sjzl20091120-ZXSS10 SS1b (V2.0.1.06.1) - Product Description Manual - EN

ZXSS10 SS1b
SoftSwitch Control Equipment

Product Description Manual
Version V2.0.1.06.1
ZTE CORPORATION
ZTE Plaza, Keji Road South,
Hi-Tech Industrial Park,
Nanshan District, Shenzhen,
P. R. China
518057
Tel: (86) 755 26771900
Fax: (86) 755 26770801
URL: http://ensupport.zte.com.cn
E-mail: support@zte.com.cn
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The ultimate right to interpret this product resides in ZTE CORPORATION.
Revision History
Revision No. Revision Date Revision Reason

R1.0 2009-9-20 First Edition
Serial Number: sjzl20091120

Contents
About This Manual.............................................. i

System Introduction .........................................1
Overview....................................................................... 1
SoftSwitch Structure ....................................................... 1
Edge/Access Layer.......................................................... 2
Core Transfer Layer ........................................................ 4
Control Layer ................................................................. 5
Service Layer ................................................................. 6
System Features ............................................................ 8
Primary Purposes ..........................................................11
Product Standards .........................................................12
Product Structure ............................................ 17
Overview......................................................................17
Hardware Structure .......................................................17
Software Structure ........................................................20
System Working Principles ..............................................20
System Functions ..........................................................21
Interfaces and Protocols ................................. 25
Overview......................................................................25
Physical Interfaces.........................................................26
Protocol Interfaces.........................................................26
Protocol Overview..........................................................28
Megaco/H.248 Protocol ..................................................28
MGCP Protocol ..............................................................30
SIGTRAN Protocol..........................................................32
SCTP Protocol ...............................................................33
M3UA Protocol...............................................................34
SIP Protocol..................................................................35
H.323 Protocol Stack......................................................38
SS7 Signaling ...............................................................39
V5.2 Protocol ................................................................41
Recommendation Q.931 .................................................44
R2 Signaling .................................................................44
China No.1 Signaling......................................................46
Service Provision............................................. 49
Overview......................................................................49
Primary Voice Services ...................................................51
Primary PSTN Voice Services and Supplementary
Services ...............................................................51
IP Centrex ....................................................................57
IP Public Phone .............................................................63
Traditional Intelligent Services.........................................66
F3G Intelligent Services..................................................68
Hybrid Number Assignment Service .................................69
WAN Centrex Services....................................................70
Number Portability (NP)..................................................71
Pre-paid Service ............................................................71
PHS Hotel Service..........................................................72
Callee CRBT ..................................................................72
Caller CRBT ..................................................................72
Non-Personalized CRBT ..................................................73
One Number Multiple-Phone............................................73
IP-based Value-Added Service .........................................74
SoftDA Service ..............................................................76
Multimedia Service ........................................................77
Web Call Service ...........................................................78
Web 800 Service ...........................................................78
Web Conference Service .................................................80
Unified Messaging Service (UMS) .....................................82
Networking Configuration ............................... 85
Overview......................................................................85
Networking Mode...........................................................85
Toll Network Solution .....................................................86
Local Tandem Office Solution...........................................87
Local Access Solution .....................................................88
Hybrid Networking Mode.................................................94
System Configuration.....................................................94
Technical Specifications .................................. 99
Overview......................................................................99
Technical Specifications ..................................................99
Busy Hour Call Attempts (BHCAs) ....................................99
Maximum Subscriber Capacity....................................... 100
Signaling Processing Capability...................................... 100
Protocol Processing Capability ....................................... 100
Call Authentication....................................................... 101
Charging Performance .................................................. 103
Time Monitoring .......................................................... 103
Reliability and Availability ............................................. 103
Physical Specifications.................................................. 104
Environmental Requirements ........................ 105
Power Supply and Power Consumption ........................... 105
Grounding Requirements .............................................. 106
Antistatic Requirements................................................ 107
Anti-Electromagnetic Interference.................................. 108
Dust-proof Requirements.............................................. 108
Temperature and Humidity Requirements........................ 108
Noise ......................................................................... 109
Lighting...................................................................... 109
Atmospheric Pressure................................................... 109
Air Pollution ................................................................ 109
Figures .......................................................... 111
Tables ........................................................... 113
List of Glossary.............................................. 115
About This Manual
Purpose This manual describes the ZXSS10 SS1b SoftSwitch product in

terms of its system structure, working principles, interface proto-
col, network management system, service provision, networking
configuration and performance specifications.
This manual provides users of ZTE SoftSwitch product ZXSS10
SS1b SoftSwitch control equipment an overall understanding in
terms of its software system, establishing a basis for other sup-
port manuals.
Intended This manual is intended for engineers and technicians who per-
Audience form operational activities on the ZXSS SS1b SoftSwitch control
equipment.
What Is in This This manual contains the following chapters:
Manual
TABLE 1 CHAPTER SUMMARY
Chapter Summary
Chapter 1, System This chapter describes the ZXSS10 SS1b

Introduction SoftSwitch control equipment in terms of its
overall structure, features, primary purposes
and product standards.
Chapter 2, System This chapter describes the ZXSS10 SS1b

Structure system in terms of its hardware structure,
software structure, working principles and
functions.
Chapter 3, Interfaces This chapter describes the connection of

and Protocols the product to the whole network, provided
internal and external interfaces, adopted
signaling/protocol, helping users obtain the
knowledge of connection mode of the product
to other devices.
Chapter 4, Service This chapter describes the service provision of

Provision ZXSS10 SS1b.
Chapter 5, This chapter describes ZXSS10 SS1b in terms

Networking of its service provision and implementation
Configuration modes, helping subscribers have the knowledge
of the system service functions.
Chapter 6, Technical This chapter describes ZXSS10 SS1b in terms

Specifications and of its technical specifications such as system
Environmental processing capability, charging performance,
Requirements time monitoring, load capability, reliability and
availability and its environmental requirements.
Related The following documentation is related to this manual:

Documentation
� ZXSS10 SS1b SoftSwitch Control Equipment Documentation
Guide
Confidential and Proprietary Information of ZTE CORPORATION i

ZXSS10 SS1b Product Description Manual
� ZXSS10 SS1b SoftSwitch Control Equipment Hardware De-

scription Manual
� ZXSS10 SS1b SoftSwitch Control Equipment Software De-
scription Manual
ii Confidential and Proprietary Information of ZTE CORPORATION

Chapter 1
System Introduction
Table of Contents
Overview........................................................................... 1
SoftSwitch Structure ........................................................... 1
Edge/Access Layer.............................................................. 2
Core Transfer Layer ............................................................ 4
Control Layer ..................................................................... 5
Service Layer ..................................................................... 6
System Features ................................................................ 8
Primary Purposes ..............................................................11
Product Standards .............................................................12
Overview
This chapter describes the ZXSS10 SS1b SoftSwitch control equip-
ment in terms of its overall structure, features, primary purposes
and product standards.
SoftSwitch Structure
The NGN based on the SoftSwitch system adopts the multi-layer
architecture model. The ZXSS10 SoftSwitch completely conforms
to and supports the standard multi-layer architecture model.
The ZXSS10 SoftSwitch system architecture falls into four layers:
� Service layer
� Control layer
� Core transfer layer
� Edge/access layer
Figure 1 shows the multi-layer architecture of the ZXSS10 SS1b.
Confidential and Proprietary Information of ZTE CORPORATION 1

FIGURE 1 MULTI-LAYER ARCHITECTURE OF ZXSS10 SS1B
Standard interfaces are used for communication between layers.

