicates that the TCP throughput rate will reach or nearly
reach the minimum link bandwidth of whole IP Path (such as the air interface
bandwidth) more quickly.

Figure 3-3 Relationships between TCP throughput rates and E2E RTT in the TCP slow start phase
TCP throughput rate

Congestion control
E2 Available air
E
RT Packet loss interface bandwidth
T timeout
re
du
cti
on
Slow start Slow start

Initial slow start rate

Traffic gains caused
by RTT reduction

Time

2. TCP throughput rate in the congestion avoidance phase (stable phase)

TCP throughput rate = MIN (Wm/RTT, Link-Bandwidth), Wm = MIN (CWIN, RWIN)
− Wm: send window
− CWIN: congestion window. Determined by link packet loss rate, delay/jitter, server
sending performance, and restrictions on the single user available bandwidth. After
slow TCP start, the size of the CWIN will become the same as that of the RWIN for
continuous data transmission when the packet loss rate is 0 and there is no timeout.
− RWIN: Receive window. Currently, the size of the window is large. The maximum
size can reach 1 GB through scaling factors. Therefore, RWIN is not a bottleneck in
most cases.
− Link-Bandwidth indicates the minimum bandwidth of nodes or links between a
terminal and server. For wireless networks, the air interface bandwidth is considered
as the main bottleneck currently.

3.2.2 User Experience Formula for Services Based on TCP

When packet loss is not considered, TCP throughput rates are determined by the TCP send
window and E2E RTT. In the slow start phase, smaller RTT indicates quicker TCP send
window growth and larger volume of data that can be transmitted. In the congestion control
phase, the TCP window size is determined by E2E link bandwidths. Higher link bandwidths
indicate larger TCP send window sizes. The sum of TCP throughput rates for the services
accessed by a single user will finally reach or nearly reach the available air interface
bandwidth that is most likely to become a bottleneck on E2E links.

Figure 3-4 TCP service experience formula

White Paper on Network Architecture's Impacts on OTT Service 3 Relationships Between User
Experience and Air Interface Rate Requirements（2014H1） Experience and TCP Throughput Rates

Service Content Size

Experience 
MINNumber of TCPs (Send Window Size/E2E RTT) , Available Air Interface Bandwidth 

When the air interface bandwidth is not a bottleneck, total TCP throughput rates are related to
the TCP send window, RTT, and number of concurrent TCP connections. Reducing service
E2E RTT is a key measure to improve the service rate.
 When the available air interface bandwidth is not a bottleneck, service TCP throughput
rates are determined by RTT. Smaller RTT indicates better user experience.
 When RTT is reduced to a certain degree, the service throughput rate exceeds the
available air interface bandwidth. In this case, the available air interface bandwidth will
become a user experience bottleneck.

3.2.3 TCP Characteristics Analysis for Web Services

Figure 3-5 Statistics on the number and sizes of embedded objects on web pages

 The Figure 3-5 shows that the sizes of 80% of embedded objects on web pages are
smaller than 20 KB. The objects can be downloaded within 3.5 RTT in most cases
(excluding 1.5 RTT for TCP connection setup).
 According to mLAB's top page statistics, the number of embedded objects on 90% of
mobile phone web pages is smaller than 50. The number of embedded objects on PC web
pages is smaller than 150 in most cases.
 Web page loading latency includes main page loading latency, main page parsing latency
(which depends on client performance and is excluded in subsequent calculations in this
document), loading latency for embedded objects on web pages, and page rendering
latency (which is excluded in subsequent calculations in this document). Main page
loading latency and loading latency for embedded objects on web pages are calculated as
follows:
− Main page loading latency includes main page DNS request latency (which can be
calculated based on 1 RTT), main page TCP connection setup latency (which can be
calculated based on 1.5 RTT), and main page loading latency.
− Object loading latency can be calculated as follows: Object loading latency = (Object
DNS request latency (which is excluded if the objects and the main file belong to the
same host) + TCP connection setup latency + Accumulated object loading latency) x
(Total number of objects/Number of loading objects in a single TCP/Number of
concurrent TCP connections).
 For example, when the number of embedded objects is 50, and the sizes of both the main
file and embedded objects are 20 KB (the total size is 1 MB), the page loading latency is
calculated as follows:
White Paper on Network Architecture's Impacts on OTT Service 3 Relationships Between User
Experience and Air Interface Rate Requirements（2014H1） Experience and TCP Throughput Rates

