AG - BT01 - E1 Overview of Soft Switch V1.0 60p

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AG_BT01_E1
Overview of Soft Switch

Version 1.00

Chapter
Data-communication
Basics
Key points
Historical background of NGN.
Basic concepts of IP network.
Usual IP voice coding and compression methods
Background story of NGN

As the world is now becoming a small village, people, who
locate in different places in the earth, can communicate with
each other in no time with telephone, email or video etc. The
credit of such advantage goes to the development of modern
telecommunication technology. Going through the history of the
telecommunication will help us to understand the reason and
underlying philosophy of all sorts of current techs, and give us
the insight of what will happen tomorrow.
PSTN network evolution history

As known to all, Dr. Alexander Graham Bell is the father of
telephone, born in 1847, Scotland, 28 year-old Bell braced his
invention device with tears while confirming that his device is
able to transmit voice. At that night, he wrote to his mother and
announced that the day friends staying at their home can still
talk to each other is coming! During the following long period,
telephone business has been supported merely by manualswitching,
the
experiencing
maching-switching
and
programmed-switching. As shown in Figure 1.
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Birth of Telephone
FIGURE 1 TELEPHONE HISTORY (SWITCHING HISTORY)
The first automatic dialing system was patented in 1891 by a

Kansas City undertaker who believed that crooked operators
were sending his business elsewhere, with his main objective
being to eliminate the operators. The first coin telephone was
installed in Hartford, Connecticut in 1900. Party lines were soon
developed to lower the cost of the telephone for individual
families, especially those in rural locations In 1920s, nearly 15%
of the telephone in the world is automatic dialing phone, with
the development of automatic dialing technology, almost 80% of
the telephone in the world is automatic phone by the end of
1950s.
From analog to digit

There are two principal means by which telephone signals are
transmitted: analog or digital radio signals. As the electronic
current that transmit the voice is actually imitating the voice
behavior, it has the various shape.; while the telegraph transmits
only yes or no with definite meanings, which belongs to
data-communication. In the early days of telephone history, the
switching system is controlled by analog switches. As technology
had advanced enough by the late 1970s where 100% digital
systems were being developed for use in the telephone industry.
Digital systems "sample" the analog signal and handle telephone
calls internally as binary digits, then convert them back to
analog to be compatible with regular telephones. This again
made the switch smaller and more reliable. It also allowed new
technologies to be added faster by using modular techniques
(adding systems to the original system without complete
redesign/reinstallation) and by using advanced computer
software.
Chapter 1 Data-communication Basics
Emergence of time-switching
While traditionally, telephone switches connected physical
circuits (e.g., wire pairs), modern telephone switches use a
combination of space- and time-division switching. In other
words, each voice channel is represented by a time slot (say 1 or
2) on a physical wire pair (A or B). In order to connect two voice
channels (say A1 and B2) together, the telephone switch
interchanges the information between A1 and B2. It switches
both the time slot and physical connection. To do this, it
exchanges data between the time slots and connections 8000
times per second, under control of digital logic that cycles
through electronic lists of the current connections. Using both
types of switching makes a modern switch far smaller than
either a space or time switch could be by itself.
The structure of a switch is an odd number of layers of smaller,
simpler subswitches, interconnected by a web of wires that goes
from each subswitch, to a set of the next layer of subswitches.
In most designs, a physical (space) switching layer will alternate
with a time switching layer. The layers are symmetric, because
every call is symmetric (there's a connection in both directions).
A space-division subswitch uses digital multiplexers controlled by
a cyclic memory. This takes physical space for the wiring.
A time-division subswitch reads a complete cycle of time slots
into a memory, and then writes it out in a different order, also
under control of a cyclic computer memory. This causes some
delay in the signal.
Development of VOIP
Telephone systems have been slowly adopting the "IP" (Internet
Protocol) technology over the last decades. Early experiments
proved that voice can be converted to digital "packets" and sent
over the Internet. The packets would be collected and converted
back to analog voice. The quality of the calls was not great but it
showed that it could be done. The major problem was something
called "packet loss" which is common with TCP/IP connections.
By the early 2000s, the IP telephony (or "VoIP" - Voice over
Internet Protocol) technology had improved. Using "classes" of
service, reliable connections could be obtained and packet loss
reduced to minimum levels. Business systems started using VoIP
technology in their PBX (Private Branch Exchange) switches. The
telephones themselves were almost like small computers that
had their own analog/digital conversion systems and TCP/IP
networking technology all the same system. The phone could
"piggy-back" on their existing computer network system. Hence
having voice AND data traffic over the same wires! As we see
the history of VOIP, it can be divided into 3 phases:
AG_BT01_E1 Overview of Soft Switch
Phase 1 of the VoIP project: early 1998 to 1

October 1999
This phase was to get the VoIP technology working and to test it
in a large pilot between PABXs. During this phase Cisco provided
significant support which was to the mutual benefit of Cisco,
CSIRO and AARNet. This phase was completed on 1 October
1999.
Key successes of this phase:
VoIP from PABX to PABX became fully functional, as indicated by
the testing and major pilot implementations.
The VoIP technology acted as a fully transparent ISDN or Q.Sig
network to the PABXs. The Q.Sig included all the advanced PABX
features including Call Back, Centralised Operator, Message
Waiting indicator, Calling Name Display as well as Calling
Number Display.
During the initial setup of a voice call the originating
triggered to route the call to the Public Switched
Network (PSTN) if a problem was encountered in
network while setting up the call, without either of
being aware.
PABX was
Telephone
the data
the users
A light weight Network Management program was written to

stop VoIP during outages, usually on microwave links.
The voice quality was near toll quality.
Facsimile traffic was supported up to 14,400 bits per second, i.e.
up to ITU G.3.
A Billing System was written that invoiced the caller and
generated MIS reports.
CSIRO had seven Gateways that connected PABXs into the
AARNet Voice Service. All Long Distance telephone traffic from
CSIRO sites to the major capital cities (not Hobart or Darwin)
was sent via AARNet. Calls to the local call area hopped out of
AARNet into a CSIRO PABX, which then completed the call by
making a Local call into the PSTN.
CSIRO was averaging 3,500 calls per working day with an
average call duration of 237 seconds per call, with a 90% saving
on call charges (60% of CSIRO calls were between CSIRO sites
and thus did not include the Local call charges). Other AARNet
Members were likely to achieve savings of 70% to 80%.
Gatekeepers worked and vastly simplified routing tables.
Key caveats of Phase 1:
Modem traffic WAS NOT SUPPORTED, and was not likely to be in
the future.
Past generation Video Conference equipment that relied on clear
64kbps ISDN channels WAS NOT SUPPORTED. However the
H.323 VoIP technology, which was initially designed to support

video, would enable advanced video technology using IP.
Phase 2 of the VoIP Project: 1 October 1999 to

2001
This phase was to roll the technology out to all the AARNet
Members and to install Gateways where it was cost effective to
do so, that is, there was enough traffic to the local area to
warrant the installation of a Gateway. This phase had several
parallel components:
Installed Gateways to allow Member telephone traffic to hop out
of the AARNet Voice Service at key areas. Initially they were in
Canberra, Sydney, Brisbane, Perth, Adelaide and Melbourne.
Selected a Carrier to carry the Local call from the above
Gateways into the Public Switch Telephone Network (Last one
installed October 2000). Public information about the Tender is
available.
Wrote and deployed a Network Monitoring Application AARNet
call VoIPMonitor that was scalable to a network the size of
AARNet and stopped VoIP over links that were failing.
Assisted AARNet Members to connect to the AARNet Voice
Service.
A loose status of the installation of VoIP at each member
institution was maintained.
This phase of the roject suffered a massive delay while
negotiating a contract with the Carrier for the ISDN connections
to connect the AARNet Gateway back into the PSTN (several
months). There was also an average 124 day delay waiting for
installation (some 3.5 months late).
Phase 3 On-going operation

This phase is the ongoing operational component of the project
and includes:
support; and dimensioning the network to carry the traffic.
Projects that may begin where this project ends are:
Video over IP;
User directory services for global roaming; and
User Authentication and Authorization for roaming Voice and
Video over IP.
March towards NGN