Under the control of the SoftSwitch control equipment, the related
NE devices cooperate together to implement service functions of
the system.
Edge/Access Layer
At the edge/access layer, the existing access gateways and new
access terminal devices interwork with the existing communication
networks and implement the access from various communication
terminals (such as analog telephone, SIP Phone, PC Phone visual
terminal and intelligent terminal) to the IP core layer.
ZTE provides the following types of edge/access layer devices:
1. ZXMSG 9000 media gateway:
2 Confidential and Proprietary Information of ZTE CORPORATION

Chapter 1 System Introduction
The ZXMSG 9000 can work as the Trunk Gateway (TG), Sig-
naling Gateway (SG) or Access Gateway (AG) or realize the
integrated functions of the three gateways via varying board
configuration. It is located in the edge/access layer of the
SoftSwitch network architecture
� When working as an AG, the ZXMSG 9000 is located in the
access layer of the Packet Switched Network (PSN) and en-
ables PSTN subscribers to access the IP network. It accepts
the access of common Z interface subscribers, ISDN digital
subscribers, V5 subscribers and DSL subscribers, imple-
menting the function of conversion between the voice/fax
signals at the PSTN/ISDN user side and the voice/fax sig-
nals at the IP network side.
� When working as a TG, the ZXMSG 9000 is located in the
core layer of the PSN. It accepts the access of No.7 trunk
subscribers and PRI subscribers. It enables PSTN sub-
scribers to access the NGN, implementing the function of
conversion between the voice/fax signals at the PSTN/ISDN
trunk side and the voice/fax signals at the IP network side.
� When working as an SG, the ZXMSG 9000 is located in
the core layer of the PSN and can effectively implement
the interworking of signaling between the Switched Cir-
cuit Network (SCN) and the IP packet network. It receives
and sends standard SS7 signaling messages at the circuit
switching network side and converts the signaling mes-
sages into the standard SIGTRAN protocol messages at the
packet network side so as to interwork with the SoftSwitch
core control device
2. Session border control (SBC) gateway
This kind of products covers the ZXSS10 B100/B200 (the
B100/B200 for short). The B100/B200 is located between the
customer premises network and the operator’s service net-
work to complete address translation and user access control.
The address translation function of the B100/B200 involves
signaling address translation and media address translation.
The user access control function of the B100/B200 includes
user access control and media stream channel control. The
ZXSS10 B100 (the B100 for short) enables/disables the media
channel under the control of the SoftSwitch. The ZXSS10 B200
(the B200 for short) is a high-performance, large-capacity car-
rier-class broadband gateway product launched by ZTE follow-
ing the B100. Located at the edge of the IP-IP network, the
B200 controls the IP-based session signaling protocols (e.g.
SIP, H.323, MGCP and MEGACO/H.248), provides edge access
and network interworking, implements IP address translation,
ensures service security and guarantees SLA service perfor-
mance.
� The B100/B200 can be placed between the operator’s ser-
vice network based on public network addresses and the
customer premises network based on private network ad-
dresses. All the signaling and media information from the
private network ports of the B100/B200 are transferred in a
loss-free way to the public network ports of the B100/B200
and vice versa.
� The B100/B200 can be placed in the operator’s service net-
work or VPN based on public network addresses to connect

multiple customer premises networks based on private net-

work addresses via Network Address Translation (NAT) or
HTTP agent.
3. IAD
The Integrated Access Device (IAD) is a kind of SoftSwitch ac-
cess device, which is designed for individual and small-scale
enterprise users. It enables user terminal devices such as PCs,
phone sets and fax machines to access the PSN. The IAD over
the IP network supports transparent transmission of data ser-
vices and can process VOIP services and provide users with
multiple services.
ZTE provides a variety of IADs. The IAD device is configured
with different physical interfaces depending on the type. For
more information, refer to the IAD-related manuals.
Core Transfer Layer

The core transfer layer refers to the PSN composed of backbone
transmission devices such as IP routers. It is the bearer basis of
the SoftSwitch network.
At present, the ZXR10 T64E/T128 is the backbone router. The
ZXR10 T64E/T128 can be used to satisfy the application needs of
the convergence layer and the backbone layer of enterprise net-
works or operator’s networks. The system adopts the modular de-
sign concept and different functions of the router are implemented
by different hardware boards. The 1:1 redundancy design of the
key modules of the system make the system more reliable and
extensible and facilitates the user’s configuration.
1. The ZXR10 T64E/T128 chassis has a total of eight or sixteen
interface board slots that provide abundant types of interfaces,
including:
� 2.5G POS interface
� 155M POS interface
� 622M POS interface
� 622M ATM interface
� 155M ATM interface
� SFP GE interface
� GBIC GE interface
� 10/100Base electrical interface
� 10/100Base optical interface
� Channelized E1 interface
� Channelized CP3 interface
� Channelized CE3 interface
2. In terms of software, the ZXR10 T64E/T128 adopts the ZXROS
network OS platform, which is completely owned by ZTE with

regard to the IPR and supports rich protocols. The platform

supports the following protocols and standards:
� Link layer protocols: PPP, MPPP
� Network layer protocols:IP, ICMP, ARP, V-SWITCH, SMART-
GROUP
� Transmission layer protocols: TCP, UDP
� Routing protocols: RIP v1/v2, OSPF v2, BGP4, IS-IS
� IPv6protocol
� RIPng, OSPF v3, BGP4+ and IS-IS6 routing protocols
� MPLS/VPN, VPWS, VPLS, QOS and RSVP TE strategic rout-
ing, load balancing
� Tunneling protocol: GRE
� Application layer protocols: Telnet, FTP, TFTP
� Network layer control and application: NAT, ACL, URPF
� NM protocol: SNMP v1/v2/v3, RMON v1, NTP
Control Layer
The control layer refers to the SoftSwitch control equipment that
implements 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. Functions and features of the control layer are
as follows:
� The ZXSS10 SS1b is large-capacity SoftSwitch control equip-
ment that can process millions of calls.
� In the SoftSwitch architecture, the ZXSS10 SS1b is the core
device independent of the bottom-layer bearer protocols and
completes the functions such as call control, protocol adapta-
tion and processing, media gateway access control, resource
allocation, protocol processing, routing, authentication and
Call Detail Record (CDR).
� The ZXSS10 SS1b can provide all primary call services, supple-
mentary services and point-to-point multimedia services of the
PSTN. Through cooperating with the service layer equipment
SCP and the application server, it can also provide users with
traditional intelligent services, value-added IP services, diverse
third-party value-added services and new intelligent services.
� The ZXSS10 SS1b can place user data in to its imbedded data-
base or in to the Smart Home Location Register (SHLR).