− If the initial window size for slow start is 1, loading both the main file (20 KB) and a
single embedded object takes 3.5 RTT, as shown in Figure 3-2. Assuming that the
TCP concurrency rate is 10 and two objects are transmitted in each TCP, the time for
consecutively loading two objects is the sum of 3.5 RTT (loading latency of the first
object), 1 RTT (Get request), and 1 RTT (loading latency of the second object), that is,
5.5 RTT. The loading time of the first page is calculated as follows: Loading time of
the first page = Main file loading latency [1 RTT (DNS request) + 1.5 RTT (TCP
connection setup) + 3.5 RTT] + Embedded object loading latency [1.5 RTT (TCP
connection setup) + 5.5 RTT + 1 RTT (interval between two TCP connections)] x
(50/2/10) ≈ 30 RTT.
− If the initial window size for slow start is 2, both main page loading latency and first
TCP object loading latency decrease by 1 RTT. The page loading latency is calculated
as follows: Page loading latency = Main file loading latency (1 RTT + 1.5 RTT + 2.5
RTT) + Embedded object loading latency (1.5 RTT + 4.5 RTT + 1 RTT) x (50/2/10) ≈
26 RTT.
− Therefore, the loading latency of the web pages in the example is between 26 RTT
and 30 RTT. Page parsing latency and page rendering latency also needs to be
considered in actual service experience.

Figure 3-6 Statistics on TCP characteristics of web services

 According to distribution of single TCP transmission objects for web services shown in
Figure 3-6, only one object is transmitted for 66% of TCP connections, two to four
objects are transmitted for 28% of TCP connections, and more than five objects are
transmitted for about 6% of TCP connections. The average number of objects is 1.78.
 According to sizes of initial TCP send windows for web services shown in Figure 3-6,
the size of the initial TCP send window for LTE web services is consistent with that for
UMTS web services (smaller than 10 which is the number of initial continuous downlink
packets in most cases), the size of 25% of initial send windows is 1, that of 75% of initial
send windows is 2 to 10. The initial send windows whose sizes are 10 account for about
8.6%. The average size is about 3.4.
 According to sizes of maximum TCP send windows for web services shown in Figure
3-6, the sizes of the maximum TCP send windows for LTE web services are consistent
with those for UMTS web services. The sizes of 87% of maximum TCP send windows
are smaller than 10. (This is because 80% of objects are smaller than 20 KB and have
been transmitted when the size of the window reaches 8.)
 According to statistical analysis, the number of concurrent TCP connections for one host
is 10 in most cases. If the objects embedded on web pages are distributed on multiple
hosts, the number of concurrent TCP connections may exceed 10.
White Paper on Network Architecture's Impacts on OTT Service 3 Relationships Between User
Experience and Air Interface Rate Requirements（2014H1） Experience and TCP Throughput Rates

Figure 3-7 Relationship Between the Page Display Latency and Number of Embedded Objects
(LTE)

The 5000 ms point in the chart is used as an example. The blue dot indicates that the page display
latency ranging from 4500 to 5000 accounts for 3.9%. The red dot indicates that the average number of
embedded objects on all web pages within this range is 60.

Figure 3-7 shows that the number of embedded objects on web pages is also an important
factor that affects user experience. According to the preceding analysis, when the number of
embedded objects is 50, the page loading latency is 26~30 RTT in most cases. When the RTT
value is 120 ms, the page loading latency is 3~3.6s in most cases. The whole page display
latency is obtained by adding the DNS request latency for the URL of each object and the
page processing and rendering latency to the page loading latency. The figure shows that
when the number of embedded objects exceeds 50, the page display latency is larger than 3.5s
in most cases.