With the sustainable growth of data services, the current
network framework is evolving to Data Networks. Therefore,
both the traditional telecom carriers relying mainly on voice
services and the emerging Data Network carriers pay much
attention to the new-generation Telecom Network that converges
voice, video, data and etc. and integrates the PSTN circuit
switching with packet switching technology.
The network integration-oriented Soft switch architecture is a
total solution of the new-generation multimedia services. It
implements the interconnection of various service networks
including PSTN/ISDN, PLMN, IN and Internet etc. By optimizing
the network architecture, it realizes network integration, and
more important, the service integration, thus enabling the
inheritance of rich service functions of the traditional Circuit
Switching Network, meanwhile providing new services that are
hard to be implemented in the traditional networks all over the
new-generation network.
Basic concepts of datacommunication

OSI Model
The Open Systems Interconnection Reference Model (OSI Model
or OSI Reference Model for short) is a layered abstract
description for communications and computer network protocol
design, developed as part of the Open Systems Interconnect
initiative. It is also called the OSI seven layers model.
The OSI model divides the functions of a protocol into a series of
layers. Each layer has the property that it only uses the
functions of the layer below, and only exports functionality to the
layer above. A system that implements protocol behavior
consisting of a series of these layers is known as a 'protocol
stack' or 'stack'. Protocol stacks can be implemented either in
hardware or software, or a mixture of both. Typically, only the
lower layers are implemented in hardware, with the higher
layers being implemented in software
The physical layer defines all electrical and physical
specifications for devices. This includes the layout of pins,
voltages, and cable specifications. Hubs and repeaters are
physical-layer devices.
The data link layer provides the functional and procedural means
to transfer data between network entities and to detect and
possibly correct errors that may occur in the Physical layer. The
addressing scheme is physical which means that the addresses
(MAC address) are hard-coded into the network cards at the
time of manufacture
The network layer provides the functional and procedural means
of transferring variable length data sequences from a source to a
destination via one or more networks while maintaining the
quality of service requested by the Transport layer. The Network
layer
performs
network
routing,
flow
control,
segmentation/desegmentation, and error control functions. The
router operates at this layer -- sending data throughout the
extended network and making the Internet possible, although
there are layer 3 (or IP) switches. This is a logical addressing
scheme - values are chosen by the network engineer. The
addressing scheme is hierarchical.
The transport layer provides transparent transfer of data
between end users, thus relieving the upper layers from any
concern with providing reliable and cost-effective data transfer.
The transport layer controls the reliability of a given link. Some
protocols are stateful and connection oriented. This means that
the transport layer can keep track of the packets and retransmit
those that fail. The best known example of a layer 4 protocol is
TCP.
The session layer provides the mechanism for managing the
dialogue between end-user application processes. It provides for
either duplex or half-duplex operation and establishes
checkpointing,
adjournment,
termination,
and
restart
procedures. This layer is responsible for setting up and tearing
down TCP/IP sessions.
The presentation layer relieves the Application layer of concern
regarding syntactical differences in data representation within
the end-user systems. MIME encoding, encryption and similar
manipulation of the presentation of data is done at this layer. An
example of a presentation service would be the conversion of an
EBCDIC-coded text file to an ASCII-coded file.
The application layer interfaces directly to and performs common
application services for the application processes. The common
application services provide semantic conversion between
associated application processes. Examples of common
application services include the virtual file, virtual terminal (for
example, Telnet), and "Job transfer and Manipulation protocol".
FIGURE 2 OSI MODEL
FIGURE 3 OSI MODEL AND ITS PROTOCOLS
TCP/IP protocol suite

TCP/IP was initially designed to meet the data communication
needs of the U.S. Department of Defence (DOD). In the late
1960s the Advanced Research Projects Agency (ARPA, now
called DARPA) of the U.S. Department of Defence began a
partnership with U.S.universities and the corporate research
community to design open, standard protocols and build multivendor networks.
Together, the participants planned ARPANET, the first packet

switching network. The first experimental four-node version of
ARPANET went into operation in 1969. These four nodes at three
different sites were connected together via 56 kbit/s circuits,
using the Network Control Protocol (NCP). The experiment was a
success, and the trial network ultimately evolved into a useful
operational network, the "ARPA Internet". In 1974, the design
for a new set of core protocols, for the ARPANET, wasproposed in
a paper by Vinton G. Cerf and Robert E. Kahn. The official name
for the set of protocols was TCP/IP Internet Protocol Suite,
commonly referred to as TCP/IP, which is taken from the names
of the network layer protocol (Internet protocol [IP]) and one of
the transport layer protocols (Transmission Control Protocol
[TCP]).
TCP/IP is a set of network standards that specify the details of
how computers communicate, as well as a set of conventions for
interconnecting networks and routing traffic. The initial
specification went through four early versions, culminating in
version 4 in 1979.
FIGURE 4 TCP/IP PROTOCOL
IPv4 Address Introduction

For a node in the network, the Internet protocol address (IP
address for short) is a logic address. It is independent from any
network hardware and network configuration. It has the same
format no matter the type of the physical network. An IP
address is a 4-byte number, which is actually composed of two
parts: the first part is the IP network No. while the second part
is the host No. Generally, such a 4-byte IP address is separated
with small dots, in which each byte is indicated with a decimal
number. For example: for 130.130.71.1, the network No. is
130.130 and the host No. is 71.1.
FIGURE 5 IP ADDRESS EXAMPLE
IP addresses can be divided into five classes, i.e., Class A, Class

B, Class C, Class D and Class E. Indicates with binary codes, the
highest bit of Class A addresses is 0; the highest two bits of
Class B addresses are 10; the highest three bits of Class C
addresses are 110; the highest four bits of Class D addresses
are 1110; and the highest five bits of Class E addresses are
11110. Since Class D addresses are only used in special
definition of the host group and Class E addresses are reserved
for future use, only one type in Class A, B and C addresses can
be allocated for a specific network.
F I G U R E 6 I P A D D R E S S I L L U S T R AT I O N
IP
32 bits
Dotted
Decimal
Network
255
Maximum
1
255
255
8 9
16 17
255
24 25
32
Example
172
16
Decimal
Example 10101100 00010000
Binary
122
128
64
32
16
8
4
2
1
128
64
32
16
8
4
2
1
128
64
32
16
8
4
2
1
11111111 11111111 11111111 11111111

128
64
32
16
8
4
2
1
Binary
Host
204
01111010 11001100
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The IP addressing mechanism allows three types of possible

network configuration. IP addresses are the basis for inter-node
communication that uses the IP protocol. This is true for either
the private TCP/IP network or public Internet.
If a user does not hope to add a network to the public Internet,
the user can select IP addresses by forced stipulation. If this
method is adopted, the IP addresses for all nodes on this
network should meet the following stipulations:
1. The network No. part of each IP address is the same;
10
2. The IP address of each node on the network should be sole.