Service Layer
The service layer provides the network with applications and ser-
vices, customer-oriented integrated intelligent services and ser-
vice customization.
1. ZXWN SHLR
The ZXWN SHLR is developed by adding some functions of the
fixed network HLR based on the standard HLR of the 3G mo-
bile network. It stores the logical numbers and physical num-
bers of fixed network subscribers as well as their mapping rela-
tions and the information such as intelligent service subscrip-
tion data of fixed network subscribers. It provides integrated
and centralized subscriber data processing for mixed number
assignment of the local network, Number Portability (NP) ser-
vices, global intelligent service triggering, and transition from
PHS to 3G and convergence of the fixed network and the mo-
bile network.
2. ZXUP10
As the control point for value-added services and IN services
in the SoftSwitch network, the ZXUP10 interacts with the call
control entity to implement call control. The Service Switching
Point (SSP) of the SoftSwitch or the SSP of the intelligent net-
work can serve as the call control entity.
The ZXUP10 can control SIP calls of the SoftSwitch. Such calls
can be initiated by the SoftSwitch or by the ZXUP10. The SIP
protocol is applied at the interface between the ZXUP10 and
the SoftSwitch. SIP protocol is used to establish, terminate
and manage end-to-end call sessions.
Upon receipt of a call from the SoftSwitch, the ZXUP10 in-
vokes the corresponding value-added service according to the
obtained call information. It operates the call according to the
service execution needs, for example, it redirects the call, ac-
cepts and transfers the call or processes the call as the agent.
In addition to the above functions, the ZXUP10 supports initi-
ating calls to the SoftSwitch, that is, it can initiate a call in the
place of the user to the SoftSwitch according to the request
of third party application such as Web, Email, push or instant
messaging, so as to implement call control based on web and
messages.
The ZXUP10 is an integrated system platform. Using the Parlay
gateway as the core device, it consists of multiple devices such
as the media server, application server, Text To Speech (TTS)
server, O console and signaling gateway.
Devices contained in the ZXUP10 are listed in the following:
� Parlay gateway
The Parlay gateway is the core device of the ZXUP10 prod-
uct. It converges and encapsulates different bottom-layer
network resources, externally provides open APIs based on
the Parlay protocol and offers the platform for developing
third party’s services. The Parlay gateway is an indepen-
dent component irrelevant to the SoftSwitch in the control
layer, so the services are separate from call control and new

services can be easily introduced. Structurally, the Parlay

gateway is divided into three functional modules: Protocol
adapter, basic function and, service control.
� Application server
This is the server on which service applications run. The
services running on it are generally developed by third
party application providers. Multiple application servers
can be connected to and use the resources of the same
Parlay gateway but they run their own services and are
logically independent from each another.
The application server and the Parlay gateway may be sep-
arated or combined in the same server. The Parlay API is
used between the application server and Parlay gateway.
� Media server
The Media Server (MS) is an independent device provided
by the ZXUP10 system especially for the media resource
function. It provides media processing functions in intelli-
gent services, including resource functions, communication
functions and management & maintenance functions (e.g.
collection and decoding of DTMF signals, signal tone gen-
eration and sending, announcement transmission, confer-
ence function and conversion between codec algorithms).
� TTS server
The Text To Speech (TTS) server is a device used for voice
synthesis and converting text information into voice to play
to telephone users.
� VIP customer (VIC) system
The ZXSS10 VIC system is a new service provision system
based on the SoftSwitch. Multiple functions are developed
on the basis of the old standard attendant console, includ-
ing instant charging and hotel system interface, to meet the
communication and service management requirements of
the VIP customers better.
The call control between VIC and SS is performed via the
simple and resilient SIP protocol, featuring with straight-
forward service communication and strong expansibility.
The VIC attendant console system can provide the stan-
dard console, simple console and computer console. They
work separately or cooperate with each other if required.
� SIP short message center
In the SIP short message center, the processing of sig-
naling and services are separated via different modules
hence greatly promoting the system processing capability
and system work division.
SIP short message center expands the service provision of
the short message system via overlapping modules. The
broad expansion scale and simple implementation mode
provide perfect solution for the continuous function update.
SIP short message center can expand capacity for specific
module in the system if the upgrade is required by the cus-
tomer. The resilient capacity expansion modes of the short
message center minimize the influence on the system and

therefore protect the customer investment and system ex-

pansibility to the maximum extent.
� WiMAX
World Interoperability for Microwave Access (WiMAX) is
the wireless Metropolitan Area Network (MAN) technology
based on the IEEE802.16 standards. The ZTE WiMAX
system supports the encryption technology specified
according to the 802.16e standards, encrypting the sub-
scriber traffic and signaling to guarantee the confidentiality
and integrity of the user data on the air interface. The
ZTE WiMAX system supports QoS. The service provider
can customize various policies, such as subscriber priority,
bandwidth assignment on demand and delay or jitter
requirements.
System Features
The ZXSS10 SS1b is designed with the most advanced distributed
system, not only lowering the overall running risk, but also im-
proving the processing capability and efficiency of the system. A
dedicated communication equipment platform serves as the hard-
ware platform of the system and a real-time multi-task operating
system serves as the software platform, which completely guar-
antees the requirements of carrier-class operation equipment for
large capacity and high reliability and ensures excellent cost-per-
formance ratio.
As the core equipment of the SoftSwitch network, the ZXSS10
SS1b features with the following in terms of performance:
1. Powerful networking capability, Tandem networking capability
and long distance traffic balancing.
� The combinations of the ZXSS10 SS1b and various gateway
devices, such as SS+MSG, SS+AG or SS+IAD, lead to the
buildup of long-distance network and local network, pro-
viding the long-distance, local solutions and convergence
of long-distance and local solutions. Besides, the combina-
tions support solution of dual-Tandem office.
� The system depends upon widely distributed data networks
as bearer networks and supports various network topolo-
gies such as star, tree, ring, and start + ring.
� The system has powerful penetration capability and sup-
ports various network scales. Therefore, it can not only
provide services for a large number of centralized users,
but also can expand all services to scattered users on the
edge of cities.
� The system inherits the advantages from existing networks
perfectly and supports the complete interworking with
PSTN/ISDN networks, No.7 signaling networks and mobile
communication networks.
� The system provides perfect integrated Network Manage-
ment System (NMS) and standard network interfaces so
that it can be connected to the local network management