3.2.4 TCP Characteristics Analysis for Video Services

1. TCP Characteristics of Video Services

Figure 3-8 Statistics on TCP characteristics of video services

White Paper on Network Architecture's Impacts on OTT Service 3 Relationships Between User
Experience and Air Interface Rate Requirements（2014H1） Experience and TCP Throughput Rates

The preceding data sources are obtained from test data on the mLAB LTE network. Data related to TCP
connections only for video segment download is included but that for non-segment-download is not
included.
 When video segments are downloaded, only one segment is transmitted in one TCP
connection currently.
 Figure 3-8 shows that the sizes of initial send windows for LTE and UMTS are
consistent when video segments are downloaded. The sizes of 50% of windows are 2 to
3 and the TCP connections for which the sizes of initial send windows are 10 account for
about 15% to 20%.
 Figure 3-8 shows that the sizes of maximum TCP send windows for LTE when video
segments are downloaded are larger than those for UMTS. The maximum send windows
whose sizes are larger than 30 for LTE account for about 80% while those for UMTS
account only for about 40%. The maximum send windows whose sizes are larger than 60
for LTE account for about 45% while those for UMTS account only for about 15%.
 The sizes of initial send windows determine the initial segment download rate.
Increasing the sizes of initial send windows can reduce the impact of slow start, thereby
improving user experience. The sizes of maximum send windows determine the largest
attainable TCP rate. Larger sizes of maximum windows indicate larger TCP peak rate.
2. Statistics on Initial Video Buffer Size and segment Size During Video Playing

Table 3-1 Video initial buffer and segment statistics

OTT Initial Buffer Size (KB) Segment Size During
Video Playing (KB)

240P 360P 720P 240P 360P 720P

YouTube 250 600 2200 3000 7000 15000
YouKu 300 600 1200 14000 22000 30000
Sohu 250 600 1200 1000 3000 6000

 Currently, video segments on the video clients are downloaded through the single TCP
connection. For details, see the slow start transmission mechanism on page 12 of this
document (assuming that the size of the initial window is 2):
− When the initial buffer size is 250 to 300 KB, the loading is complete within 8 RTT
(1.5 RTT + 6.5 RTT).
− When the initial buffer size is 600 KB, the loading is complete within 10 RTT (1.5
RTT + 8.5 RTT).
− When the initial buffer size is 1,200 KB, the loading is complete within 11 RTT (1.5
RTT + 9.5 RTT).
 Therefore, when the air interface bandwidth and video server performance are not the
bottlenecks, the video service experience is related to RTT. (When the RTT value is 100
ms, the rate reaches 15.3 Mbit/s, 30.7 Mbit/s, and 61.4 Mbit/s at the seventh, ninth, and
tenth RTT, respectively.
 The bandwidth on the live network can hardly exceed 15 Mbit/s in most scenarios.
Therefore, the TCP window size is stabilized after the TCP throughput rate reaches the
air interface bottleneck (according to statistics on Figure 3-8, the size is about 64 x 1500
White Paper on Network Architecture's Impacts on OTT Service 3 Relationships Between User
Experience and Air Interface Rate Requirements（2014H1） Experience and TCP Throughput Rates

bytes, which is stable) and the experienced latency on the network is always larger than
the preceding theoretical value.
3. Analysis of Video Service Experience and RTT

Figure 3-9 Statistics on TCP characteristics of video services

Initial buff duration of 240P Video (CDF) Initial buff duration of 360P Video (CDF)

According to statistics in the Figure 3-9, the initial video buffer latency for 240P videos for
LTE is 0.2s larger than that for UMTS. The initial video buffer latency for 360P videos for
LTE is 0.5s larger than that for UMTS. The RTT on the tested UMTS network is about 25 to
40 ms larger than that on the tested LTE network.
White Paper on Network Architecture's Impacts on OTT Service 4 Comparison of User Experience in
Experience and Air Interface Rate Requirements（2014H1） Typical Network Scenarios

4 Comparison of User Experience in Typical

Network Scenarios

4.1 E2E RTT Analysis in Typical Network Scenarios

Figure 4-1 E2E RTT analysis in typical network scenarios

As shown in Figure 4-1, E2E RTT is the sum of (e)RAN RTT, S1 RTT or Iu RTT, PS Core
RTT, Gi RTT, Internet RTT, and CDN RTT.
This document analyzes only UMTS/HSPA and LTE. The impact of RTT on user experience
in the following scenarios is focused:
 For UMTS and LTE networks, the RTT caused by UMTS and LTE, and its impact on
user experience are focused.
 In the Gi Cache and non-Cache network scenarios, the RTT caused by Internet and its
impact on user experience are focused.
 In the national and international Internet scenarios, the RTT caused by Internet distance
differences and its impact on user experience are focused.
 In the eastern and western China network scenarios, the RTT caused by CDN
deployment locations and its impact on user experience are focused.
White Paper on Network Architecture's Impacts on OTT Service 4 Comparison of User Experience in
Experience and Air Interface Rate Requirements（2014H1） Typical Network Scenarios