The highest bit 0 of a Class A address and its subsequent 7 bits
belong to the network No. part; while the remaining 24 bits
indicate the intra-net host No. In this case, there may be 126
Class A networks in an interconnected network (network No.
ranging between 1~126. 0 and 127 are reserved). While in a
Class A network, there may be 16,000,000 nodes. Therefore,
Class A addresses are only used in very large regional networks,
e.g., MLNET in the U.S. and some large-scale commercial
networks.
The highest two bits 10 of a Class B address and the subsequent
14 bits belong to the network No. part; while the remaining 16
bits indicate the intra-net host No. In this case, there may be
about 16,000 Class B networks. While in a Class B network,
there may be more than 65,000 nodes. Class B addresses are
generally used in networks constructed by large institutions and
companies.
The highest three bits 110 of a Class C address and the
subsequent 21 bits belong to the network No. part; while the
remaining 8 bits indicate the intra-net host no. In this case, an
interconnected network is allowed to include 2,000,000 Class C
networks. In each Class network, there can be a maximum of
254 nodes. Class C addresses are used in small-scale institutions
and companies.
If you do not like to use the binary system, you can also divide
the three types of networks according to the decimal numbers of
the first bytes of IP addresses. Class A address range between
1~126; Class B addresses range between 128~191; and Class C
addresses range between 192~223.
As in the convention, when the binary codes of the entire
network No. part (the first byte for Class A; the first two bytes of
Class B; and the first third bytes for Class C) are all zeros, the
network No. is considered as the local network; when the binary
codes of the host No. part (the last three bytes for Class A; the
last two bytes for Class B; and the last byte of Class C) are all
ones, the host No. is considered as the broadcast address within
the local network.
11
FIGURE 7 IP ADDRESS CLASSES
IP
8 bits
8 bits
8 bits
8 bits
Host
Host
Host
Host
Host
Host
Host
Host
Host
Host
Class A:
Network
Network
Class B:
Network
Network Network
Network
Class C:
Network
Network Network
Network Network
Network
Class D:
Host
Host
Multicast
Class E:
Research
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Network category
Local Area Networks
Local Area Networks (LANs) accommodate users within a
building or on a campus. Ethernet is the leading network
technology for LANs or networks contained in buildings or on a
single site. Ethernet was first implemented as a shared network
using hubs, but high usage caused congestion that could
dramatically slow or even shut down the network. Subsequently
Ethernet was implemented using a switched architecture scaling
performance of the network. The Ethernet ports operated in full
duplex, gradually the speed increased form 10 Mbps to 10 Gbps,
packet processing at higher layers and routing were added.
Wide-Area Networks and Remote Access

Wide-Area Networks (WANs) connect LANs in campuses across
cities, states, and countries. Remote access refers to the
connection of an individual user or very small branch office to
the central network. A single user can dial up the corporate
network directly or gain access via the Internet. Internet Service
Providers (ISPs) aggregate multiple users through a router to
the campus. In general, LAN speeds are greater than WAN and
remote-access speeds. WAN technologies:
1. Dedicated and Dial-Up Access
2. Integrated Services Digital Network (ISDN)
3. Asynchronous Digital Subscriber Line (ADSL)
4. Cable Access
5. Wireless Access
12
Metropolitan Area Network

Metropolitan Area Networks (MANs) channel traffic within the
metro-domain (i.e., inter-business, inter-office, metropolitan
connectivity) to and from large long-haul POPS (points of
presence). With the successful deployment of many long-haul
fiber networks, the focus has shifted towards MANs. The goal is
to bring the cost benefits and networks efficiencies of the optical
networking to end-users wanting to link campuses with multigigabit LANs.
F I G U R E 8 L AN , M AN A N D W AN
LAN
MAN
WAN
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Basic concepts of IP
telephone
The transmission of realtime voices via the IP network is
different from that of ordinary data. In the former case, the
relevant application devices must meet the realtimeness of
voices. The voice packet transmission requires the network to
provide sufficient bandwidth in time. Therefore, for most of the
current IP networks that do not provide so high rates, the voice
compression technology is the key for implementing IP voice
communication. Now, we will present a brief description of the
frequently used voice coding and compression modes at present:
1. PCM
Pulse code modulation (PCM) is the earliest
technology, which does not include any
algorithm. It transmits voice signals with
bandwidth, i.e., taking 8,000 samples per
digital voice
compression
the 64kbps
second and
13
acquiring an 8-digit voice signal per sample. PCM is the

standard coding mode adopted in G.711.
2. CELP
Code excited linear prediction (CELP) is the most advanced
voice transmission technology at present. The CELP
algorithm is to compare analog signal samples with curves in
the predefined code book; send codes in the code book
closest to these analog signal samples to the receiving end;
and regenerate original signals after comparison again with
the code book at the receiving end. The sampling interval of
original signals is very short. Therefore, the regenerated
signals are very close to the original signals after being
filtered. CELP is the basis of numerous advanced patented
voice compression modes. Voices can be compressed to
5.3kbps, 8kbps or 9kbps.
3. CS-ACELP
Conjugate structure algebra code excited linear prediction
(CS-ACELP) or G.729 is the 8kbps voice compression and
coding standards of International Telecommunications Union
(ITU). CS-ACELP is a new algorithm, which is able to encode
8kbps voice signal bit streams (while the rate of ordinary
PCM signals is 64kbps). The bandwidth efficiency is eight
times as that of PCM and four times as that of 32kbps
ADPCM. At present, CS-ACELP is the most welcome voice
encoding/decoding plan.
When actually selecting a voice compression algorithm, it is
necessary to take various factors into consideration. For
example: the pursue of higher bit rates guarantees sound voice
quality but requires to occupy more system resources. While
lower bit rates will influence voice quality and increase delay.
Therefore, to keep better voice quality in the precondition of
lower bit rates is the principle for compression algorithm
selection.
Given the operational factors that can affect voice quality, how
can end-to-end voice quality be assessed? There are several
standard methods to measure voice quality. One obvious
method, the Mean Opinion Score (MOS or ITU-T P.800) is
obtained by conducting a controlled test in which a large number
of human listeners make calls to each other and rate the voice
quality. Evaluations are derived directly from the individuals who
will experience the voice call. MOS is the mean score calculated
from this large volume of human opinions (1-5, with 5 being
excellent and 4 considered toll quality). Disadvantages of MOS
are that it's time-consuming, costly, and inconvenient.
Therefore, voice quality measurement is usually done using one
of the following analysis algorithms: Perceptual Speech Quality
Measure (PSQM, ITU-T P.861, obsolete), Perceptual Analysis
Measurement System (PAMS), or Perceptual Evaluation of
Speech (PESQ, ITU-T P.862). These measurements provide the
metrics necessary to objectively assess and quantify voice
clarity. Most voice-quality test tools provide one or more of these
14
measurements and corresponding

MOS estimation
for
comparison purposes. Though these standards do not reflect the
quality of endpoints, they can be used for quick network
assessment. However, even the newest and preferred of these
algorithms, PESQ, does not provided a comprehensive
evaluation of transmission quality. PESQ measures the effects of
one-way speech distortion and noise only, not the effects of twoway interaction such as loudness, delay, and echo. A network
can have a high PESQ score and yet have poor connection
quality. Most importantly, these measurement methods do not
provide enough analytical information about where in the
network problems might occur.
Network infrastructure components (PSTN trunks or lines, voice
gateways, analog and IP phones, etc.) affect voice quality by
creating a loss insertion. Therefore, in the network planning or
revising stage a comprehensive voice-quality analysis--a
transmission loss and level plan--needs to be performed for the
entire network. The design needs to be optimized to the desired
level of voice quality between any two users of the private
network as well as between the network and other--usually,
public--networks.
Homework
1. Please briefly discuss the telephone history
2. What are the common decoding/encoding algorithm in
telephone history, describe the differences
15

Chapter
NGN Architecture
Key points
The next generation

softswitch technology.
network
ZTE softswitch product series.
NGN network protocol family.
architecture
based
on
Why NGN
At present, two totally independent networks exist: the PSTN
network and data network, which provide the voice service and
basic data service respectively.
Network separation and isolation of operation & maintenance
have been keeping the general network operation &
maintenance costs on a high level, and furthermore, a network
cannot provide complicated convergence services, although the
network convergence has been an inevitable trend.
Since a traditional voice network is a closed network with
monopolized resources, it has become a common understanding
in the telecom industry that the packet network (typically, the
Internet), with the advantages such as open architecture, low
costs and large scale, will replace the PSTN to become the basic
frame of the next generation of convergence networks and that
the construction of the next generation of networks will be based
on current packet networks.
It is necessary for carriers to consider resource utilization and
investment protection during construction of future networks. On
one hand, carriers should trace the latest technologies; and on
the other hand, they should try to utilize existing technologies
and resources. Thus, carriers can provide users with large
numbers of services economically and rapidly to make the
17
highest profits,
alteration.
without
the
need
of
large-scale
network
The solution of smooth transition from existing networks to the

next generation networks is the key to the problem. The
Softswitch solution based on softswitch technology is just a
mainstream solution to smooth network evolution
ZTE NGN Softsiwtch

