Packet Error Rate and Latency Requirements for a Mobile

Wireless Access System in an IP Network

Lai King (Anna) Tee {atee@sta.samsung.com}
Wireless Solutions Lab, Samsung Telecommunications America
1301 E. Lookout Dr., Richardson, TX 75082
Abstract - In this paper, the requirements on latency and packet MSS = Maximum Segment Size in bytes
error rate for an IP network based wireless system will be RTT = Average round trip time: average duration between
discussed. A TCP model will be used to show the effects of the time when a segment was transmitted to the time when
latency and packet loss rate on throughput performance. A an acknowledgement arrived for that segment.
survey of published QoS requirements for various portions of an
end-to-end IP network with wireless air interface will also be
reviewed and compared. The objective is to derive a range of In practice, the TCP throughput could be quite different when
latency and error rate requirements for the wireless air interface packet losses are considered together with the behavior of
for various types of applications, to reduce excess delay or TCP such as slow-start, congestion avoidance, triple-duplicate
throughput limitation at the TCP layer, and to ensure acknowledgements and time-out effects. The PFTK model as
compliance with the QoS support at the IP network, so as to meet described in [3] has captured these TCP behaviors in the
the end-to-end performance requirements as expected by the realistic wireless links with packet losses. The PFTK model is
user, according to the application class. a model for the TCP Reno congestion avoidance mechanism
for bulk transfer packet flows. Thus the variations in the
I. INTRODUCTION receive window sizes as caused by packet losses are modeled.
The effects of packet error rate and delay on TCP/IP Specifically, PFTK model has included effects of:
throughput based on the PFTK model are computed in Section ƒ Triple duplicate acknowledgment
II. The error rate and latency requirements for various ƒ Time-out retransmission
standards are reviewed in Section III, followed by a discussion ƒ Maximum window limitations
on the IP network delay in Section IV. Traffic classifications
as supported by IETF are reviewed in Sections V. Section VI Based on TCP Reno, a packet is re-transmitted after the triple-
will suggest a set of requirements for an air interface that can duplicate condition is detected, and the congestion window
support various applications in an IP based network, followed size will be reduced by half. On the other hand, if the re-
by the conclusion in Section VII. transmission (packet loss) is caused by the time-out condition,
the congestion window will be reduced to one segment.
II. EFFECTS OF WIRELESS LINK ON TCP PERFORMANCE
TCP is originally designed for use in the wired network to Parameters of the PFTK model can be defined as follows:
control the flow of IP packets. As reported in previous work p = prob( packet loss )
[1], the performance of TCP throughput over wireless links To = Time Out
deteriorates significantly because of errors caused by the
W ( p) = Window size
wireless link. For example, it is reported that the TCP
throughput is only 4.3 Mbps, while the physical bit rate is 11 Q( p, W ) = prob(loss caused by Time − out )
Mbps for IEEE 802.11b, i.e., only 39.1% of the maximum
data rate is achieved. The implication is that, although the TCP throughput can be computed as follows:
physical bit rate is 11 Mbps as supported by the physical layer
design (PHY), 60.9% of this throughput is lost to overheads in  1 − p W ( p)
+ + Q( p, W ( p ))

the PHY & MAC headers and relatively high error rate at the 
p 2
, W ( p) < Wm
air interface. Although the error or lost packets can be  RTT (W ( p ) + 1) + Q ( p ,W ( p ))G ( p )To
 1− p
retransmitted, the additional delay results in time-out events RTCP ( p ) = 
 1 − p W
detected at the TCP layer, reducing the effective throughput.  p
+
2
m
+ Q ( p , Wm )
 , otherwise
 RTT ( m + − p + 2) +
W 1 Q( p, Wm )G ( p )To
The dependency of TCP throughput on latency in the  4 pWm 1− p
transmission for error free, loss free transmissions may be
computed using Equation 1 [2]: Where:
W ⋅ MSS 2 4(1 − p ) 4
RTCP = m W ( p) = + +
RTT 3 3p 9
… (1)  (1 − (1 − p) 3 )(1 + (1 − p) 3 (1 − (1 − p )W −3 )) 
where: Q( p, W ) = min 1, 
 1 − (1 − p) W 
Wm = the maximum TCP receive window in segments
G ( p) = 1 + p + 2 p 2 + 4 p 3 + 8 p 4 + 16 p 5 + 32 p 6

