HUAWEI SE2900 Session Border Controller

V300R002
Technical White Paper for QoS

Issue 01

Date 2016-01-15

Huawei Technologies Co., Ltd.

Copyright © Huawei Technologies Co., Ltd. 2016. All rights reserved.
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About This Document

Overview
This document is not for the communication with customers and therefore is not a
commitment to customers. The MO is responsible for the preparation of the materials used in
the communication with customers. The solutions described in this document may not be
implemented in the released version. For details about the schedule and implementation,
contact the SBC Product Management Department of Core Network Product Line.

Intended Audience
This document is intended for the following engineers who participate in the life cycle of the
SE2900:
 Marketing engineers
 Technical sales engineers
 Product management engineers

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Contents

About This Document ....................................................................................................................... ii

1 Overview .......................................................................................................................................... 1
1.1 Introduction .................................................................................................................................................................. 1
1.2 Value ............................................................................................................................................................................. 2
1.3 Service Models ............................................................................................................................................................. 2

2 Key Technologies ............................................................................................................................. 4

2.1 Anti-Packet Loss and Anti-Jitter Technologies ............................................................................................................. 4
2.1.1 Packet Loss Compensation ........................................................................................................................................ 4
2.1.2 RFC2198 Redundancy ............................................................................................................................................... 4
2.1.3 De-Jitter Buffer .......................................................................................................................................................... 5
2.2 Bandwidth Control Technologies.................................................................................................................................. 6
2.2.1 Policy Control (Rx) .................................................................................................................................................... 6
2.2.2 Call Admission Control ............................................................................................................................................. 6
2.2.3 Committed Access Rate ............................................................................................................................................. 7
2.2.4 Voice Activity Detection/Comfort Noise Generation ................................................................................................. 8
2.3 Forwarding Priority Control ......................................................................................................................................... 8
2.3.1 Differentiated Services Code Point ............................................................................................................................ 8
2.4 QoS Monitoring ............................................................................................................................................................ 9
2.4.1 Voice Quality Reporting ............................................................................................................................................ 9

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Technical White Paper for QoS on SE2900

V300R002
Key words: QoS
Abstract:
List of abbreviations:

Abbreviations Full spelling

QoS Quality of Service
CAR Committed Access Rate
MOS Mean Opinion Score

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1 Overview

1.1 Introduction
Quality of Service (QoS) refers to the capability of the service provider to meet needs of its
customers. For network services, QoS involves concepts such as the bandwidth, latency, jitter,
and packet loss rate. QoS can be used to manage or avoid network congestions and reduce the
packet loss rate by providing differentiated quality for various types of network services, such
as audio, video, and data services.

Major QoS indicators are as follows:

 Bandwidth/throughput
The maximum transmission rate between two network nodes in byte/s or bit/s is termed
as the throughput.
 Packet loss rate
The percentage of lost packets within a certain period of time during network
transmission is termed as the packet loss rate. In normal cases, the packet loss rate is 0.
When network congestion occurs, packets can be discarded as required.
 Latency
The duration from the time point when a packet is sent from a node to the time point
when the packet is received at another node is termed as the latency. For a network
device, the latency refers to the duration between the arrival of the first bit of a data
stream to the departure of the last bit of the data stream.
 Jitter
The variation in packet delay is termed as the jitter. The latencies for different types of
data packets within the same connection are different. For example, the first packet
within a connection takes 10 ms to be transmitted from a network node to another, and
the second packet within the connection takes 15 ms. The jitter between the two
transmissions is 5 ms.

QoS can be used to implement the following functions:

 Categorization
Categorization refers to the capability to distinguish the data packets of each application
from those of other applications. With the categorization function, the SE2900
implements differentiated management on the data packets of various applications.

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 Tagging
A data packet must be tagged after being identified so that it can be processed by
switches or routers with a proper priority.
 Priority marking
Data packets of different services must be allocated with different priorities so that
important services can be ensured when the network is overloaded or congested. All the
service traffic must be identified by the devices on the backbone network so that
different services are properly processed based on their priorities.