Architecuture
Hierarchical models are adopted for the Softswitch-based next
generation network. The entire network can be divided into four
layers: Service Layer, Control Layer, Core (Transport) Layer and
Edge Access Layer, as shown in Figure 9.
F I G U R E 9 N E X T - G E N E R AT I O N N E T W O R K B A S E D O N T H E S O F T S W I T C H
TECHNOLOGY
Service layer
Database
SCP
AAA Server
Application Server
ZXSS10 SS
Softswitch
Control layer
Policy Server
ZXSS10 SS
Softswitch
IP Router/ATM Swtich
Core Packet Network
Core layer
Edge layer
SG
TG
AG
NAS
H.323
GW
IP
PBX
M SAG
Broadband Access
No.7 Network
SG: signalling gateway
PSTN / ISDN
TG: trunk gateway
NAS: narrow-band access server

wireless access gateway
IAD
WAG
Wireless
AG: access gateway
IAD: integrated access device
WAG:
H.323GW: H.323 gateway IP PBX: IP-based private branch exchange

MSAG: multi-service access gateway
18
Chapter 2 NGN Architecture
1. The edge access layer refers to various access gateways and

new types of access terminal devices related to the current
network. It implements interworking with the existing
various types of communication networks and provides
access of various communication terminals (e.g., the analog
phone, SIP Phone, PC Phone visual terminal and intelligent
terminal) to the IP core layer.
2. The Core (Transport) Layer refers to a packet switching
network composed of backbone transmission equipment such
as IP router or broadband ATM switch, which is the bearer
basis of a softswitch network.
3. The Control Layer refers to Softswitch control units, which
completes integrated control processing functions such as
call processing control, access protocol adaptation,
interconnection and interworking and provides an application
support platform for the entire network.
4. The Application Layer provides a network with various
applications
and
services,
client-oriented
integrated
intelligent services and service customization.
Where, standard interfaces are used for communication between
layers. Under the control of the core equipment (i.e., the
Softswitch control equipment) and based on division of labor and
cooperation of work, the related NE equipment implements
various service functions of the system.
In softswitch architecture, the softswitch control equipment is
the core, which is independent of the bottom-layer bearer
protocols and implements functions such as call control media
gateway access control, resource allocation, protocol processing,
routing, authentication and accounting. The softswitch control
equipment can provide all basic call services, supplementary
services and point-to-point multimedia services a PSTN can
provide. Furthermore, with the cooperation of the Service Layer
equipment (SCP) and Application Server, the equipment also can
provide users with traditional intelligent services, value-added IP
services, diverse third-party value added services and new
intelligent services.
ZTE Softswtich
ZXSS10 SS1a is a piece of softswitch control equipment with
medium capacity, which can process hundreds of thousands of
calls. ZXSS10 SS1b is a piece of softswitch control equipment
with large capacity, which can process millions of calls.
19
ZXSS10 SS1A/B: softswitch core

control device
The ZXSS10 SS1a shelf is 19-inch wide and 4U high. The boards
in the shelf are mounted in the front / rear face-to-face mode.
Therefore, the interface boards with cables can be mounted from
the back, preventing cabling in the front. The integrated shelf is
composed of a standard and integrated shelf, a PCB and a power
box. It provides six slots for horizontally-mounted boards. The
backplane which is 3.2 mm deep can accommodate twelve
boards mounted in the front / rear face-to-face mode. Figure
10 is the front view of the ZXSS10 SS1a shelf.
FIGURE 10 FRONT VIEW OF THE ZXSS10 SS1A (4U)
Figure 11 is the rear view of the ZXSS10 SS1a shelf.

FIGURE 11 REAR VIEW OF THE ZXSS10 SS1A (4U)
Its dimension is 482.6 mm 175 mm 466 mm (W H D). Its

weight is 21.5 kg
The ZXSS10 SS1b shelf is 19-inch wide and 12U high. The entire
shelf is composed of a standard and integrated shelf, a PCB, two
power shelves, a power distribution box and a fan box. It has 17
slots. The backplane which is 4.8 mm deep can accommodate 27
boards mounted in the front / rear face-to-face mode.
Figure 12 shows the front view of the ZXSS10 SS1b shelf.
20
FIGURE 12 FRONT VIEW OF THE ZXSS10 SS1B SHELF (12U)
Figure 13 shows the rear view of the ZXSS10 SS1b shelf

FIGURE 13 REAR VIEW OF THE ZXSS10 SS1B SHELF (12U)
Its dimensions are 482.6 mm 531 mm 545 mm (W H D).

Its weight is 56 kg
ZXMSG 7200: integrated media

gateway
In addition to the channel associated signaling and DSS1
signaling processing functions, the ZXMSG 7200 provides the
trunk media gateway, access media gateway, Media-Server and
21
SS7 gateway functions. These functions can be applied

separately on the ZXMSG 7200 or combined with other functions
as an integrated media gateway. The ZXMSG 7200 provides the
hairpin function between TDM interfaces, and the equipment can
be used as the PSTN/PLMN tandem or toll office.
The ZXMSG 7200 No.7 signaling system is developed strictly in
accordance
with
the
relevant
standard
and
ITU-T
recommendation, fully implementing the SS7 MTP2, MTP3 and
SCCP and other functions and realizing such protocols as the
SCTP, M3UA and M2PA of the SIGTRAN protocol stack. In
addition, it features the powerful processing capability, high
reliability and complete maintenance function, and can work in
the SG proxy mode or the SG signaling transfer mode. When the
ZXMSG 7200 functions as an SG, the GS works as the built-in
signaling gateway of the media gateway. If the required
signaling processing capability is stronger, it is recommended to
configure the dedicated independent signaling gateway.
ZXMSG 9000: high-capacity

integrated media gateway
The ZXMSG9000 is based on brand new 3G platform. And it has
huge capacity. The rack structure is shown in Figure 14, for
every rack it has 4 shelves, and every shelf has 17 boards.
FIGURE 14 ZXMSG9000 RACK STRUCTURE
ZXMSG 5200: multi-service access

gateway
22
The ZXMSG 5200 is applied in the edge access layer of the NGN
network. It can provide abundant services for various users. For
ordinary users, it can perform many services such as Video On
Demand (VOD), online stock purchasing, online teaching, telemedicine and cell network administration. After commissioning
the IPTV service in later period, it can provide more abundant
video service for users. For corp subscribers such as enterprises,
banks, governments, hotels and schools, it provides more
service types including complex dedicated line service besides
voice and data service access. So the demand of access mode is
also various. For low speed dedicated line service, it can provide
narrowband data dedicated lines such as subrate, 64k, N*64k,
2M by ZXMSG 5200. For broadband dedicated line demand, it
can provide the access of dedicated data service such as
:LAN,ADSL,G SHDSL and VDSL conveniently.
ZXSS10 A200: Access Gateway

A200 (ZXSS10 A200 access gateway) connects analog telephone
subscriber to SOFTSWITCH network and can provide V5
subscriber access and some data services. Combining with
DSLAM, it also can provide subscribers with broadband data
access.
AG can be placed in cells and buildings to replace the end office
in PSTN network. It is flexible to use and convenient to network,
and can be maintained and managed uniformly.
ZXSS10 I500/I600/I700: Integrated

Access Device
IAD series are small-capacity media gateway devices launched
by ZTE recently, with such features as flexible networking, easy
installation and simple deployment. It has many functions, such
as media conversion between the packet network and the PSTN
network. There are three major categories: 500s, 600s and 700s
as shown in Figure 15.
23
F I G U R E 1 5 I AD S E R I E S P R O D U C T S
ZXSS10 B100: Broad-band gateway

The ZXSS10 B100 provides the following functions:
Supporting network address translation (NAT) of H.248 / MGCP /
SIP protocols.
Supporting NAT traversal
Supporting dynamic user registration.
Supporting the backup calling agency.
Providing command line interfaces (based on Telnet and serial
port).
Supporting remote FTP version upgrade
Supporting bandwidth-based strategy control.
Supporting a hidden network topology.
NGN network protocols