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The TCP throughput based on the above model was computed
for ranges of packet loss rates and round trip delays for
maximum window size of 123 segments with maximum
segment size MSS of 536 bytes. The results are plotted as
shown in Figure 1.
To derive the requirements on packet error rates and latencies
for the air interface of a wireless network, the effects of TCP
flow control need to be considered for applications that are
based on TCP, to ensure the end-to-end quality of service
requirements of the users can be met.

The results show that for the selected values of maximum 8

10
Effects of Round Trip Delay and Packet Loss Rate on TCP Throughput

window and segment sizes, the TCP throughput decreases Packet lost rate = 0.0001
0.001
significantly as the packet loss rate increases beyond 10-4. It 0.01
Max. congestion window = 123 segments
can also be observed that at low packet loss rates, the TCP MSS (Max. Segment Size) = 536 bytes
PFTK TCP Reno model
throughput decreases rapidly as the values of round-trip delay 7
10
RTT increases. As the packet loss rate increases, it becomes
the limiting factor for the TCP throughput, even when RTT is

TCP Throughput (bps)

low. The variation of TCP throughput as a function of RTT,
for packet loss rates of 1%, 0.1% and 0.01% is shown in 6
10
Figure 2.

Based on the computation results of the model, as shown in

Figure 1 and Figure 2, the packet loss rate should be better 5
10

than 10-4 while keeping the round trip delay as low as possible
for the best performance in terms of TCP throughput.

Effects of Round Trip Delay and Packet Loss Rate on TCP Throughput 4
10
8
10 0 100 200 300 400 500 600 700 800
Round Trip Delay (ms)

7 Figure 2 Changes in TCP throughput performance as round trip

10
delay, at different packet loss rates.

OVERVIEW OF REQUIREMENTS IN STANDARDS

6
10 III.

In this section, the packet error rate and latency requirements

TCP Throughput (bps)

of a few published standards will be reviewed. These include

10

requirements specified for the network layer and application

layer. A review of these requirements will provide information
4
10 RTT = 10 ms
RTT = 50 ms
RTT = 100 ms
RTT = 150 ms on the corresponding requirements for the air interface.
3 RTT = 250 ms
10 RTT = 400 ms
Max. congestion window = 123 segments
MSS (Max. Segment Size) = 536 bytes A. Data applications that are based on TCP
PFTK TCP Reno model
Most of the current Internet traffic such as HTTP, FTP is
2
10

based on TCP flow control. TCP uses the checksum to

10
1
−5 −4 −3 −2
10 10 10 10
Packet loss rate
Figure 1 TCP throughput performance at different packet loss rates
and round-trip delays
Data transmission over a wireless network is far more
challenging than in a wired network. This is because the time-
varying, dispersive characteristics of the mobile radio
propagation channel can result in higher packet loss. In
addition to propagation delay, there can be additional delays
as the limited radio resource is shared among a number of
users, e.g., access delay, scheduling delay. Packet losses may
result in the re-transmission of packets incurring further
delays. In general, the round-trip delay of a data packet varies
with a range that is much wider than that of wired network.
Thus, when a transmitted packet experiences a delay longer
than the Time-Out value, the TCP protocol will react as if the
packet has been lost. This will result in re-transmission of the
same packet, consuming air interface resources unnecessarily.
−1 0
identify any error in the IP packet. When an error is detected,
10 10
the IP packet will be re-transmitted. While this ensures the
data that arrives at the receiving end could have no
information loss eventually, there will be additional delay
caused by the re-transmission as shown in the previous
section. Thus, for applications that are delay sensitive, it may
not be applicable to use TCP as the transport protocol. Instead,
UDP or RTP may be used. UDP does not cause re-
transmission of IP packets. If the checksum feature is enabled,
an IP packet will be dropped when any error is detected.
Otherwise, the whole IP packet will be forwarded regardless
of the error.
1) Requirements on IP error rates in ITU-T Y.1541
The performance objectives for different classes of traffic in
an IP packet network are specified in ITU-T Y.1541 [4]. The
parameters that are used in Y.1541 are defined in ITU-T
Y.1540 [5]. These parameters include IP packet transfer delay
(IPTD), IP packet delay variation (IPDV), IP packet loss ratio
(IPLR) and IP packet error ratio (IPER). Six different classes