1.2 Value
QoS can guarantee end users to obtain the expected service experiences. For example, the
maximum downlink rate is 1 Mbit/s when end users download files from the Internet; the
maximum downlink rate is 512 Kbit/s when end users use P2P applications; VoIP services
with acceptable voice quality are available when end users browse web pages on the Internet;
Smooth video streaming is available when end users send and receive email messages.
QoS can help carriers gain the following benefits:
 Guaranteeing service quality
QoS enables carriers to accurately manage bandwidth resources and provide standard
services for end users.
 Earning interest on differentiated services
QoS enables carriers to provide differentiated services for different types of users.
Specifically, carriers can categorize users into different groups, allocates different
bandwidths and priorities to users in these groups, and charges them accordingly.
 Ensuring user experiences to improve brand trust
QoS enables carriers to ensure user experiences by preferentially allocating bandwidths
for applications, such as VoIP and media streaming, that are sensitive to network latency
and jitters.
 Avoiding network congestion to improve user satisfaction
QoS enables carriers to control the bandwidth allocated to specific services, such as P2P,
that consume much more bandwidths than other services, avoiding network congestion.

1.3 Service Models

Common QoS service models are as follows:
 Best-Effort service
This service model applies to the scenario where only one type of service is available. In
this case, no guarantee is provided for the bandwidth, latency, jitter, and packet loss rate.
This service model is the most common one on the Internet and applies to the most data
applications, such as FTP, WWW, and email.
 Integrated service (IntServ)
This service model is a comprehensive model that provides E2E QoS assurance using
RSVP. Before sending packets, UEs initiate requests for bandwidths and latencies. Each
involved node allocates resources to the UEs based on the resource usage on the node
and monitors the status of each passing data stream. This service model is difficult to

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deploy and expand. In addition, the storage and processing capabilities of involved
routers in this service model may be exhausted as network traffic surges.
 Differentiated service (DiffServ)
This service model usually applies to the scenario where multiple services are available
and helps distribute packets of various priorities into corresponding service streams. In
this service model, each node involved in packet forwarding implements a per-hop
behavior (PHB) for the packets of each service. This service model is easy to deploy and
expand. However, this service model cannot provide E2E QoS assurance due to the lack
of signaling communication between the network and end system.

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2 Key Technologies

2.1 Anti-Packet Loss and Anti-Jitter Technologies

2.1.1 Packet Loss Compensation
Because of network congestion, buffer overflow, and transmission errors, packet loss is
common in connectionless IP networks. For audio services, packets are transmitted in
temporal order. If certain packets are lost, letting them be lost is better than retransmitting
them. For the most audio compression algorithms in IP telephony technologies, frames are the
minimal unit in network transmission. Continuous packet loss affects the quality of the
decoded voice on the receiving end. The packet loss compensation (PLC) algorithm is used
for the SE2900 to prevent the impact of media packet loss. Based on the dependencies of the
context voice information, the lost frames are regenerated during decoding, which guarantees
the quality of the received voice.
Figure 2-1 shows the PLC mechanism.

Figure 2-1 Mechanism of the PLC algorithm

2.1.2 RFC2198 Redundancy

RFC2198 redundancy enables soft terminals to access the SE2900 or interwork with other
UEs. Specifically, RFC2198 redundant transmission of media packets is implemented
between the SE2900 and soft terminals to compensate for packet loss on the IP network
because of the complex Internet environment and severe packet loss, ensuring call quality on
soft terminals.
Figure 2-2 shows the mechanism of RFC2198 redundant transmission.

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Figure 2-2 Mechanism of RFC2198 redundant transmission

IP access IP core
network network

Packet 4 Packet 3 Packet 2 Packet 1 Packet 4 Packet 3 Packet 2 Packet 1

Frame 3

Frame 4

Frame 2

Frame 3

Frame 1

Frame 2

Frame 1

Frame 3

Frame 2

Frame 1

Frame 1
Packet 4 Packet 3 Packet 1

Frame 1
Frame 3

Frame 4

Frame 2

Frame 3
In RFC2198 redundancy, each packet carries certain frames in the previous packet and certain
new frames. Once a packet is lost, the receiving end still receives the new frames carried in
the packet because those frames are also carried in the subsequent packet.

2.1.3 De-Jitter Buffer

Jitters are inevitable when audio packets pass the IP bearer network. However, audio packets
must be transmitted in a specific sequence because they are sensitive to temporal order. In this
case, a jitter buffer with sound performance is a must. To reduce the impacts caused by jitters
on IP networks, the receiving end can postpone the play of received audio packets by caching
them in the buffer. The delay is determined by the capacity of the jitter buffer. The larger
capacity the jitter buffer has, the more packets the SE2900 receives before playing, which
means that fewer media packets are lost. However, a longer duration for audio packet to be
cached in the jitter buffer causes a longer latency for the E2E call. The size of a jitter buffer is
determined based on requirements. Overflow and redundancy usually cause packet loss,
packet transmission delay, and jitters. The SE2900 is able to adjust the jitter buffer size in real
time based on network situations when audio transcoding is implemented.
Figure 2-3 shows the de-jitter buffer (DJB) mechanism.