Table 4-1 E2E RTT analysis in typical network scenarios

Key Factor
Analysis
UMTS LTE

eRAN RTT (The eRAN RTT on

RAN RTT The air interface invoking mechanism and architecture for LTE and UMTS are greatly
the NGMN is required to be
(RTT for HSPA different. The eRAN RTT has been reduced to smaller than 10 ms. Unlike UMTS, the RNC
smaller than 10 ms and that on
is smaller than node and Iub RTT for LTE do not exist. In addition, RAN RTT is determined by RNC load
the live network smaller than
50 ms) and performance but eRAN RTT is affected only by eNodeB load and performance.
20 ms.)

Iu interfaces are interconnected through backbone networks provided by carriers in most

cases and the RTT is relatively smaller. S1 RTT is similar to Iub RTT. The NGMN requires
IU RTT S1 RTT
the RTT to be smaller than 10 ms and 5 ms is the best. The NGMN requires the RTT to be
smaller than 10 ms and 5 ms is the best.

PS Core RTT (smaller than 10 ms in most cases) Determined by network load.

Determined by the carrier's firewall, and deployment and performance of related
Gi RTT
service inspection NEs (such as DPI).

Determined by whether direct interconnection exists between the OTT content delivery
Internet RTT network and the networks accessed by users, and the transmission performance between
the networks.
Determined by internal architecture of the OTT server, including the architecture of
CDN RTT
the network through which the CDN and OTT server are interconnected.

Currently, most terminals can meet requirements of the current APP performance of
Client receive and send processing latency
processing one task.

Server receive and send processing latency In busy hours, the processing capability of the OTT server may become an RTT bottleneck.

4.2 Comparison of User Experience Between UMTS and

LTE
4.2.1 Comparison of Web Service Experience Between UMTS and
LTE
Figure 4-2 Comparison of web service experience between UMTS and LTE

 The tested web pages are under China's top 30 web URLs. The same web pages are accessed for
LTE and UMTS. Each page is tested for more than 10 times.
 The tests are performed in Shanghai. China Unicom's UMTS/HSPA Ping tests through the Speedtest
show that the average RTT is 93 ms and the LTE Ping test shows that the average RTT is 70 ms.
 When users access China's top 30 web URLs through the LTE network provided by
China Unicom, the average RTT (between the client and server) is 50 ms and the average
page display latency is 4.856s.
White Paper on Network Architecture's Impacts on OTT Service 4 Comparison of User Experience in
Experience and Air Interface Rate Requirements（2014H1） Typical Network Scenarios

 When users access China's top 30 web URLs through the UMTS network provided by
China Unicom, the average RTT (between the client and server) is 106.5 ms and the
average page display latency is 5.63s.

4.2.2 Comparison of Video Service Experience Between UMTS

and LTE
Figure 4-3 Comparison of video service experience between UMTS and LTE

 Sohu, Youku, and iqiyi are tested. Three HD videos are selected for each OTT and each video is
tested for more than 10 times.
 China Unicom's UMTS/HSPA networks are tested. China Unicom's UMTS/HSPA Ping tests through
the Speedtest show that the average RTT is 93 ms and the LTE Ping test shows that the average RTT
is 70 ms.
 When videos provided by Sohu TV are tested on LTE networks, the video source server is located on
the network of Guangdong Telecom. Therefore, RTT is large.
 When users access China mainland's top video websites through the LTE network
provided by Shanghai Unicom, the average RTT (between the client and server) is 64.8
ms and the average initial video buffer download latency is 0.351s which is improved by
over 50% compared with the latency for UMTS.
 When users access China mainland's top video websites through the UMTS/HSPA
network provided by Shanghai Unicom, the average RTT (between the client and server)
is 105.1 ms and the average initial video buffer download latency is 0.801s.
White Paper on Network Architecture's Impacts on OTT Service 4 Comparison of User Experience in
Experience and Air Interface Rate Requirements（2014H1） Typical Network Scenarios

4.3 Comparison of User Experience Between Gi Cache and

Non-Gi Cache Scenarios
4.3.1 Comparison of Web Service Experience Between Gi Cache
and Non-Gi Cache Scenarios for LTE
Figure 4-4 Comparison of web service experience between Gi cache and non-Gi cache scenarios
for LTE

 When users access China's top 30 web URLs through the mLAB LTE network (the Gi
interface is located in Shenzhen), the average RTT (between the client and server) is 135
ms and the average page display latency is 5.22s.
 The preceding top web pages are cached in the server deployed over the Gi interface for
accessing. The average RTT (between the client and server) is 18 ms. The average page
display latency is 0.67s. Experience differences between web pages are slight and user
experience is stable.