center flexibly and conveniently. The ZXSS10 SS1b pro-

vides detailed CDRs and sends the CDRs to the Billing Cen-
ter via a standard interface so as to implement centralized
settlement in the Billing Center.
2. Consummate integrated access capability
� The system supports direct access of various types of gate-
ways, including trunk gateways, access gateways, NASs,
signaling gateways, media gateways, wireless access gate-
ways, and integrated access devices.
� The system supports direct access of media servers.
� The system supports direct access of various digital termi-
nal devices such as PC, Softphone, IP Phone, SIP Phone
and MGCP Phone.
� The system supports policy-based device control, dynamic
addition of new devices and dynamic domain area adjust-
ment.
� The system supports direct access of existing H.323 net-
work devices such as H.323 gateways and H.323 terminals.
� It supports access of V5 access network equipment through
the trunk gateway.
� It supports access of PBXs via Q.931, R2, and No1 proto-
cols.
3. Powerful network interworking capability
� Using different gateways, the system supports the com-
plete interworking with various existing networks such as
PSTN/ISDN, No.7 signaling, Intelligent Network (IN) and
Public Land Mobile Network (PLMN).
� Using standard protocols, the system supports the inter-
working between the same or heterogeneous networks of
different manufacturers and carriers.
4. Excellent expansibility
� The system structure features with multiple layers. Every
layer is independent of each other and can develop sepa-
rately.
� The distributed design is adopted by the system for conve-
nience of smooth capacity expansion.
� With the plane structure, the system supports the network-
ing mode with multi-domain expansion, so that users can
plan the network construction scale according to specific
requirements.
5. Powerful service provision capability
� With the multi-layer architecture, the services provided by
the system are independent of the network and media.
Once a service is generated, it can be shared by the en-
tire network.
� As a unified service proxy and service bearer platform, the
SoftSwitch control equipment masks the complexity of the
bottom-layer network for the service layer, for the conve-
nience of the generation of complicated integrated services.

� There are three service provision modes: the SoftSwitch

control equipment provides services itself; the equipment
cooperates with the traditional SCP to provide services and
the equipment cooperates with the Application Server to
provide services. Such service provision modes completely
guarantee that the system has superb flexibility in utilizing
existing resources and developing future services and users
can select and expand services flexibly and conveniently
according to specific requirements.
6. Outstanding manageability and operability
� The equipment provides an integrated NMS, supports three
interface modes (GUI, CLI and WEB) and implements net-
work management functions such as centralized mainte-
nance, network management, fault management and static
configuration management.
� The system provides complete operation and maintenance
functions itself, including Element Management System
(EMS) functions such as CLI, authority management, log,
traffic statistics, signaling tracing, SNMP, data configu-
ration, alarm and diagnosis test. In addition to this, it
can provide detailed CDRs, so that the Billing Center can
perform centralized billing. The system uses a large net-
work database as its data platform and provides multiple
charging data backup modes to guarantee the security of
charging data and fully meet the actual requirements of
telecom operation.
� It uses the ZTE NetNumen integrated NMS as its operation
and maintenance interface. The NetNumen integrated NMS
allows you to manage all NEs provided by ZTE in the net-
work.
7. High reliability
� With respect to hardware design, the equipment provides
hot backup for all important components and all cards sup-
port hot swapping with full fault tolerance, so as to guar-
antee that no single-point fault will occur during operation.
� The redundant active/standby mode or distributed cluster
mode, including remote disaster recovery, is used in equip-
ment networking. The components in each key system are
also redundant and hot swappable.
� As for software, the equipment provides hot backup for all
vital data. If the active system fails, the standby system
will take over all calls under conversation and ringing, com-
plete the calls and save the call records.
� It provides remote disaster recovery. Two soft switches
control the user groups and relevant gateways (TG, SG, AG,
and so on) in two domains. Under normal circumstances,
active/standby mode or load-balancing mode is used for
the access device. When a node fails, all gateways con-
trolled by the domain register at the disaster recovery soft
switch, so that the services in the domain can be imple-
mented properly.
� The system satisfies the requirement for 99.99% reliability
of carrier-class equipment.
8. Security

� Network security
The system supports configuration of different user levels,
which prevents a user of lower level from performing op-
erations of a higher level. Real-time login information of
unauthorized users that attempt to access the system is
recorded in the background maintenance terminal. Alarms
with text message and images will be generated if there is
any request that imposes risks on the equipment, such as
multiple login failure, connection to IP address, and a large
amount of concurrent traffic. In addition, the ZXSS10 SS1b
has built-in safety check tools to defend against suspicious
attacks and report them to the system.
Also, ACL and port on-off are supported.
� User security
ZTE SoftSwitch system sets a device authentication code
for each authorized device and judges whether it should
allow or refuse the access of a user based on the authenti-
cation code.
Primary Purposes
A network based on SoftSwitch technology has the basic features
of a Next Generation Network (NGN), which can provide integrated
multimedia services such as data, voice and video in a centralized
manner. Currently, the SoftSwitch system primarily offers voice-
related and IP based value-added services, covering C0 through
C5:
1. It can work as the end office (C5 office), completely compat-
ible with all service provision capabilities of the PSTN switch,
supporting signaling and protocols such as MGCP, H.248, V5,
DSS1, SIP and H.323, supporting the access of multiple termi-
nals such as POTS, ISDN and packet terminals such as MGCP,
H.248, SIP, and H.323, and offering colorful voice, data and
multi-media service provision capabilities.
2. It can work as the Tandem office (C4 office) or toll office, sup-
porting traditional signaling such as SS7, SS5, R2, DSS1 and
V5 protocol. When combining with devices such as SG and TG
to perform networking, it has the access and trunk networking
capabilities as the existing PSTN switch.
3. It supports functions such as black/white list, call authentica-
tion, call interception and large-capacity bill storage. Also, it
can work as a gateway office.
4. It supports the INAP and MAP/MAP+ protocols. In addition, it
can work as SSP in the traditional intelligent network or in the
whole network IN system.
5. It provides WEB-based IP value-added services.
6. It provides Point-to-point multimedia communication services.
7. In the above various applications, the ZXSS10 SS1b, as the
control core of the system, is responsible for processing basic
functions such as call control, protocol adaptation and resource
management. Furthermore, as the charging collection point of
the system, the device provides CDR of all calls as the basis
for the billing system.

In the above various applications, the ZXSS10 SS1b, as the control

core of the system, is responsible for processing basic functions
such as call control, protocol adaptation and resource manage-
ment. Furthermore, as the charging collection point of the sys-
tem, the device provides CDR of all calls as the basis for the billing
system.
Product Standards
The ZXSS10 SS1b complies with the following standards, as shown
in Table 2.
TABLE 2 PRODUCT STANDARDS
Protocol Standard
H.248 IETF RFC3525
ITU-T H.248.1
ITU-T H.248.2R1
ITU-T H.248.4
ITU-T H.248.14
ITU-T H.248.15
ITU-T H.248.23
ITU-T H.248.23R1
ITU-T H.248.24
ITU-T H.248.25
ITU-T H.248.25R1
ITU-T H.248.26
ITU-T H.248.28
ITU-T H.248.28R1
ITU-T H.248.32
ITU-T H.248.39
ITU-T H.248.42
MGCP IETF RFC2705
IETF RFC3435
IETF RFC3660