Figure 2-3 De-jitter buffer mechanism

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2.2 Bandwidth Control Technologies

2.2.1 Policy Control (Rx)
Policy control (Rx) enables the SE2900 to interwork with the policy and charging rules
function (PCRF) over the Diameter-based Rx interface. The SE2900 extracts the session
information about a UE, such as the signaling address, media address, and media bandwidth,
from SIP messages, and sends such information to the PCRF over the Rx interface. The PCRF
uses the session information to instruct the policy and charging enforcement function (PCEF)
to implement PCC so that the bearer resources are under control and bandwidth is guaranteed,
ensuring service quality and improving user experience.
Figure 2-4 shows the mechanism of policy control (Rx).

Figure 2-4 Policy control (Rx) mechanism

PCRF
Gx

R
x

UE PGW SE2900
(P-CSCF)

2.2.2 Call Admission Control

Call admission control (CAC) is a function which enables the SE2900 to restrict the resources
available for registrations, calls, and subscriptions, based on locally configured CAC policies.
This function enhances the security of the SE2900 and core servers and guarantees QoS for
registration and call services.
Figure 2-5 shows the CAC mechanism.

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Figure 2-5 CAC mechanism

Implements the control over

the resources used by users Implements control over the
on the entire access network. Implements the control resources used by a single
Access over the resources used by core network user.
network A the entire core network.

Core Core
network A network B

Access
network B
SE2900
(A-SBC) SE2900
(I-SBC)
Implements control over the
resources used by a single
user on the access network.

2.2.3 Committed Access Rate

The committed access rate (CAR) function is an important and efficient bandwidth
management function. It is usually configured on a device at the network border and enables
the device to restrict the incoming packet rate, which helps carriers provide guaranteed QoS.
The SE2900 detects the bandwidth resources used by media streams. If the media streams
from a user consume excessive bandwidth resources, the SE2900 discards subsequent media
packets, preventing bandwidth theft.
Figure 2-6 shows the implementation of CAR using token bucket model:

Figure 2-6 CAR mechanism

Tokens

p B - Burst Size
p - Token Arrival Rate
Overflow
Tokens

Packets
Arriving Conform

Exceed

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2.2.4 Voice Activity Detection/Comfort Noise Generation

Voice activity detection (VAD) is a technique used in speech processing in which the presence
or absence of human speech is detected. It makes an important component of IP telephony by
helping retain service quality, reduce the use of voice circuits, and reduce the E2E latency.
Users do not occupy the voice channels all the time when placing calls. According to the
statistics on the traditional telephone services, a party actually occupies not more than 40% of
the conversation period during a call. This is caused by the following reasons: One party is
listening to the other party. Pauses in the talk are inevitable due to thinking and breaks. Pauses
may be caused by hesitation, breathing, and stammer. The first instance has a long pause and
is rare. The third instance has a short pause and is common. The pause and frequency of the
second instance is between the first and the third instances. The basic principle of VAD is to
judge the voice signal energy. When the energy is below a certain threshold, the UMG8900
considers that the party is in the silence state.
There is one issue in VAD applications. In the mute periods, the comfort noise generation
(CNG) technology is used to generate comfortable noise so that the users will not feel
unnatural in conversation or regard call interruption by mistake. Therefore, a comfort noise
generator must be configured at the receiving end. The receiver adopts a mechanism to
regenerate the main features of the background noise. The background noise parameters are
specified by the transmitting end. The noise generation methods must guarantee the
synchronization of the decoder and the encoder. Even if the encoder does not send any bit in a
certain period, the decoder can identify the absence of bit, which ensures the smooth
assembling of voice segments and silence segments when the voice is restored.

2.3 Forwarding Priority Control

2.3.1 Differentiated Services Code Point
The IP network uses the best-effort mechanism to transmit IP packets. Consequently, when
real-time audio data, video data, or control data are transmitted over the IP network, it cannot
ensure the reliable data transmission. When the network is congested, the quality of services
with a high demand for the QoS cannot be guaranteed.
The differentiated services code point (DSCP) enables UEs to apply for services at various
levels by setting the DiffServ field. The first six bits in the DiffServ field define a DSCP value.
Packets with the same DSCP values are in a set called a behavior aggregate (BA). For each
DSCP value, a PHB ID is configured on the router to identify the behavior, such as traffic
monitoring, traffic throttling, or queuing, that meets certain forwarding requirements. When a
packet arrives, the router distributes the packet into a BA and selects a PHB based on the
DSCP value carried in the packet and forwards the packet accordingly.
The SE2900 supports the DiffServ technique and sets different DSCP values for signaling and
media packets so that packets with high DSCP values are preferentially forwarded, ensuring
service quality.
Figure 2-7 shows the DSCP mechanism.