4.3.2 Comparison of Web Service Experience Between Gi Cache

and Non-Gi Cache Scenarios for UMTS
Figure 4-5 Comparison of web service experience between Gi cache and non-Gi cache scenarios
for UMTS

 When users access China's top 30 web URLs through the mLAB UMTS network (the Gi
interface is located in Shenzhen), the average RTT (between the client and server) is 165
ms and the average page display latency is 7.663s.
White Paper on Network Architecture's Impacts on OTT Service 4 Comparison of User Experience in
Experience and Air Interface Rate Requirements（2014H1） Typical Network Scenarios

 The preceding top web pages are cached in the server deployed over the Gi interface for
accessing. The average RTT (between the client and server) is 42 ms. The average page
display latency is 1.144s. Experience differences between web pages are slight and user
experience is stable.

4.3.3 Comparison of Video Service Experience Between Gi Cache

and Non-Gi Cache Scenarios for LTE
Figure 4-6 Comparison of video service experience between Gi cache and non-Gi cache scenarios
for LTE

 When users access China mainland's top video websites through the mLAB LTE network,
the average RTT (between the client and server) is 122 ms and the average initial video
buffer download latency is 1.04s.
 Videos are cached in the server deployed over the Gi interface for accessing. The average
RTT is 15.4 ms. The initial video buffer download latency is 178.2 ms. Zero wait time is
achieved. User experience is stable.

4.3.4 Comparison of Video Service Experience Between Gi Cache

and Non-Gi Cache Scenarios for UMTS
Figure 4-7 Comparison of video service experience between Gi cache and non-Gi cache scenarios
for UMTS

 When users access China mainland's top video websites through the mLAB UMTS
network, the average RTT (between the client and server) is 143.9 ms and the average
initial video buffer download latency is 1.454s.
White Paper on Network Architecture's Impacts on OTT Service 4 Comparison of User Experience in
Experience and Air Interface Rate Requirements（2014H1） Typical Network Scenarios

 Videos are cached in the server deployed over the Gi interface for accessing. The average
RTT is 35.5 ms. The initial video buffer download latency is 0.345s. Experience
differences between OTT videos are slight and user experience is stable.

4.4 Comparison Between National and International User

Experience
4.4.1 Comparison of User Experience in Accessing Top30 National
and International Web Pages
Figure 4-8 Comparison between national and international web service experience

 When users access China's top 30 web URLs through the LTE network provided by
China Unicom, the average RTT (between the client and server) is 50 ms and the average
page display latency is 4.856s.
 When users access the top 30 international web URLs through the LTE network provided
by China Unicom, the average RTT (between the client and server) is 333.9 ms and the
average page display latency is 8.016s.

4.4.2 Comparison Between National and International Video

Service Experience
Figure 4-9 Comparison between national and international video service experience
White Paper on Network Architecture's Impacts on OTT Service 4 Comparison of User Experience in
Experience and Air Interface Rate Requirements（2014H1） Typical Network Scenarios

 When users access China mainland's top video websites through the mLAB LTE network
(the Gi interface is located in Shenzhen), the average RTT (between the client and server)
is 121 ms and the average initial video buffer download latency is 1.04s.
 When users access China mainland's top video websites through the mLAB LTE network
(the Gi interface is located in Hong Kong), the average RTT (between the client and
server) is 164 ms and the average initial video buffer download latency is 1.42s.

4.5 Comparison of User Experience Between Eastern and

Western China
4.5.1 Comparison of Web Service Experience Between Eastern and
Western China
Figure 4-10 Comparison of web service experience between eastern and western China

 The tested web pages are under China's top 30 web URLs. The three cities are tested almost within
the same period. Each URL is tested for more than 10 times.
 China Unicom's UMTS/HSPA networks are tested. The Ping test through Speedtest in Shanghai
shows that the average RTT is 93 ms, that in Chengdu shows that the average RTT is 108 ms, and
that in Xi'an shows that the average RTT is 151 ms.