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of traffic based on IPTD and IPDV objectives are defined. The
upper bounds specified for IPTD range include: 100 ms, 400
ms and 1 s. Upper bound on IPDV is 50 ms for classes 0 and
1, but not specified for the other 4 classes. Except for the
unspecified class 5, upper bounds on IPLR and IPER are 10-4
and 10-3 respectively.
2) Requirements on error rates in IEEE Std. 802 [6]
The IEEE standard for local and metropolitan area networks:
Overview and Architecture, IEEE Std. 802-2001, defines the
compliance with the family of IEEE 802 Standards. It
describes the relationship of the IEEE 802 standards to the
OSI Basic Reference Model and explains the relationship of
these standards to the higher layer protocols.
Subsection 7.3 of IEEE Std. 802-2001 states that the require
error performance of IEEE 802 LANs and MANs shall be less
than 8x10-8 per octet of MAC service Data unit (MSDU)
length. While this error performance has to be accomplished at
the physical layer for wired or optical fiber physical media, it
is allowable for this error performance to be accomplished at
the MAC service boundary in the case of wireless media.
For example, for an MSDU packet with 1500 octets, the
required packet error rate will be approximately 1.2x10-4:
PER ≈ 1500 × 8 × 10 −8 = 1.2 x10 −4
Note that this value agrees closely with the IPLR requirement
specified in ITU-T Y.1541 as discussed in Section 1.
3) Error Rate Requirements in IEEE 802.16.3 [7]
Error rate requirements for the wireless metropolitan area
network, IEEE 802.16.3, are specified in the section for
quality of service support for different classes of services. The
maximum BER allowable are 10-6 and 10-4 respectively for
full and standard quality telephony respectively. The
corresponding maximum one-way access delay, as measured
at the MAC interface with the upper layers, are 20ms and 40
ms respectively. For time critical packet services, the
maximum BER required by 802.16.3 is 10-6, with maximum
one-way access delay of 20ms, whereas the BER requirement
for non-time critical services is 10-9, without requirement on
the maximum delay. The corresponding octet and packet error
rates can be computed as follows. Assuming bit errors are
independent, for BER = 10-9,
Octet Error Rate ≈ 1 − (1 − BER ) = 8 × 10 −9
8
Thus, for a MSDU with 1500 octets, the packet error rate,
assuming independent octet errors, is approximately 1.2x10-5.
Comparing the error performance requirement for 802.16.3
non-time critical packet services with that specified in IEEE
Std. 802-2001 in subsection 2), this exceeds the latter
requirement by approximately 10 times.
4) Error rate performance supported by 3GPP
The ranges of error rate performance that are supported by
3GPP have been specified as part of the QoS requirements in
[8]. The ranges of BER that the network is required to support
are: 10-3 to 10-7 for real time applications, 10-5 to 10-8 for non-
real time applications. Further details are given in [9] which
specified a range of BER and SDU error ratio requirements for
each of the four ITU traffic classes, i.e., Conversational,
Streaming, Interactive and Background classes. Assuming bit
errors are independent, for BER = 10-8, octet error rate is
approximately 8x10-8. Again, assuming independent octet
errors, the MSDU packet error rate is approximately 1.2x10-4.
This result is 10 times higher than that for IEEE 802.16.3 in
the Section 3), while consistent with that in Section 2).
5) Error rate performance supported by GSM
The error rate performance of GSM as reported in [1] can
support a bit error ratio of 10-3, which is then reduced to 10-8
in the nontransparent mode radio link protocol (RLP), at the
expense of variable, additional delay due to retransmissions,