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Figure 2-7 DSCP mechanism

SE2900 IP network
SE2900

QoS The packets marked

processing
with higher priorities
UE are processed and
forwarded first.
QoS high
DSCP 1 1 DSCP
XXX XXX
…
DSCP 2 QoS medium 2 DSCP
YYY YYY

DSCP 3 … 3 DSCP
ZZZ ZZZ
QoS low

2.4 QoS Monitoring

2.4.1 Voice Quality Reporting
Voice quality reporting enables the SE2900 to measure voice quality of calls in real time,
including the packet loss rate, jitter, round-trip delay, number of received/sent RTP packets,
number of bytes of received/sent RTP packets, and mean opinion score (MOS).Voice quality
reporting enables the SE2900 to implement the following functions:
1. Reports QoS statistics to the EMS so that carriers can monitor the network performance
based on the performance measurement, adjust and optimize network, and improve
service quality.
2. Reports QoS statistics to the EMS using user tracing messages for real-time visualization
of the service quality.
3. Reports QoS statistics to the LMT.
4. The SE2900 includes QoS information in INFO requests.
5. The SE2900 includes QoS information in BYE requests.

Voice quality statistics are calculated based on the fields related to RTP and RTCP.
Figure 2-8 shows an example RTP packet:

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Figure 2-8 An example RTP packet

Figure 2-9 shows an example RTCP-SR packet:

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Figure 2-9 An example RTCP packet

The calculation involves the following fields:

 Sequence number:
The sequence number starts from a random number and increments by one when an RTP
packet is sent. The receiving end detects packet loss and assembles packets based on the
sequence number.
 Time stamp

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The time stamp indicates the sampling time of the first bit in an RTP packet. The sampling
time precision must meet the synchronization requirement to facilitate the synchronization
and jitter calculation. The start value of the time stamp is random. The time stamp value
ascends with the data size.
 Last sender report (LSR):
The LSR is a field in RTCP packets and contains the 32 bits in the middle of the 64-bit
NTP timestamp reported by the latest sender of the RTCP packet from SSRC_n. If no SR
is received, this field is set to 0.
 Delay since last sender report (DLSR):
The DLSR refers to the duration between the receiving and sending of the SR from
SSRC_n. If no SR is received from SSRC_n, this field is set to 0.
 Packet loss calculation method
The packet loss rate is the total number of packets received in a call within a specific
duration divided by the difference between the sequence number of the last received
packet and that of the first received packet within the duration.
 Jitter calculation method
Assume that Si is the RTP time stamp of packet i and Ri is the arrival time of packet i in
the RTP time stamp, for packet i and packet j, D(i,j) = (Rj - Sj) - (Ri - Si). Use the
difference (D) between packet i and its preceding packet i-1 (calculated in arrival order,
not in sequence number order) to consecutively calculate the arrival time jitter after each
packet i is received from the source SSRC_n. The calculation formula is J = J +
(|D(i-1,i)| - J)/16.
 Round-trip delay calculation method
Assume that SSRC_r is the receiver of an SR message. Time point A when the SSRC_n
receives the SR message from SSRC_r can be used to calculate the round-trip delay.
Specifically, use the LSR to calculate the duration of the round trip (A-LSR), and then
subtract DLSR from A-LSR to obtain the round-trip delay.

1.1 Multimedia Priority Service (MPS)

• During the registration of a user, the P-CSCF obtains priority information of the user
from the S-CSCF.
• After receiving the INVITE request from the user, the P-CSCF delivers the obtained
priority information of the user to the PCRF so that the PCRF ensures the quality of
services available for the user based on the priority information.
• The P-CSCF includes the Resource-Priority header in the INVITE request before
forwarding the request to the core network so that the core server can identify the
priority of the user.
• The P-CSCF preferentially ensures the quality of services available for high-priority
users.

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1) Store the MPS of

subscribers
2) Get MPS from HSS and
synchronize to other NEs I/S-CSCF 1 HSS

3
Resource-Priority
3) Indicate MPS to PCRF
and IMS-Core during call SE2900
P-CSCF/ATCF/ATGW

3 1) Implement ARP policy

PCRF

4
4) Map MPS to ARP S/P-GW MME

eNodeB

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