The preceding test results provide the average values of China's top 30 web pages. The
following conclusions can be drawn:
 Web service experience of users in Shanghai, which is located in eastern China, is better
than that in Chengdu and Xi'an, which are located in western China. The average page
display latency in Shanghai is 3.4s and 2.3s smaller than that in Chengdu and Xi'an,
respectively.
 The E2E RTT in Shanghai is 50 ms and 36 ms smaller than that in Chengdu and Xi'an,
respectively.
White Paper on Network Architecture's Impacts on OTT Service 4 Comparison of User Experience in
Experience and Air Interface Rate Requirements（2014H1） Typical Network Scenarios

4.5.2 Comparison of Video Service Experience Between Eastern

and Western China
Figure 4-11 Comparison of video service experience between eastern and western China

 Sohu, Youku, and iqiyi are tested. Three HD videos are selected for each OTT. The three cities are
tested almost within the same period and each video is tested for more than 10 times.
 China Unicom's UMTS/HSPA networks are tested. The Ping test through Speedtest in Shanghai
shows that the average RTT is 93 ms, that in Chengdu shows that the average RTT is 108 ms, and
that in Xi'an shows that the average RTT is 151 ms.
 When users access China mainland's top video websites in Shanghai, the average RTT
(between the client and server) is 105.1 ms and the average initial video buffer download
latency is 0.801s.
 When users in Chengdu access China mainland's top video websites, the average RTT
(between the client and server) is 121.5 ms and the average initial video buffer download
latency is 1.230s. When users in Xi'an access China mainland's top video websites, the
average RTT (between the client and server) is 126.6 ms and the average initial video
buffer download latency is 1.732s.

4.6 E2E RTT and User Experience Summary in Typical

Network Scenarios
Table 4-2 E2E RTT and user experience in typical network scenarios
White Paper on Network Architecture's Impacts on OTT Service 4 Comparison of User Experience in
Experience and Air Interface Rate Requirements（2014H1） Typical Network Scenarios

When users access top 30 international web pages through China Unicom's LTE network, the average
E2E RTT is 334 ms and the average page display latency is 8.16s.
 Gi Cache brings great experience benefits. The experience is stable. Basically zero wait
time for LTE can be achieved.
 Video service experience on China Unicom's LTE network is greatly improved compared
with that on the UMTS network. The initial buffer download latency for HD videos is
reduced to smaller than 0.5s.
 On the live network, most OTT servers are deployed in developed cities such as China's
eastern coastal cities. Therefore, the E2E RTT between the terminal and server in eastern
cities is 20 ms to 50 ms smaller than that in western cities.
 Larger RTT indicates a longer path, more network nodes between the terminal and server,
more factors that cause RTT jitter, and larger RTT jitter. User experience becomes poorer
accordingly when RTT is larger. The impact of RTT jitter on user experience will be
further studied.
White Paper on Network Architecture's Impacts on OTT Service
Experience and Air Interface Rate Requirements（2014H1） 5 Summary

5 Summary

5.1 Service Inflow Determining Air Interface

Requirements
Figure 5-1 Relationship between air interface requirements, and RTT and number of concurrent
TCP connections

Based on the current OTT service design mode, RTT and the number of concurrent TCP
connections together determine service TCP throughput rates and air interface requirements
(the service inflow determines air interface requirements). When the E2E bandwidth is not a
bottleneck, RTT improvement is proportional to user experience improvement. Continuous
improvement of RTT is a key factor that determines user experience improvement.
White Paper on Network Architecture's Impacts on OTT Service
Experience and Air Interface Rate Requirements（2014H1） 5 Summary

5.2 Air Interface Bandwidth and RTT Together Determine

OTT Service Rates on the Wireless Network
Figure 5-2 Relationship between air interface requirements and RTT

When E2E RTT is small, the service inflow (actual service rate) is higher and requirements
for air interface bandwidths are higher accordingly. In this case, it is more likely that air
interface bandwidths will be restricted. When E2E RTT is larger, the service inflow becomes
relatively smaller and requirements for air interface bandwidths are lower accordingly. In this
case, it is more likely that air interface bandwidths are not fully utilized.