Protocol Standard
SIP IETF RFC3204
IETF RFC3261
IETF RFC3262
IETF RFC3264
IETF RFC3265
IETF RFC3311
IETF RFC3515
IETF RFC2976
IETF RFC3327
IETF RFC3323
IETF RFC3325
IETF RFC3372
IETF RFC3398
IETF RFC3665
IETF RFC4123
IETF RFC3666
IETF RFC4028
And related supplementary

protocols
SIP-T IETF RFC3372
IETF RFC2976
IETF RFC3204
IETF RFC3398
SIP-I ITU-T Q.1912.5
H.323 ITU-T H.323include H.225.0, H.245,

H.450
Q.931
NCS ITU-T J.169
ETSI TS 101 909-9

Protocol Standard
SIGTRAN RFC2960
RFC3332
YD/T1192-2002
RFC3807
RFC3057
NO.7 GF001-9001
Q.700~Q.707,Q.709
GF001-9001
Q.721~Q.725
YDN-038-1997
Q.761~Q.764
Q.765~Q.766
Q.731~Q.737
Q.698~Q.699
GF010-95
Q.711~Q.714,Q.716
GF 011-1995
Q.771~Q.775
GF 017-1995
Q.1551,Q.1600,Q.1601
YD/T1226-2002
YD/T1201-2002
GF017-95
ITU-T Q.122409/97
ITU-T Q.122809/97
China Talecom PHS MAP Protocol

Technical standards
As same as the supplementary

requirements of SHLR on the
SoftSwitch equipment
R2 ITU Q.400~Q.499

Protocol Standard
NO.1 China No.1 comply international

GF002-9002’s DL signaling and MFC
signaling standard
R5 Q140~Q164
DSS1 YDN034-1997
V5 YDN 020-1996(V5.1)
YDN 021-1996(V5.2)
YDN 108-1998
ITU-T Q.964,Q.965
YD 1380.1-2005-I.pdf
SNMP RFC1157
FTP RFC0959
RADIUS RFC2865
RFC2866
DTMF RFC2833
SNTP RFC1769

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Chapter 2
Product Structure
Table of Contents
Overview..........................................................................17
Hardware Structure ...........................................................17
Software Structure ............................................................20
System Working Principles ..................................................20
System Functions ..............................................................21
Overview
This chapter describes the ZXSS10 SS1b system in terms of its
hardware structure, software structure, working principles and
functions.
Hardware Structure
Compared with the traditional switch, the hardware structure of
the SoftSwitch control equipment is simpler. It comprises the
foreground real-time processing part, background (OSS, Database
Server) and switched Ethernet unit (SSN) used for interconnect-
ing all internal modules. Figure 2 shows the connection of ZXSS10
SS1b.

FIGURE 2 CONNECTION OF ZXSS10 SS1B
The description of each part is as follows:

� The SUN work station can be used as the background data
server and the 1+1 redundancy of servers is supported. Also,
the PC server can be used as the data server.
� ZXSS10 SS1b adopts the dedicated hardware platform based
on the multi-processor card. Compared with the PC platform
and commercial work station platform, it features with the ad-
vantages such as large capacity, high density, high reliability
and excellent cost performance.
� ZXSS10 SS1b is installed in the 19-inch cabinet. Figure 3
shows the single-shelf ZXSS10 SS1b equipment.

Chapter 2 Product Structure
FIGURE 3 ZXSS10 SS1B CABINET
� ZXSS10 SS1b, based on the Ethernet structure with high

speed serial data bus as core, is a dedicated hardware platform
with multi-processor hardware structure adopted. The internal
communication uses the switching Ethernet bus. Figure 4
shows the hardware logical relationship of ZXSS10 SS1b.
FIGURE 4 HARDWARE LOGICAL RELATIONSHIP OF ZXSS10

SS1B
� In the system slots, the main control processor module (board)

controls the whole system and network, assigning responsibil-
ity of each processor.
� As the system core, protocol processor module performs the
core functions such as network control and service generation.
Processor modules are connected with high speed Ethernets,

composing a concurrent multi-processor system in a “loose

coupling” mode.
� The network interface module provides external interfaces con-
necting gateway devices via IP network and internal interfaces
connecting with the application server, database, authentica-
tion server, management and maintenance terminals and IN
SCP. For the purpose of security, physically, the two kinds of
interfaces are independent of each other.
Software Structure
The ZXSS10 SS1b software system is distributed, featuring with
the multi-layer and modulized structure. The software system
is independent of the hardware platform, convenient to be up-
graded.Figure 5 shows system structure of the SoftSwitch control
equipment software.
FIGURE 5 SOFTWARE STRUCTURE OF SOFTSWITCH CONTROL

EQUIPMENT
System Working Principles

ZXSS10 SS1b, based on the Ethernet structure with processor
modules are connected with high speed Ethernets, is a concur-
rent multi-processor system in a “loose coupling” mode.
The network interface module provides external interfaces con-
necting gateway devices via IP network and internal interfaces
connecting with the application server, database, authentication
server, management and maintenance terminals and IN SCP. For
the purpose of security, physically, the two kinds of interfaces are
independent of each other.
The external interfaces can provide the routing function, configur-
ing static routing to ensure the resilient and convenient access of
IP core network of the SoftSwitch devices.

IP packets carried with various protocols received by the network

interface module are processed on the processor module.
On the processor module, protocol processing, service generation
and control are performed via software. According to varied con-
figuration, the processor module can process different protocols of
different network nodes.
System Functions
Functions realized by ZXSS10 SS1b include the following:
1. Call processing control
It implements the primary and enhanced call processing.
It identifies events reported by the media gateway, including
hook-off, dialing and hook-on, controls the media gateway on
sending tones of various signals to users, such as dialing tone,
ringing tone and ring-back tone, and provides the dialing plan
meeting the requirements of carriers.
It controls the establishment, hold-on and disconnection of the
primary call, including call processing, connection control, trig-
ger detection of the intelligent call and resources control. It
accepts the request of monitoring from the service exchange
function and processes call-related events in the request. It
accepts the call control information from the service exchange
function, supporting establishing and monitoring the call.
It supports the primary two-party and multi-party call control.
Here, the supported multi-party call control covers the its par-
ticular logic relationship, the entry/exit/segregation/sitter-in of
the conversation members involved and control of the voice
mixing process.
� The cooperation between the ZXSS10 SS1b and SG heads
to establish a call and free it. Here, the No.7/IP protocol can
be used, while take SCTP as the bearer protocol. Moreover,
the ZXSS10 SS1b can control the media gateway to send
IVR for implementing services such as second dial.
� ZXSS10 SS1b can be directly connected to the H.248 ter-
minal, MGCP terminal, SIP client terminal, H.323 terminal
and NCS terminal to provide services.
� ZXSS10 SS1b can work as the end office, toll office, Tandem
office and gateway office. It application covers C0~C5.
� ZXSS10 SS1b supports call processing like the tandem of-
fice and imports the SHLR to support the MAP/MAP+ pro-
tocols, performing tandem exchange for fixed network in-
telligenlization and supporting double tandem function.
� The ZXSS10 SS1b supports multi-area code function.
2. Access protocol adaptation
It is responsible for implementing the adaptation processing of
various access protocols (signaling).
The ZXSS10 SS1b is an open and multi-protocol entity with
standard protocols used to communicate with various media