reducing the user throughput.
B. Video over IP error rate requirements
For real-time video signal, the contribution to ITU-T SG13
[10] has stated that IPLR must be at least 10-5. This is derived
based on a BER of 10-9 for typical fiber optic network, and the
worst-case assumptions that the packet size is 1500 bytes and
that a bit error caused the whole packet to be lost.
In most cases, User Datagram Protocol (UDP) is used for the
transport of video streaming applications. If the UDP
checksum is enabled, a packet may be discarded because of a
single bit error. Since UDP does not allow a re-transmission of
the lost packet, the effects of losing a complete video packet
could result in serious disruption to the video signal. However,
the UDP checksum is normally not enabled. In that case, the
packet with bit errors will be received together with the error
bits. Depending on the location of the error bits, the effects of
the lost may be tolerable to the user at the receiving end.
C. Requirements for various applications and service classes
Some examples of error rate and delay requirements are given
in the 3GPP technical specifications for services and QoS
architecture [8], [9] & ITU-T recommendations G.114 [13],
G.1010 end-user multimedia QoS categories [14]. These are
summarized in the following subsections.
1) Real-time Conversational classes
For conversational voice, acceptable performance for
maximum frame erasure rates is 3%. The requirement is also
dependent on the error resilience of speech codec. The
requirements for AMR codec is shown in Table 1.
For videophones, the delay requirement should be similar to
that of conversational voice, with additional requirement to
the limits of lip-synch. The maximum acceptable FER is 1%.
For interactive games, one-way delay value of 250 ms has
been proposed in [8]. Detail studies on multiplayer network
gaming have been reported in [11]. The studies showed that
the range of acceptable maximum round-trip time varies
between 200 ms and 40 seconds depending on the type of
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games, the experience level of the players. Round-trip time is IV. LATENCY OR PACKET TRANSFER DELAY
the total time for a packet to travel from the server to the client
A. Statistics of Internet delay
and the time used to send a reply back from the client to the
server. Delay tolerance is based on the nature of the Internet Some studies have been performed regarding the statistics of
game: Action (Conversational), Real-time (Interactive) or Internet delay for local and International routes [12]. The
Turn-based (Interactive/Background), as shown in Table 1. probability distribution function of one-way Internet delay is
The residual BER requirements correspond to 10-2/10-3, found to follow the shifted Gamma distribution. The mean
10-4/10-6 SDU error ratio depending on the SDU size. delay values for local and International routes are about 10 ms
and 110 ms respectively.
For Two-way control telemetry, an one-way delay limit of 250
ms is proposed in [8], because of the importance of receiving Delays in the public IP network may not be controllable nor
this category of data in a timely manner. In addition, there guaranteed until QoS capabilities such as DiffServ or IPv6 are
should be “0” information loss at the receiving end. widely deployed. However, as the importance of including an
efficient IP network increases, service providers may route the
Table 1 Error rate and delay requirement for various application types
data traffic through private IP network, which may provide
Application Type/ Bit Error Rate End-to-end Delay Source ample capacity to avoid the packet delays due to congestion.
Bit rate As an example, some statistics on packet round-trip time