ZXSS10 SS1A/B core control device is a protocol entity,
depending on several protocols, to enable the intercommunicating with various networks. It is a high-compatible
solution to deal with the future breakthrough of a new
technology. Once there is a new tech, all it needs is upgrading
the software version and provide associated devices in access
layer, the hardware structure of the system could still remain
intact.
Generally, ZXSS10 SS1A/B supports the standard protocols
include: MGCP, MEGACO/H.248, SIP, ISUP/TUP, SNMP, M3UA,
SCTP, H.323, etc. The protocol system is illustrated in Figure
16.
24
F I G U R E 1 6 N G N P R O T O C O L F AM I LY
Megaco/H.248 protocol
The Megaco/H.248 protocol is used between the SoftSwitch
control equipment and various media gateways. Two
abstractions are defined in the protocol, i.e. Termination and
Context.
A Termination sources and/or sinks one or more streams. In a
multimedia conference, a Termination can be multimedia and
sources or sinks multiple media streams. The media stream
parameters, as well as modem, and bearer parameters are
encapsulated within the Termination.
A Context is an association between a collection of Terminations.
There is a special type of Context, the null Context, which
contains all Terminations that are not associated to any other
Termination. For instance, in a decomposed access gateway, all
idle lines are represented by Terminations in the null Context.
There are eight commands defined in H.248. They are Add,
Modify, Subtract, Move, AuditValue, AuditCapability, Notify and
ServiceChange.
A group of commands between the MG and the SoftSwtich
control equipment implement the transaction interaction. One
event interworking can be identified by a TransactionID.
Transactions consist of one or more Actions. An Action consists
of a series of Commands that are limited to operating within a
single Context. Consequently each Action typically must specify
a ContextID. Figure 17 shows a graphic representation of the
Transaction, Action and Command relationships.
25
F I G U R E 1 7 T R A N S A C T I O N , A C T I O N A N D C O M M A N D R E L AT I O N S H I P S
Transactions guarantee ordered Command processing. It means:

the commands in an event interworking are executed in order.
Ordering of Transactions is NOT guaranteed - transactions may
be executed in any order, or simultaneously.
At the first failure of a Command in a Transaction, the processing
of the remaining Commands in that Transaction will stop.
Applications SHOULD implement an application level timer per
transaction. Expiration of the timer should cause a
retransmission of the request. Receipt of a Reply should cancel
the timer. Receipt of Pending should restart the timer.
SIGTRAN protocol
The SIGRAN protocol stack transmits the SCN Signaling in the IP
network, supports the interface with the standard primary
language of the SCN Signaling application without any
modification, thus to guarantee that the existent SCN Signaling
application can be put into use without modification. Meanwhile,
it utilizes standard IP transmission protocol as the lower layer
transmission Signaling, and satisfies the requirement by SCN
Signaling transmission through adding the function of the
protocol. Figure 18 shows the model of a universal Signaling
transmission function defined by IETF.
26
FIGURE 18 SIGTRAN S IGNALING TRANSMISSION FUNCTION MODEL
IP SCP/IP HLR
Sigtran
SCN
signaling
Sigt ran
SCN
signaling
Sigt ran
Signaling gateway (SG)
Signaling gateway (SG)
Sigtran
Sigt ran
M edia gateway
controller (M GC)
Media gateway
controller (MGC)
M edia
stream
M GC
P
M GCP
Media gateway
(MG)
RTP stream
Media gateway
(MG)
M edia
stream
In the model, the Signaling gateway terminates the Signaling in

the circuit switching network, and then transmits the Signaling
content to the Media Gateway Controller (MGC) for processing
through SIGTRAN. MG terminates the inter-office trunk and
controls the trunk according to the control instruction from MGC.
SIGTRAN protocol
adaptation layer,
transmission layer.
is composed of three parts:

Signaling transmission layer
Signaling
and IP
The Signaling adapting layer provides the standard primary

language interface with SCN Signaling, the Signaling
transmission layer provides the real-time reliable transmission
service required by SCN Signaling, and the bottom layer is the
standard IP protocol layer. Figure 19 shows the model of
SIGTRAN protocol.
F I G U R E 1 9 S I G T R AN P R O T O C O L S TAC K M O D E L
TCAP
Q931/QSIG
MTP3
IU
A
M2UA/M2PA
SCCP
ISUP
M3UA
SCTP
IP
TUP
TCAP
Signaling
application layer
SUA
Signaling
adaption layer
Signaling
transmission laye r
IP protocol layer
In the following Sections 3.2, 3.3, 3.4, 3.5 and 3.6, descriptions
of M3UA, M2UA, M2PA, SCTP and SUA protocols will be given
respectively.
27
M3UA protocol
Being the third level adaptation protocol of MTP, the Signaling
adaptation layer protocol of SIGTRAN protocol stack M3UA is
used to support the transmission of the MTP-3 user message in
IP network, including ISUP, SCCP and TUP messages. TCAP or
RANAP message is transmitted transparently by M3UA as SCCP
payload. The M2UA protocol employs SCTP for the reliable
Signaling transmission at the low transmission layer.
Protocol Overview
1. Terms:
AS (Application Server): It is a logical concept. Each AS is an
endpoint that implements some service function in the
network. An AS always serves a specified routing key.
ASP (Application Server Process): It is a process instance in
AS. More than one ASP can serve one AS in the
active/standby or load-sharing mode. One ASP can also
serve more than one AS.
SCTP connection: It provides the service of transmitting
MTP3 user Signaling protocol data and the message between
peer layers of M3UA.
Routing key: Each routing key in the gateway defines a
collection of routing characteristics of SS7 Signaling
message. The SS7 Signaling message with the required
characteristics is processed by a designated AS.
Routing Context: These are parameter values of ASPM
message between gateway and ASP. The parameter uniquely
identifies the SS7 Signaling message related to a specific AS,
that is to say, the message keeps a one-to-one relationship
with the routing key or AS.
2. Signaling Transmission Model
The SS7 message is sent from SCN to the IP network via the
Signaling gateway by using M3UA protocol. The transmission
mode of Signaling protocol is shown in Figure 20.
28
F I G U R E 2 0 S C H E M AT I C D I A G R A M O F I S U P M E S S A G E T R A N S M I T T E D T H R O U G H
SG
SEP/STP
SS7
IP
SG
NIF
ISUP, etc.
ASP
ISUP, etc.
MTP3
M TP3
M 3UA
M 3UA
MTP2
M TP2
SCTP
SCTP
L1
L1
IP
IP
In SG, the MTP-TRANSFER indication primitive received from

the MTP3 upper layer interface is transmitted to the local
M3UA to translate and map the network address and then
routed to the final IP destination. The MTP-TRANSFER
primitive received by the function of local M3UA network
address translation and mapping serves as the MTPTRANSFER Request primitive, and is transmitted to the MTP3
upper layer interface, and further routed to SEP in SCN.
SS7 Signaling gateway may include the functions of the
SCCP protocol layer, may perform SCCP global title
translation (GTT) of the local SCCP address pointed out in
the message. If the GTT result of the SCCP message
generates a peer DPC or DPC/SSN address in the IP domain,
the MTP-TRANSFER Request primitive generated will be sent
to the local M3UA website translation and mapping modules,
and will be routed to the final IP destination. Such Signaling
transmission mode is shown in Figure 21.
F I G U R E 2 1 S C H E M AT I C D I A G R A M O F S C C P S I G N A L I N G T R A N S M I S S I O N
SEP/STP
SS7
IP
SG
SCCP
SCCP User
ASP
SCCP User
SCCP
SCCP
MTP3
MTP3
M3UA
M3UA
MTP2
MTP2
SCTP
SCTP
L1
L1
IP
IP
In respect with MTP3 management message, it is required

that MTP3 user protocol of ASP should be able, like the SEP
node in the SCN network, to receive instructions on the SS7
Signaling point availability, network congestion and user part
29
availability. To accomplish this function, gateway SG should

have a Signaling network management function, capable of
transmitting Signaling network management messages on
the SCN network and IP network. Such message
transmission model is shown in Figure 22.
FIGURE 22 MANAGEMENT MESSAGE TRANSMISSION MODEL
SEP/STP
SG
SS7
IP
ASP
NIF
30
MTP3
MTP3
M3UA
M3UA
MTP2
MTP2
SCTP
SCTP
L1
L1
IP
IP