gateways, terminals and networks. The protocols cover the

following: H.248, SCTP, ISUP, TUP, INAP, CAP, H.323, NCS,
RADIUS, SNMP, SIP /SIP-T/SIP-I, M3UA, MGCP, Q.931, Q.51,
V5UA, IUA, BICC, MAP/MAP+, R1, R2, R5, China No.1, R1.5
and so on.
3. Service provision/interface
� The ZXSS10 SS1b directly provides services offered by the
PSTN/ISDN switch, including primary services and supple-
mentary services.
� The ZXSS10 SS1b is configured with the standard INAP
interface, which cooperates with the traditional IN SCP to
offer services provided by the existing IN.
� The ZXSS10 SS1b is configured with the standard INAP/SIP
interface, which cooperate with the application server to
provide multiple value-added services. It is also configured
with the interface to connect the application server for the
convenience of third-party service provider developing ser-
vices.
4. Interconnection and interworking
As the external interface of the SoftSwitch system, the ZXSS10
SS1b is responsible for interworking with other equal entities.
� The ZXSS10 SS1b implements the interworking between
PSN and the existing SS7 network via SG. It implements
the interworking between PSN and the existing CAS net-
work via R2 signaling and China No.1 signaling.
� Interworks with the IP telephony network of the existing
H.232 system via H.323 protocol.
� Provides the interworking among the H.248, MGCP, H.323,
SIP and NCS terminals inside the IP network
� Uses the SIP protocol to interwork with the SIP network
system.
� Uses the SIP/SIP-I/SIP-T protocols to implement the inter-
connection and interworking among the SoftSwitch control
domains.
� Uses the V5 protocol to access the access network with the
cooperation of TG.
� Uses the ISDN protocol to access the ISDN PBX group with
the cooperation of MG.
� The ZXSS10 SS1b interworks with the existing IN via SG
to provide users with multiple intelligent services. The IVR
function needed by the intelligent services is implemented
by the media server and media gateway under the control
of the ZXSS10 SS1b.
� Uses the MAP or MAP+ protocol to interwork with SHLR,
querying user data or user subscription information from
SHLR.
5. Application support
The ZXSS10 SS1b provides application support, including
charging and authentication.

� The ZXSS10 SS1b proper does not offer the charging sys-
tem but generate Call Detail Record (CDR) or metering bill.
Once the conversation is over, the charging data can be
output. For the long time conversation, the charging data
can be output during the conversation.
� The ZXSS10 SS1b can be connected to the charging center
via standard protocols to output the charging data, namely,
CDR. Here, for the general services, the charging center
can use the FTP protocol to collect the charging data pro-
vided by the SoftSwitch equipment on a regular basis. The
minimum cycle of collecting data is five minutes. For the
credit or pre-paid service, the SoftSwitch equipment uses
the Radius protocol to send the real-time charging data to
the charging center. This service can be real-timely discon-
nected.
� The ZXSS10 SS1b can send user information such as open-
ing account, canceling account and service attribute mod-
ification to the charging system via the standard XML or
MML interface. With the introduction of fixed network in-
telligentization, assigning a number to the user and the
related subscription can be implemented via SHLR. Then,
SHLR connects to the charging system via interfaces.
� The ZXSS10 SS1b supports the authentication on users and
gateway devices to avoid invalid user/device access.
6. Address resolution
ZXSS10 SS1b is responsible for converting the E.164 address
to IP address and providing the address resolution.
7. Voice processing
� The ZXSS10 SS1b can control the media gateway to decide
whether to use the voice compression and provide optional
voice compression algorithms such as G.711, G.723, G.726
and G.729.
� ZXSS10 SS1b can resiliently control the media gateway to
decide whether to use the echo cancellation technology.
� ZXSS10 SS1b also can resiliently control the media gate-
way to set size of the voice packet buffer to minimize the
influence of jitter on the voice quality.
8. Fax/MODEM processing
ZXSS10 SS1b provides Fax/MODEM processing function, sup-
ports T.30 and T.38 protocol.
9. Resources control
ZXSS10 SS1b provides the resources management function,
centrally managing resources in the system, including the as-
signment, release and control of the voice resources.
10. Operation and maintenance
ZXSS10 SS1b provides the consummate operation and mainte-
nance functions, supporting local maintenance management.
ZXSS10 SS1b supports the SNMP-based NM mechanism and
remote centralized NM. It can be brought to under the unified
management of the NM center along with other NEs in the sys-
tem.

Usually, NetNumen, the unified network management system

developed by ZTE Corporation, manages, operates and main-
tains ZXSS10 SS1b and other ZTE SoftSwitch devices.
NetNumen is applied to manage all data devices in the ZTE
SoftSwitch system, covering NE management, NM, service
management and so on.
ZXSS10 SS1b supports the third-party interfaces, including the
SNMP V2 alarm interface, performance data connection inter-
face, NE-supported FTP interface and MML billing interface.
The interface machine shall be installed.
� NE management layer
NE managers of various NEs work at the NE management
layer, support the centralized NE management on the uni-
fied platform, and implement operations and maintenance
on one or more device. All the NE managers work as the
functional component imbedded in the NM system. The
NE manager can be loaded according to the device types
of the managed network to implement the point-to-point
management on these devices.
All the NE managers interact with the managed devices via
the SNMP management protocol and TELNET interface, im-
plementing functions such as configuration management,
rack status monitoring, performance management, fault
management and MIB browsing.
� NM Layer
The NM layer provides network views of the whole network,
monitoring and managing all subnets formed by various
devices in the managed network. Moreover, it provides
the traffic statistics and comprehensive fault analysis at
the network layer, supporting the device management and
network optimization under the full network circumstances.
The NM layer manages the network in terms of its topology,
performance, fault and security and the NM policy.
� Service management layer
The service management layer functions the following: ser-
vice provision, service control and monitor, service-related
charging, breakdown diagnosis plan & task arrangement,
unified service report management. The service manage-
ment layer is based on the NE and network management
layers, implementing the user-oriented service manage-
ment.
The service management subsystem includes the fault re-
port system, VPN service management, report manage-
ment and provision of standardized open interface and API.

Chapter 3
Interfaces and Protocols
Table of Contents
Overview..........................................................................25
Physical Interfaces.............................................................26
Protocol Interfaces.............................................................26
Protocol Overview..............................................................28
Megaco/H.248 Protocol ......................................................28
MGCP Protocol ..................................................................30
SIGTRAN Protocol..............................................................32
SCTP Protocol ...................................................................33
M3UA Protocol ..................................................................34
SIP Protocol......................................................................35
H.323 Protocol Stack .........................................................38
SS7 Signaling ...................................................................39
V5.2 Protocol ....................................................................41
Recommendation Q.931 .....................................................44
R2 Signaling .....................................................................44
China No.1 Signaling..........................................................46
Overview
ZXSS10 SS1b is a protocol entity cluster. Based on various pro-
tocols, ZXSS10 SS1b interworks with various networks. The im-
plementation mode guarantees the system with excellent adapt-
ability of future technologies. If new technologies appear, upgrade
the software version and provide interfaces on the gateway at the
edge layer without the large-scale hardware alteration involved.
Protocols supported by ZXSS10 SS1b cover the following: MGCP,
MEGACO/H.248, SIP, ISUP/TUP, SNMP, M3UA, SCTP, H.323,
SCCP, TCAP, INAP, PRA signaling protocol and China No.1 signal-
ing protocol.
ZXSS10 SS1b is the control core in the SoftSwitch system.
Meanwhile, it works as the external interface of the whole system.
ZXSS10 SS1b supports multiple signaling protocols and provides
interaction with other networks such as the PSTN, H.323, IN and
SIP networks.
Moreover, as a NE in PSN, ZXSS10 SS1b provides the Ethernet
interface to connect with the data network.