-4
Conversational Class 1: 10 Preferred: 0-150 ms G.114
voice (AMR codec)/ Class 2: 10-3 Acceptable: G.1010
(RTT) have been collected through Sprintlink Looking Glass
4.75-12.2 kpbs Class 3: 10-2 150-400 ms tools. The average RTT results for a ping packet with a size of
Multiplayer Gaming Action: 10-3 <= 80 ms [11] 100 bytes are summarized in Table 2. In general, the delay in
Interactive: 10-5 250 ms [8] sending packets within the same state is about 15 ms; between
/6x10-8 US East and West coasts is about 75 ms; between the US East
Turn based: 6x10-8 < 40 s
Two-way control ~0 (end-to-end) < 250 ms [8]
coast and Asia/Australia is about 210ms/225 ms, and between
telemetry the US West coast and Europe is about 160 ms.
Web browsing, E- ~0 (end-to-end) Interactive: 2-4 s /
Table 2 Average round-trip time from Sprintlink looking glass
Commerce, Email 0.5s (recommended)
-2
Audio streaming 10 (packet loss) 10 s [14] Stockton,CA Hong Kong Sydney Hamburg
Video Streaming 10-6 – 10-3 150 – 400 ms New York 75 ms 213 ms 225 ms 86 ms
(MPEG4)/ Anaheim, CA 15 ms 158 ms 168 ms 158 ms
24-128kps
Fax, SMS 10-6 30 s [14] B. TCP Round Trip Delay
For a single user link, it can be observed from Figure 1 that it
2) Interactive class would be close to ideal to have RTT of 10 ms in order to
Applications that fall into this class include voice messaging achieve TCP throughput of about 40 Mbps at IPLR of 10-4.
which has similar error rate requirement as that of the While a high user throughput is desirable for applications that
conversational class, but the delay can be up to a few seconds. require block transfer of significant amount of data, it may not
Other applications in this class include: web browsing, E- be realistic to have RTT of 10ms due to the constraints on the
Commerce and Email, as shown in Table 1. PHY and MAC layers, fairness considerations in the
scheduling algorithm in a multiple access network. Moreover,
3) Streaming class the delay in IP network as discussed in the above section
Audio streaming is mainly an one-way application from the indicated that the RTT exceeds 10 ms in most cases.
server to the user. Specific contents of the audio stream may
include high quality music or broadcasting. Thus the error rate V. IETF QoS CLASSIFICATIONS
requirement is more stringent than conversational class but the The Internet Engineering Task Force (IETF) has defined a
delay requirement is more relaxed in comparison. service classification [15] for DiffServ in order to support the
performance requirements for the end user as recommended
The situation with video streaming is similar to audio by ITU-T [4, 5, 14]. The service classes are shown in Table 3
streaming. For MPEG-4 video, the BER ranges between 10-6 with the associated DiffServ Code point associated with each
and 10-3 with significant degradation for the latter. Similar class. It can be observed that further granularity in
requirements apply to the transmission of still image. The classification is provided in the Assured Forwarding (AF)
error tolerance is mainly dependent on the encoding and class.
compression formats. The requirements for audio and video VI. RECOMMENDATION
streaming are shown in Table 1.
As it is desirable for a mobile broadband wireless access
4) Background class system to support a variety of traffic classes, it is necessary to
There is no stringent requirement on delay for the background define the latency and packet error rates performance
class of services. The requirement for fax, and low priority requirements, in order to meet the end-user QoS requirements
transaction services such as short message services (SMS) can for the various applications, as recommended by ITU G.1010,
be found in Table 1. Y.1541. These traffic classes should be mapped to the