5.3 Factors Affecting E2E RTT

Figure 5-3 Factors affecting E2E RTT

Many factors affect the E2E RTT between a terminal and OTT server. Such factors include
wireless network performance, OTT CDN architecture (CDN RTT) and their deployment
position (Internet RTT), and application server and terminal performance. Wireless network
factors affecting E2E RTT include but are not limited to the following: wireless network
deployment architecture (such as the CN integrated deployment policy, the interface network
between NEs, and Gi interface connect to OTT CDN directly), network capacity, air interface
coverage, RATs, air interface scheduling algorithms, related parameters, and QoS policies
deployed on the network. Based on the preceding factors, Huawei provides a series of product
features and solutions (such as xMbps) to help customers construct networks that provide best
White Paper on Network Architecture's Impacts on OTT Service
Experience and Air Interface Rate Requirements（2014H1） 5 Summary

user experience. In addition, Huawei provides corresponding network architecture

optimization services to continuously reduce E2E RTT.

5.4 Air Interface Optimization and Architecture

Optimization Together Achieve Optimal User Experience
Figure 5-4 Air interface optimization and architecture optimization together improving user
experience

 User experience is determined not only by air interface coverage (xMbps) but also by the
RTT caused by network architecture.
 When the air interface bandwidth is not restricted, user experience is related to RTT.
Attention needs to be focused on network architecture optimization to reduce RTT. The
optimization includes RTT optimization for wireless networks and OTT network
architecture optimization (such as CDN deployment).
 When the link bandwidth is not a bottleneck, smaller RTT indicates higher TCP
throughput rates and higher requirements of services for air interface bandwidths.
 Air interface coverage optimization must be synchronized with RTT optimization to
achieve greatest user experience improvement at the lowest costs.

5.5 Routine Work Introduction for mLAB User Experience

Study
5.5.1 Objective
 Perceive the OTT service experience status and change trend.
 Study OTT network requirements (bandwidth and RTT) and development.

5.5.2 Working Mode

 In the OTT experience and network requirement research field, mLAB provides a
professional and efficient tool, MBB Explorer, as well as a large number of experienced
White Paper on Network Architecture's Impacts on OTT Service
Experience and Air Interface Rate Requirements（2014H1） 5 Summary

experts. In addition, mLAB has established long-term partnerships with well-known

universities and professional research institutes inside and outside China.
 mLAB experience and network requirement research center provides the routine output
every six months. Related research results are widely used on the networks constructed
by Huawei or the networks on which Huawei performs operation and maintenance.
These research results serve as the basis of Huawei xMbps solution.
 Monitor the QoE and RTT of mainstream carrier networks routinely to provide
references for optimization of carriers' networks.

5.5.3 Contact
The author of this document is He Wensheng whose employee ID is 00150170 and email is
hewensheng@huawei.com.
He Wensheng is in charge of mLAB experience requirement research center.
You can contact Huawei mLAB personnel ((MBB lab) through MBBlab@huawei.com.

5.5.4 Disclaimer
This report is prepared by Huawei mLAB. Information provided in this report is for reference
only. Data used for this report is collected by mLAB MBB Explorer. Due to the limited
sample number and the rapid development of tested networks and OTT services, Huawei
mLAB reserves the rights to update this report in the future, and does not bear liabilities for
any impacts that may arise due to the update.
This report cannot be used as a basis for investment or research decision-making, or as a basis
or proof for ethic, responsibility, or law purpose, either expressed or implied. Huawei mLAB
may supplement, correct and revise information in this report without notice. Every effort has
been made in the preparation of this report to ensure accuracy of the information. Huawei
mLAB does not bear liabilities for any direct or indirect investment profit and loss due to
information in this report.
No part of this report may be reproduced, transmitted, published in any form or by any means
without prior written consent of Huawei mLAB. Shall information in this report needs to be
quoted, it must be clearly stated that the information is prepared by Huawei mLAB, and the
quoted information must not be abbreviated or modified deliberately against its original
meaning.
White Paper on Network Architecture's Impacts on OTT Service
Experience and Air Interface Rate Requirements A Terms