Physical Interfaces
Following is the detail of interfaces.
� The 100Base-T Ethernet (RJ 45) interface serves as the inter-
face between ZXSS10 SS1b and the data network. It complies
with the Ethernet standard: IEEE802.3, IEEE 802.3u.
� The local maintenance and management interface shall use the
10/100BaseT auto-sensing interface.
� The 10/100/1000BaseT interface is used between ZXSS10
SS1b and the local charging server.
� The 10BaseT/100BaseT interface is used between it and the
NMS and charging center. The 10BaseT and 100BaseT inter-
faces shall comply with the IEEE802.3 and IEEE802.3u stan-
dards respectively.
Protocol Interfaces
ZXSS10 SS1b is a multi-protocol entity, which interacts and coor-
dinates with other NEs on the soft switching network via various
standard protocols (interfaces) to perform functions needed in the
system together.
ZXSS10 SS1b supports the following protocols:
1. Call processing protocols
ISUP, TUP over IP, SIP ,/SIP-T,SIP-I, H.323, V5.2, Q.931, R2,
R1, R1.5, R5, China No.1 signaling, PRI, etc
2. Transmission control protocols
TCP, UDP, SCTP, TCAP/SCCP/ M3UA, IUA, V5UA
3. Media control protocols
H.248, SIP, MGCP, NCS, H.323
4. Service application protocols
INAP, CAP, RADIUS, MAP, MAP+,SIP
5. Maintenance management protocols
SNMP, FTP, Telnet
Figure 6shows the typical application of main protocols.

Chapter 3 Interfaces and Protocols
FIGURE 6 TYPICAL APPLICATION OF MAIN PROTOCOLS
Figure description is as follow:

SG: Signaling Gateway
MG: Media Gateway
IAD: Integrated Access Device
H.323GW: H.323 Gateway
AAA: AAA server
� The SoftSwitch control equipment communicates with MG us-
ing H.248 protocol or MGCP.
� The SoftSwitch control equipment communicates with SG using
ISUP over IP.
� The SoftSwitch control equipment communicates with H.248
IAD using H.248 protocol.
� The SoftSwitch control equipment communicates with SIP in-
telligent terminals using SIP protocol.
� The SoftSwitch control equipment communicates with the Ap-
plication Server using SIP protocol or standard API interface.
� The SoftSwitch control equipment communicates with the AAA
Server using RADIUS protocol.
� The SoftSwitch control equipment communicates with Network
Management System (NMS) using SNMP protocol.
� The SoftSwitch control equipment communicates with H.323
gateway and gatekeeper using H.323 protocol.
� SIP/SIP-T/SIP-I is used between different SoftSwitch control
devices.
� The SoftSwitch control equipment communicates with SHLR
using IP-based MAP or MAP+ protocol. Or it communicates
with SHLR with MAP or MAP+ protocol through SG.

� Through MG, the SoftSwitch control equipment communicates

with the traditional switching network using common channel
signaling, associated signaling, V5 signaling or ISDN signaling.
� Through SG, SoftSwitch control equipment communicates with
the traditional SS7 network using the ISUP protocol.
� Through SG, the SoftSwitch control equipment communicates
with the traditional Intelligent Network (IN) using INAP or TCAP
protocol.
Protocol Overview
Following is the detail of different protocol.
Megaco/H.248 Protocol
Definition H.248/Media Gateway Control Protocol ( MEGACOProtocol), H.248
protocol for short, is a media gateway control protocol regulated
by IETF and ITU-T. This is a non-peer-to-peer protocol used for
the communication between the Media Gateway Controller (MGC)
and Media Gateway (MG).
The main function of the protocol is to establish a proper service
bearer connection model dividing the call connection and bearer
connection. By managing various service gateways, such as TG
and AG, the service interworking between the packet network and
PSTN network can be performed.
Connection Model The connection model defined by the H.248 protocol covers two
primary concepts: termination and context.
� Termination
Termination is a logical entity sending or accepting one or more
media streams. It has multiple feature descriptions, which
are integrated into a set of descriptors and contained in com-
mands. The termination is uniquely identified by the Termi-
nation ID, which is assigned by the media gateway when it is
built. At any time, a termination only belongs to one context.
� Context
The context indicates the connection relationship among some
terminations. It describes the topological relationship among
terminations and media mixed/switched parameters. A con-
text is composed of a group of terminations. One context can
contain multiple terminations.
Commands of The H.248 protocol provides commands to manipulate the logical
H.248 Protocol entities (terminations and contexts) of the connection model. The
attributes of terminations and contexts are under the control of
commands. In H.248, eight commands are defined, as shown in
Figure 7.

FIGURE 7 COMMANDS OF H.248 PROTOCOL
In Figure 7except the Notify and ServiceChange commands, the

rest is sent from MGC to MG, and MG responds. Notify is sent from
MG to MGC, while ServiceChange can be sent to and from each
other.Table 3lists functions of the eight commands.
TABLE 3 COMMANDS OF H.248 PROTOCOL
Command Function
Add This command is used to add a termination to a context.

When the Add command is used to add a termination to
an empty context, by default, a context is added.
Modify This command is used to modify the termination

attributes, events and signals.
Subtract This command is used to delete the association between

a termination and the related context, and feedback
the statistical features when the termination was in the
context.
Move This command is used to move a termination from one

context to another.
AuditValue This command is used to obtain the current information

of the termination attributes, events, signals, and
statistics.
AuditCap- This command is used to obtain all the possible values of

abilities the termination attributes, signals, and events.
Notify This command is used by the MG to notify the MGC of

events happened to the MG.
Service- This command is used by the MG to notify the MGC that

Change one or one group of termination is out of service or in
service again.