Chapter
ZTE NGN Solution

Key points
Key problems in softswitch deployment and their solution.
ZTE softswitch network solutions.
New services supported by ZTE softswitch system.
Key problems in softswitch

deployment
Addressing
As a node in package-switching network, softswitch device must
have its own network identifier, in IP network, which is IP
address. However, the identifier normally is limited and can not
be assigned randomly. As the development of NGN network and
user scale becoming bigger and bigger, the number of identifiers
(IPv4 addresses) will not be enough. To solve this problem is to
user private identifier, which rises another problem mapping
from private identifier to public identifier. ZXSS10 BGW device is
designed to solve such problems.
31
F I G U R E 2 3 U S I N G B 1 0 0 T O S O L V E P R I V AT E - P U B L I C A D D R E S S M A P P I N G
Routing
Any node in a network can communicate with softswitch device
directly, when the NGN network becomes larger and larger, IP
network devices grows more and more, it is inappropriate to ask
softswitch device itself maintain the routing information. A
integrated, aggregated, shared routing device becomes
necessary.
To solve the routing problems, when the country NGn network is
under construction , all softswitch routing information should be
contained in routing server, while communicating with other
softswitches, one should first consult the routing information
towards a routing server to query the proper way. While the
network involves minor number of softswitches, the softswitch
itself can maintain the routing information.
OSS implementation
For NGN, the problems like network management, billing, which
are not considered in the internet
business are the most
important required functionality that traditional tele-carriers care
about. OSS components, functionality, and working mode, the
information provided by network element device, interfaces
between OSS and application system, inter-working and
implementation with existing network management system,
billing system, and authentication system are also very
important problems which need to be solved in recent years.
ZTE softswitch EMS system partially solved the above problems
and provided a unified network management system EMS.
32
Chapter 3 ZTE NGN Solution
QoS
From the policies below, it is clear that the ZTE softswitch
system can not only ensure the voice forwarding with highest
priority but also control the bandwidth and traffic of voice and
video services in the network to ensure the QoS of voice and
video services, providing subscribers with carrier-class QoS.
Design and Construct Appropriate Network

Bandwidth to Meet the Requirements of Ensuring
QoS
Appropriate bandwidth is an important means to ensure QoS.
For instance, a comparatively clear IP telephone will seize a
network bandwidth of 12 kbps. As a result, it is impossible for a
64 kbps link to bear 8 such IP telephones simultaneously. In
addition, the delay of data transmission in the entire network will
be decreased in a certain range with the increase of physical line
bandwidth.
A basic design conception is: To select line bandwidth according
to the services that seized large bandwidth; to consider services
multiplexing the bandwidth according to the application
frequency of service. But if the services in the network are all
frequent services, only overlapping bandwidth can be employed.
It is a must to take the actual condition of carriers network
bandwidth resource into consideration to analyze when using the
line in practice.
The tandem layer and backbone layer of the

entire network employ the modes of service
differentiating and priority queuing to ensure the
QoS of voice and video
For the network that VPN can be partitioned, it is recommended
that the services of voice and video should be attached to a VPN,
differentiating from data service, consequently the QoS and
network security of voice and video can be ensured.
If VPN cannot be partitioned, it is recommended to classify the
services and forward packet according to the priority. Configure
the voice and video services with higher priority than data
service to ensure that the services of voice and video be
forwarded prior to data service, as a result, avoid the impact on
voice and video services originated by masses of burst data
service.
33
QoS Warranty Policy for LAN Access

Subscribers.
The LAN Access
Categories:
The CPN
partition.
Subscribers
transmission
Can
equipment
be
does
Classified
not
as
Two
support
Vlan
Assign the voice/video service terminal and Internet service

terminal to different Ethernets, isolated physically. The IP
address partition of voice/video service terminal and Internet
service terminal is in different network segments.
The CPN transmission equipment supports Vlan partition.
Unify the service planning in the CPN: Partition different services
into different VLANs, for instance, unify the voice service
terminal into VLAN1, video service terminal into VLAN2, and
Internet service terminal into VLAN3. Consequently, the switch
can reorder the priority of service forwarding according to VLAN
numbers, forward real-time service with high priority to ensure
QoS.
The QoS Mechanism of ZTE Softswitch System

Equipment
1. The subscriber terminal IAD supports 802.1P two queues
priority.
2. IAD supports 802.1Q VLAN technology and 802.1Q VLAN
TAG, partitions voice from data service at the fringe of
access.
3. All IADs and media gateways of ZTE softswitch support
technologies including echo-offset and delay jittering.
4. The media gateway can acquire the transmission condition of
related voice media stream, and reports to core control
device of softswitch, which control the subscriber calling
access in accordance with the traffic of subscribers in the
media gateway.
5. The broadband gateway can implement NAT between public
and private network and control subscriber access according
to the number of accessed terminal subscribers to control the
traffic of the CPN subscribers.
34
ZTE softswitch network

solution
ZTE Softswitch can be used in many different networking
scenarios.
Class 4 Replacement
Class 4 replacement is more popular among incumbent carriers
as they are not keen in replacing C5 switches in near future. C4
switches however, if at all replaced, are prime choice of carriers
as replacement candidates; in order to reduce complexity and
gain advantage of ease of managing their network.
It is what exactly NGN has to offer from its bag of tricks.
FIGURE 24 CLASS 4 REPLACEMENT SOLUTION USING SOFTSWITCH
ACCESS solution
Then there is another situation where ZTE Softswitch can
accommodate a large number of subscribers via IAD who are
interested in broadband internet access using DSL. IAD is
another unique product in NGN network, which works as kind of
a residential gateway (RGW). As an end result,
Softswitch is accessible to IAD and internal details of DSL
network are transparent to user. The key point to note here is
that DSL is being used as access technology here, and as
expected NGN network perfectly integrates.
35
F I G U R E 2 5 A C C E S S S O L U T I O N C A B L E V I A I AD 6 0 0
Similarly, where cable operators are predominant source of

subscriber reach, ZTE Softswitch again is fully ready and suitable
for deployment. All that is required at customers site is a cable
modem with integrated MGCP support and connected to IAD
600. Consequently, backbone router routes the MGCP commands
and media as well as signalling to appropriate entities, such as
Softswitch. MSG 7200, say, is used to translate media from and
to PSTN, of course under control of Softswitch. This is apparent
in above figure.
Localities where voice requirements are predominant as
compared to data, a solution based on IAD I504 can be ideal.
Each of this tiny equipment unit can provide three telephone
lines! Offers carrier quality voice with fast connection setup.
For larger capacity, I524 is product of choice. This IAD has 24 Z
ports to offer, which ensures that densely populated areas can
be served.
36
F I G U R E 2 6 A C C E S S S O L U T I O N U S I N G I AD 5 2 4
Situations do exist where there is an existing number of large

customer base, a typical example which can be quickly
referenced is an enterprise with a private branch exchange
(PBX). Here too, a customized solution is offered by ZTE the
IAD I704. This IAD is unique and compact and can offer 4 E1s to
be simultaneously connected to a PBX. In addition to supporting
H.248. I7o4 also supports PRI signaling.
FIGURE 27 ACCESS SOLUTION USING I704
37
Co-operating with existing IN

network
Softswitch has powerful service providing ability, not only
exerting in the service variety, but also in the providing speed.
And speed is the most important factor for the carriers to
withhold their competition place in the market.
The evolution of existing IN network towards NGN can be
illustrated as Figure 28: Softswitch together with MG to play as a
virtual
SSP,
connecting
SCP
with
INAP
interface,
communicating with traditional SSP or switch through SS7 or
CAS signal. In the same time, the media server in the datanetwork can replace the circuit media server to provide the addon service.
F I G U R E 2 8 I N H E R I T E X I S T I N G I N A P P L I C AT I O N P L AT F O R M
Co-operating with existing h.323