252
appropriate QoS classes as defined at the IP network. control are located within reasonable network delay.
Depending on the network configuration, the air interface (AI) Furthermore, when the network delay is the dominating factor,
should support appropriate latency and packet error rate the end-to-end delay is less sensitive to the delay at the AI .
performance targets associated with each traffic class, such
VII.CONCLUSION
that the end-to-end QoS requirements for these applications
can be achieved. Based on the PFTK model for the TCP flow control protocol,
a recommendation for the error rate and latency requirements
Table 3 Service classes, DiffServ code point mapping defined by IETF
for the air interface of a mobile broadband wireless access
Service Class DSCP name Application Example system that supports various IP based applications have been
Administrative CS7 Heartbeats derived, taking into consideration the corresponding
Network Control CS6 Network Routing
requirements for portions of the end-to-end IP network.
Telephony EF, CS5 IP Telephony
Multimedia AF41-AF43 Video Conferencing, ACKNOWLEGMENT
Conferencing Interactive Gaming
Multimedia AF31-AF33, CS4 Broadcast TV, Pay per View, The author thanks Mr. Jim Landon for providing the reference
Streaming Video Surveillance materials and his valuable comments and discussions.
Low Latency AF21-AF23, CS3 Client/Server transactions,
Data peer-to-peer signaling REFERENCES
High Throughput AF11-AF13, CS2 Store & Forward applications, 1. G. Xylomenos, F. Polyzos, P. Mahonen, M. Saaranen,
Data Non-critical OAM&P
Standard DF (CS0) Undifferentiated applications ‘TCP Performance Issues over Wireless Links’, IEEE
Low Priority CS1 Any flow that has no BW communications Magazine, April 2001.
Data assurance 2. Jean Walrand, ‘Communication Networks-A first course’,
McGraw Hill, 1998
In the case of IETF DiffServ, a recommendation on the 3. J. Padhye, V. Firoiu, D. Towsley, J. Kurose, “Modeling
requirements for the AI is shown in Table 4, based on the TCP Reno Performance: A Simple Model and Its
following assumptions: Empirical Validation”, IEEE/ACM Trans. On
Networking, Vol. 8, No.2, April 2000.
i) MAC SDU size = 1500 bytes
4. “Network performance objectives for IP-based services”,
ii) Network Delay = 20 ms or 100 ms
ITU-T Y.1541, May 2002.
iii) Size of IP packet: 1 MAC SDU / IP packet
5. “Internet Protocol Data Communication Service – IP
iv) For UDP traffic:
packet transfer and availability performance parameters”,
End-to-end One-way Delay ≈ AI Delay + Network Delay
ITU-T Y.1540, Jan 2002.
Table 4 Recommendation on requirements for the air interface delay 6. “IEEE Standard for local and metropolitan area networks:
and target packet error rate Overview and Architecture”, IEEE Std. 802-2001, March
IETF Service class Transport End-to-end Network Max. AI MAC SDU
8, 2002.
Protocol One way Delay Delay Packet 7. “Functional Requirements for the 802.16.3
Delay (s) assumed (ms) (ms) Error Rate Interoperability Standard”, IEEE 802.16.3-00/02r4,
Administrative TCP 0.10* 10 10 10-6 September 22, 2000.
Network Control
8. 3GPP TS 22.105 V 6.2.0 (2003-06), Technical
[8, 9,14] TCP 0.25* 10 20 10-6 Specification Group Services and Systems Aspects,
Telephony [8,14] Service Aspects; Services and Services Capabilities.
(Voice over IP) UDP/ RTP 0.15 100 50 3 x 10-2
Multimedia 9. 3GPP TS 23.107 V 5.10.0(2003-09), Technical
Conferencing [8,14] UDP/ RTP 0.15 100 50 10-3 Specification Group Services and Systems Aspects,
Multimedia
Streaming [8,14] UDP/ RTP 0.25 100 150 10-3
Service Aspects; QoS concept and architecture(Release 5).
Low Latency Data 10. Video Performance requirements for IP performance
[8,14] TCP 2* 10 20 10-5 recommendations,ITU-T SG13 D.228(WP4/13),Jan 14,02.
(E-transactions)
High Throughput 11. Multiplayer Game Performance over Cellular Networks,
Data e.g., Email TCP 6* 10 20 10-4 Ver. 1.0, Forum Nokia, Jan 20, 2004
[4, 8, 9,14]
Standard: e.g., FTP
12. Statistics of One-Way Internet Packet Delays, A. Corlett,
[4, 8, 9,14] TCP 10* 10 30 10 -4 D. Pullin, S. Sargood, 53rd IETF, March 18, 2002.
Low Priority Data 13. ‘One-way transmission time’, ITU-T G.114, May 2003
[4, 8, 9,14] TCP 100* 10 40 10-3
*Time required to transfer the all the packets for the email or file.
14. ‘End-user Multimedia QoS Categories’, ITU-T G.1010,
Nov. 2001
As the network delay varies greatly across different networks, 15. F. Baker et. al., IETF Draft “Configuration Guidelines for
under different loading conditions, the values shown in the DiffServ Service Classes draft-baker-diffserv-basic-
table are selected to emphasize the significance of AI delays classes-02”, Feb 13, 2004.
under the selected network conditions [16]. In addition, the 16. ‘A new option proposed for 802.20 requirements on
recommendation is based on the assumption that the data latency and packet error rates’, IEEE C802.20-04-55,
servers for time critical services in the administrative, network May 10, 2004.

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