A Terms

Term Definition
APP Application
CDN Content delivery network
DNS Domain Name System
GGSN Gateway GPRS Support Node
Gn Interface between the serving GPRS support node (SGSN) and
gateway GPRS support node (GGSN)
HTTP Hypertext Transfer Protocol
Iub Logical interface between the RNC and NodeB
Iu-PS Interface between the RNC and SGSN in the packet switched (PS)
domain
LTE Long Term Evolution
MBB Explorer OTT service experience evaluation system developed by mLAB.
MBB Explorer consists of APP clients and a data center. MBB
Explorer performs web and video service experience tests, DNS
tests, and Ping RTT tests into which Speedtest is integrated.
MSS TCP maximum segment size
OTT Over-the-top. In this document, OTT refers to different types of
MBB services or service carriers on wireless networks.
PGW Packet Gateway
RAN Radio Access Network
RTT Round-Trip Time
S1 Interface between the LTE eNodeB and EPC SGW
SGSN Serving GPRS support node
SGW Service Gateway
White Paper on Network Architecture's Impacts on OTT Service
Experience and Air Interface Rate Requirements A Terms

Term Definition
TCP Transmission Control Protocol
Number of concurrent TCP connections Number of TCP connections that exist at the same time
TCP initial window Initial TCP send window
TCP send window Maximum number of bytes that can be consecutively sent by the
TCP sender
TCP receive window Maximum number of bytes that the TCP receiver allows the TCP
sender to consecutively send
TCP throughput rate Size of data blocks transmitted per time unit over TCP. TCP
throughput rate is also named TCP data rate.
TCP congestion control window cwnd is short for congestion window. The congestion window is a
flow control mechanism used by the sender, whereas the TCP
receive window is a flow control mechanism used by the receiver.
Maximum TCP send window Maximum number of bytes that can be consecutively sent between a
TCP client and server
UMTS Universal Mobile Telecommunications System
Web object Web Object: HTTP object, which is used for processing and
rendering the webpage and is referenced by the page mark-up or
script. However, these objects are not necessarily visible on the fully
rendered page (cf. elements).
Web Element: Visual content of a webpage, which is displayed to the
user on the rendered webpage. E.g. Text, Pictures, Widgets, Videos,
etc.
Web page display latency From the time when a user clicks a web link or enters a URL to the
time when a whole web page is displayed
Air interface service inflow Volume of downlink data received by a base station per second or
volume of uplink data sent from the terminal application layer to the
terminal wireless module
Video initial buffer latency From the time when you click a video to play it to the time when a
video picture is displayed
Video initial buffer download latency Duration in which there is video segment data interaction between a
video client and server from the time when you click a video to play
it to the time when a video picture is displayed
White Paper on Network Architecture's Impacts on OTT Service
Experience and Air Interface Rate Requirements B References

B References

1. Next Generation Mobile Networks Beyond HSPA & EVDO A White Paper By Board Of
NGMN Limited
2. Next Generation Mobile Networks Radio Access Performance Evaluation Methodology
By the NGMN Alliance
3. Understanding Website Complexity: Measurements, Metrics, and Implications Michael
Butkiewicz UC Riverside butkiewm@cs.ucr.edu, Harsha V. Madhyastha UC Riverside
harsha@cs.ucr.edu, Vyas Sekar Intel Labs vyas.sekar@intel.com
4. TCP/IP Detailed Description, Volume 1: Protocol
5. Li Wenbin, Geng Bo, TCP Throughput Theory Analysis, June 2009.
6. M. Mathis, J. Semke, J. Mahdavi and T. Ott, "The macroscopic behavior of the TCP
congestion Avoidance Algorithm", Computer Communications Review, vol. 27, no. 3,
pp 67-82, July 1997.
7. J. Padhye, V. Firoiu, D. Towsley and J. Kurose, "Modeling TCP Reno performance: A
simple model and its empirical validation", IEEE/ACM Trans. on Networking, vol. 8, no.
2, pp. 133-145, April 2000.
8. Han Tao, Zhu Yaoting, TCP Throughput Model Related to Slow Start, Electronic
Journal, vol. 30, no. 10, pp 1482-1483, October 2002.
9. Pan Yajun, TCP Throughput Measurement Algorithm Study (master's thesis), Beijing
Normal University, June 2005.
10. RFC 2988, "Computing TCP's Retransmission Timer", November 2000
11. Stas Khirman and Peter Henriksen. Relationship between Quality-of-Service and
Quality-of-Experience for Public Internet Service. 2000.3 3950 Fabian Way, Palo Alto,
CA 94303
12. Alessandro Anzaloni and Alexander Bento Melo. TCP PERFORMANCE IN
WIRELESS CHANNELS. 2002.5. Instituto Tecnológico de Aeronaútica- BRAZIL