network
The first generation VOIP network has been launched for years,
and has many subscribers, it is unreasonable to throw away the
old solution to brace the new one. To considering the interworking with the existing H.323 network has been proposed in
the early age of NGN project. The inter-working protocol will use
H.323, while softswitch and H.323 belong to different carriers,
the interworking point locates between softswitch and 1-level
38
GK. While softswitch and H.323 belong to the same carriers,

inter-working point can be assigned by the carrier itself.
In pure IP environment, for existing H.323 network, softswitch
together with its controlling BGW play as a virtual H.323 GW,
softswitch uses two separate physical network interface, one
connects to NGN network, the other connects to H.323. as
shown in Figure 29.
FIGURE 29 INTER-WORKING WITH H.323 NETWORK
Co-operating with other softsiwtch

system
Softswitches are communicated with each other using standard
SIP-T (support BICC in the future). SIP-T is a flexible, inherited
protocol, which has many advantages such as support telephony,
more extensible. For the access layer device(IAD, TG, AG etc),
as long as the media coding format negotiation is correct (G.711
G723, G729), adopting standard RTP package, media can talk to
each other. Different softswitch could communicate with each
other using standard protocols.
39
FIGURE 30 INTER-CONNECTING WITH OTHER NGN NETWORK
Service provision solution

The ZXSS10 SS1a/1b softswitch control equipment has powerful
service capability. The system is designed with the hierarchical
network framework (as shown inFigure 31). The service
provisioning is independent from the network and media. Once
the service is generated, it will be shared by the entire network.
The flexible service logic creation environment enables carriers
to avoid dear hardware upgrade costs and relevant technology
and transportation costs when launching new services. Services
can be provided flexibly by developers, carriers or soft
corporations authorized by carriers and the third party.
Compared with the traditional method, carriers are able to
launch new services more quickly without lower costs
The ZXSS10 SS1a/1b softswitch control equipment supports
three service provisioning modes:
1. The ZXSS10 SS1a/1b softswitch control equipment directly
provides the basic PSTN service, supplementary service and
point-to-point service video service;
2. The equipment acts as the virtual SSP via interworking with
existing SCP to provide the traditional intelligent service.
The softswitch control equipment provides two modes for
interworking with SCP:
i.
Interworking with SCP via SG in the standard No.7 mode;
ii. Direct interworking with SCP via the extended INAP

protocol on the basis of the TCP/IP connection.
40
3. The equipment provides the IP value-added service,

intelligent service and future multimedia service by means of
supporting the application server.
The WEB Server is added into the network, which acts as the
service registration and selection platform for IP subscribers and
the service subscriber window. In the specific application, the
standard service API interface of the application server can be
provided for a third party developer, who can incessantly
develop various new services for subscribers. The application
server can also be leased to enterprise group users, who can
provide customized services for subscribers. Moreover, the
application server can be provided for network carriers as the
value-added service platform.
FIGURE 31 SERVICE PROVISIONING MODE OF SOFTSWITCH NETWORK
New service introduction

WEB 800 service
The Web800 is the Web-based free phone service. It is a new
multimedia service with special billing performance. It features
that the call to a service subscriber is paid by the called instead
of the caller.
The call is initiated from a PC, and the information is input or
selected on Web pages. The service subscriber can manage its
own information on the Web pages. And, the administrator can
manage the service subscribers through the Web pages. As
shown in Figure 32.
Web 800 provide an excellent
advertisement platform for big enterprises suck as hotel, tourist
41
organization, plane company etc. to initiate their online

consultant, online booking service etc and attract huge amount
of online customers to enhance their company charisma.
FIGURE 32 WEB 800 SERVICE
Service properties
1. service utilization
User select its 800 service number and fill in its own
telephone number, accomplish the service by clicking the
webpage button, user can also fill in known 800 service
number.
2. billing
accomplish subscriber billing function, bill goes to the user
account which has been bound to the service, while charge
no money towards the caller.
3. service data management
service user can manage it own account through webpage,
such as changing routing, password etc.
4. subscriber management
5. administrator could change the service user configuration,
including add/remove/delete users etc.
6. ONLY
When service user has multiple telephone numbers, it can
register a single one web 800 number, for those who access
the 800 number users, system will divert the calls to
different destinations according to the service user
requirements.
42
7. busy/no answer divert

one service user can define multiple destination numbers,
system could automatically divert a call from a busy
destination to a free one.
8. call blocking
service user could define the caller privilege, forbidding
subscribers from one certain area to make calls towards
another area.
9. time routing
service user could define routing table based on the time
schedule
10. temporary routing
service user could define routing table tempararily
11. destination selection
service user could select destination according to its own
requirements
12. secure access
service user could define the access password, only those
who enter the correct one are allowed to pass through.
13. management function
service user could define its own property by telephone,
add/remove/delete/query/forbit/permit/active/inactive list of
subscribers.
Web Conference
Web Conference is a web-based conference call service.
Conference information is displayed on web pages, while the
conference chair reserves and manages a conference through
web pages.
Conference members may also decide whether to attend the
conference through the web any of the two interchangeable
ways: convergence and distribution. Sub-conferences are
provided and the conference can be recorded in full. The system
supports conference joining via a variety of terminals
Web conference service poses excellent potential market for
government and educational organization. Government could
initiate meeting and communicate with upper/lower agencies,
educational organization could launch remote education easily by
using such service.
43
FIGURE 33 WEB CONFERENCE
Service Properties
1. pre-paid function
web conference has pre-paid function. Conference card
records the credit; only when the credit in the card is
enough, person who holds the card has the right to launch
the meeting. Conference fee consists of several parts:
telephone billing, conference billing, conference booking
billing. Card holder can query the conference card, and
charge it with pre-paid telephone card.
2. conference chairman
card holder has the right to book/define/modify/query the
conference, according to the conference notice feedback,
he/she has the right to decide whether or not to launch the
meeting and manage the conference in real-time
3. conference member
the person who got invitation from the chairman, there are
two ways joining the conference: convergence and radiation
4. conference administrator
two kinds of administrator: system administrator and general
administrator. System administrator can manage the general
administrator, general administrator only manages the
conference
5. conference organization
two ways of joining the conference: convergence and
radiation. Convergent way is that the conference member
dialing the conference access code and password after the
launching and been patched to the conference. Radiation way
is that the system automatically calls the conference
44
member, the member off-hook the handler and join the

conference
6. conference booking
conference chairman can book conference, filling start time,
end time, members, for radiation way, he/she needs to fill in
the telephone number.
7. conference status display/management
chairman can observe the conference status and engage
certain operations.
8. sub-conference
conference member could ask for join/quit sub-conference
9. multi-trial calling
for radiation way users, system could launch several trails to
make sure the member attend the meeting
10. address book
chairman can has its own address book, adding usual
conference member.
Web Call
Web Call is a prepaid service which initial call from Web Page.
FIGURE 34 WEB CALL SERVICE
45
Service properties
1. basic voice service
subscriber could initiate a phone call from webpage to a
telephone, which could be fix telephone or soft terminal.
2. video communication
subscriber could initiate a video call from web page to a
telephone, which could be video phone , or compatible soft
phone. Subscriber could switch off the video function as well.
3. booking
subscriber could book a telephone call by entering card
number and password in the web page, caller/calling party
could all be normal pstn telephone. Call duration can be
setup by subscriber itself.
4. family number
subscriber can set up a family number, the billing fee to such
number can be different with normal phone calls
5. voice message
subscriber can enter a text and then initiate the phone call,
when the call is established, the system can play the text
automatically.
ONLY
With the communication development, the need of users for a
more convenient information acquisition measure that allows
mutual exchange at any time and any place becomes more and
more urgent. The launch of the ONLY (One Number Links You)
conception can meet such need perfectly.
The ONLY service is an integrated personal communication
service integrating multiple services. Its core lies in that is
provides a unique ONLY number for a user.
46
F I G U R E 3 5 O N LY S E R V I C E
Service properties
1. Caller party
i.
one number/multiple phoneset
ii. temporary divert

iii. voice message
iv. time routing
v. incoming password
vi. incoming filter
2. Calling party
i.
ONLY account billing
ii. Personal password management

iii. Continuous dialing
iv. Calling limit
v. Calling destination limit
47

Chapter
Dual-homing Technology
Key points
Basic concept of dual-homing.
Operating principle of dual-homing.
Data configuration when applying dual-homing technology.
Basic concepts of dual-homing

As next generation network core control device, softswitch will
control all the access gateway and terminal devices, if something
wrong happens in ss side, the result will be disaster and affect
each and every subscribers which have been registered towards
which. There fore, the requirement of reliability of softswitch is
very high so that every board of softswitch should have backup
partners. However, this yet could not prevent some certain
accidents such as fire alarms, earthquakes etc. In order to
enhance the security and reliability of softswitch device, ZTE
proposed the dual-homing functionality for softswitch: allocate
two softswitch with same configuration to carry half of the traffic
respectively, they work together as partners and controls half of
the network, when one softswitch cannot work properly, the
subordinates it controlled will automatically switch their
commander to another one, and restore the communication in
the shortest time.
In such network configuration, every gateway and terminal will
have two devices :main and assistant.
49
FIGURE 36 BASIC CONCEPTS OF DUAL -HOMING
Dual-homing implementation
principle
In the network designation, adopting dual-homing principle,
each device will control half of the network resources.
1. Network resources have been divided equally into halves and
assigned to two zones controlled by an individual softswitch.
2. the main/assistant softswitch engage data synchronization in
the back stage to ensure the data integration
3. heartbeat mechanism ensures the working status checking
between SS and SS, SS and terminals, SS and gateways.
F I G U R E 3 7 D U A L - H O M I N G I M P L E M E N T AT I O N P R I N C I P L E
50
Chapter 4 Dual-homing Technology
Switching principle based on H.248

Service switching mechanism: auto switching / manual switching
Service switching back: while the problem softswitch restores
status, switch back the service manually or automatically
1. two kinds of heartbeats: one is coming from SS, which is the
Audit message in H.248. mainly initialized by SS to check the
gateway status, if there is no answer, SS consider the
gateway device has switched to another backup SS, and
divert the associate calling route to a backup device.
2. the other one is coming from gateway, supporting
servicechange message or notify message, if there is no
response, gateway will resend the message.
3. time intervals of message between Softswitches is 2ms,
heartbeat adopts standard SIP OPTION message. Softswitch
sends OPTIONS message, if there is no response, it will
resend the message in time interval: 0.5s, 1s, 2s, 4s, 4s, 4s,
4s,4s, 4s. and will send 11 messages at most. After 40s from
the first OPTION message, softswitch will consider it as offline.
FIGURE 38 DUAL HOMING IN H.248
51
Data configuration for dual-homing

SS side data configuration
In dual-homing, the configuration of SS that plays as partners
must keep the same, including their fore-end, back-end and
database server.
To ensure the data integration, using the philosophy as:
1. local area data configuration is mainly configured through
local clients or local billing/accounting system
2. assistant data configuration is initialized through the backstage database serer.
3. In the SS configuration, add an additional note in OTHER
SS area and set up as mapping relationship
F I G U R E 3 9 S S S I D E C O N F I G U R AT I O N F O R D U A L - H O M I N G
TG side data configuration

In the NGN architecture, TG, AG, IAD devices must register
towards one and only one softswitch, which means that these
devices can only controlled by one softswitch at a time. The data
in other softswitch is not active. Therefore, the gateway needs to
be configured a main and a backup MGC information.
52
F I G U R E 4 0 T G S I D E C O N F I G U R AT I O N F O R D U A L - H O M I N G
SG side data configuration

To communicate with PSTN switches, softswitch needs to use
SS7 signals. The role of SG and SS is considered equally in IP
network, SG owns its own SPC code as well.
In the SG
configuration, two signal links (accompanied by two ASP links)
connect the SSs respectively, while only one of them is active in
normal working mode.
Similarly, in the SS side, it is also necessary to configure two
SCTP links pointing to the SG respectively.
F I G U R E 4 1 S G S I D E C O N F I G U R AT I O N F O R D U A L - H O M I N G
53
3rd-party SS data configuration

To achieve the inter-working with
requirements to 3rd-party SS will be:
3rd-party
devices,
the
1. while configuring the subordinates of SSA, the outgoing IP

route to SSA must be setup as direct route, to SSB route as
indirect one.
2. while configuring the subordinates of SSB, the outgoing IP
route to SSB must be setup as direct route, to SSA route as
indirect one.
F I G U R E 4 2 3 R D P A R T Y C O N F I G U R AT I O N F O R D U A L - H O M I N G
Billing, accounting management in dual-homing

For billing and accounting system, in order to maintain the
subscriber data integrity, and forbid one subscriber two CDRs,
only the main user data can be modified, while the backup user
data should merely be synchronized from the other end.
Local billing/accounting system modifies the subscriber data,
accomplish data synchronization through message invoking
Billing of local call is initiated by main SS, while switching to
assistant softswitch, the assistant will take over the billing task.
54
F I G U R E 4 3 B I L L I N G , A C C O U N T I N G M A N A G E M E N T C O N F I G U R AT I O N F O R D U A L HOMING
55
Figures
Figure 1 Telephone history (switching history)........................2
Figure 2 OSI Model.............................................................8
Figure 3 OSI Model and its protocols.....................................8
Figure 4 TCP/IP protocol......................................................9
Figure 5 IP address example..............................................10
Figure 6 IP address illustration...........................................10
Figure 7 IP address classes................................................12
Figure 8 LAN, MAN and WAN..............................................13
Figure 9 Next-generation Network Based on the Softswitch
Technology........................................................................18
Figure 10 Front view of the ZXSS10 SS1a (4U).....................20
Figure 11 Rear view of the ZXSS10 SS1a (4U)......................20
Figure 12 Front view of the ZXSS10 SS1b shelf (12U)...........21
Figure 13 Rear view of the ZXSS10 SS1b Shelf (12U)............21
Figure 14 ZXMSG9000 rack structure..................................22
Figure 15 IAD series products.............................................24
Figure 16 NGN protocol family............................................25
Figure 17 Transaction, Action and Command Relationships.....26
Figure 18 SIGTRAN Signaling Transmission Function Model. . . .27
Figure 19 SIGTRAN Protocol Stack Model.............................27
Figure 20 Schematic Diagram of ISUP Message Transmitted
Through SG.......................................................................29
Figure 21 Schematic Diagram of SCCP Signaling Transmission 29
Figure 22 Management Message Transmission Model.............30
Figure 23 Using B100 to solve private-public address mapping
.......................................................................................32
Figure 24 Class 4 Replacement solution Using Softswitch.......35
Figure 25 Access Solution Cable Via IAD600.......................36
Figure 26 Access Solution Using IAD524..............................37
Figure 27 access solution using I704...................................37
57
Figure 28 Inherit existing IN application platform..................38

Figure 29 Inter-working with H.323 network........................39
Figure 30 Inter-connecting with other NGN network..............40
Figure 31 Service Provisioning Mode of Softswitch Network....41
Figure 32 WEB 800 service.................................................42
Figure 33 Web conference..................................................44
Figure 34 WEB Call service.................................................45
Figure 35 ONLY service......................................................47
Figure 36 Basic concepts of dual-homing.............................50
Figure 37 Dual-homing implementation principle...................50
Figure 38 Dual homing in H.248.........................................51
Figure 39 SS side configuration for dual-homing...................52
Figure 40 TG side configuration for dual-homing...................53
Figure 41 SG side configuration for dual-homing...................53
Figure 42 3rd party configuration for dual-homing..................54
Figure 43 Billing, accounting management configuration for
dual-homing......................................................................55
58

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ZXMSG 5200

Overview of Soft Switch

Historical background of NGN.

Basic concepts of IP network.

Usual IP voice coding and compression methods

Background story of NGN

PSTN network evolution history

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The first automatic dialing system was patented in 1891 by a

From analog to digit

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Chapter 1 Data-communication Basics

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