MGCP Protocol
Definition Media Gateway Control Protocol (MGCP) is the protocol put forward
based on the requirements of gateway separation structure. At the
very beginning, the protocol, developed on the basis of the Sim-
ple Gateway Control Protocol (SGCP) and Internet Protocol Device
Control (IPDC), is designed for the gateway controller to control
the gateway.
Connection Model According to the MGCP, the connection model comprises two basic
elements: endpoint and connection, a call supported by one or
more than one connection, and the connection and call are com-
pleted by the cooperation of one or more than one call agent. Fig-
ure 8shows the relationship among the endpoint, connection, call,
and gateway.
FIGURE 8 RELATIONSHIP AMONG ENDPOINT, CONNECTION, CALL,

AND GATEWAY
� Endpoint
The endpoint is the data source or data destination, physically
or virtually.
� Connection
The connection covers connections from point to point and
point to multi-point. The point-to-point connection means the
association between two endpoints sending data to each other.
Once the connection between two endpoints is established,
start to transfer the data.
� Call
A call, identified by a unique identifier, is created by the call
agent. A call identifier can be considered as a character string
without the structure. The call identifier must be unique in the
system. The call agent may establish more than one connec-
tion for the same call. These connections must be associated
with the same call.

Commands of Commands of MGCP are classified into connection processing com-

MGCP Protocol mands and endpoint processing commands, which cover nine com-
mands. There are nine kinds of MGCP messages happened be-
tween MGC and MG. When a message is sent to MG or MGC, the
message is called a command. When the acknowledgement mes-
sage of the command is returned from MG or MGC, this message
is called a response. Commands and responses are inseparable.
When a command is received, the gateway or MGC immediately
returns a response.Table 4 lists commands of the MGCP protocol.
TABLE 4 COMMANDS OF MGCP PROTOCOL
Command Function
EndpointConfi- This command is used to define the encoding

guration format of receiving signals by the endpoint, such as
A/u encoding.
NotifycationRe- This command is used to request the gateway to

quest send the notification, according to the specified
event on the current endpoint.
Notify This command is used by the gateway to send

the notification to the call agent, according to the
detected events that being requested.
CreateConnec- This command is used to establish the connection

tion between two endpoints. The following local
connection parameters are provided: Encoding
format, Signal packing cycle, Bandwidth, Service
type, Enable echo counteraction, mute suppression,
or Activity detection.
Modifyconnec- This command is used to change the connection

tion parameters, including the following:
1. Providing the information of the opposite end of
a connection
2. Changing the format parameters and activating
or deactivating a connection
3. Changing the send parameters of a connection
through changing the encoding format, packing
cycle, echo counteraction and so on
DeleteConnec- This command is used to terminate the connection

tion and collect the statistical information of connection
execution. Deleting a connection covers the
following situations:
1. The deletion originated by the call agent
2. The deletion originated by the gateway
3. The deletion of more than one connection
originated by the call agent
AuditEndpoint This command is used by the call agent to obtain

the status of specified endpoint.
AuditConnection This command is used by the call agent to obtain

parameters attached to a connection.
Restartinpro- This command is used by the gateway to notify the

gress endpoint of exiting service or obtaining service.

SIGTRAN Protocol
Definition SIGTRAN protocol is usually used between signaling gateways and
MGC, as shown in Figure 9. SIGTRAN protocol stack supports
transferring the SCN signaling protocol through the IP network.
This protocol stack supports the standard inter-layer primitive in-
terface defined by the layered model of the SCN signaling protocol.
In this sense, the existing SCN signaling application can be used
without any modification. Meanwhile, the protocol stack takes the
standard IP transfer protocol as the lowest transfer layer and en-
riches its functions to meet the special transfer requirements of
the SCN signaling.
FIGURE 9 APPLICATION MODEL OF SIGTRAN PROTOCOL
Structure of Figure 10shows the structure of the SIGTRAN protocol stack.

SIGTRAN Protocol
Stack FIGURE 10 STRUCTURE OF SIGTRAN PROTOCOL STACK
SIGTRAN consists of two layers: signaling transfer layer and sig-

naling adaptation layer.
� The signaling transfer layer used SCTP, providing reliable mes-
sage packet transmission in the IP network.
� The adaptation of protocols at each layer in the multi-layer
protocol model of SCN signaling is performed at the signaling
adaptation layer, responsible for inter-layer primitive transfer
between SCN and IP networks.

SCTP Protocol
Definition Stream Control Transmission Protocol (SCTP), a reliable connec-
tion-oriented transmission protocol located at the same layer with
TCP and UDP in the network model, is designed for transferring
the PSTN signaling message in the IP network.
Advantages of Compared with TCP, SCTP has the following advantages:
SCTP Compared
with TCP � The SCTP connection can direct at multiple addresses, while
the TCP connection generally leads to one address.
� TCP is based on the bit stream, while SCTP is based on the
subscriber message stream.
� One TCP connection only supports one stream, while one SCTP
connection supports more than one stream at the same time.
� SCTP is configured with instruments for prevent attacks, such
as Tag, Cookie, and MAC.
SCTP Terms � Coupling
The SCTP coupling, in fact the corresponding relationship be-
tween two SCTP endpoints, covers two SCTP endpoints and
protocol status information containing the authentication tag
and transfer sequence numbers. A coupling can be uniquely
identified by the transfer address used by the SCTP endpoint
that uses the coupling. At any time, there is only one coupling
existed between two SCTP endpoints.
� Endpoint
The SCTP endpoint is the logic receiving or sending side in
the SCTP packet. On a multi-home host, an SCTP endpoint
can be presented by the opposite host as a group of qualified
destination transfer addresses that the SCTP packet can be
sent to, or a group of qualified source transfer addresses that
can receive the SCTP packet. All the transfer addresses used
by the SCTP endpoint must adopt the same port number, but
multiple IP addresses can be used. The transfer address used
by the SCTP endpoint must be unique.
� Stream
A stream is a unidirectional logic channel established from two
SCTP endpoints. With regard to the sequential delivery service,
all the subscriber messages in the channel must be delivered
in order.
� Transfer address
The transfer address is defined based on the network layer ad-
dress, transfer layer protocol, and transfer layer port number.
When the SCTP runs on the IP network, the transfer address is
defined by combining the IP address and the SCTP port num-
ber. At this moment, SCTP works as the transfer protocol.

M3UA Protocol
Definition MTP3-User Adaptation Layer (M3UA) is the MTP3 adaptation pro-
tocol. Through the seamless cooperation of M3UA and MTP3, SS7
network gracefully extends from the SCNN network to the IP net-
work. In this sense, devices in the IP network without functions
of the SS7 physical layer, data link layer, and complete network
layer, can provide services for the subscriber part of SS7.
Location of M3UA M3UA uses the SCTP protocol as the reliable signaling transfer pro-
tocol at a lower layer. Figure 11shows the location of M3UA when
it is being applied.
FIGURE 11 LOCATION OF M3UA
M3UA Terms � Application Server (AS)

It is a logic concept. An AS is the endpoint that performs a
service in the network. An AS always serves a specified routing
key.
� Application Server Process (ASP)
It is a process case on the AS. Multiple ASPs can serve one AS
in the active-standby mode or load balancing mode. One ASP
can also serve multiple AS.
� Routing Key
Each routing key on the gateway defines a set of routing fea-
tures of SS7 message. The SS7 message that meets these
features are processed by specified AS.
� Routing Context
It is the parameter of ASPM message between the gateway
and ASP. This parameter uniquely identifies the SS7 message
related to the specified AS, that is, strictly corresponding to
the routing key or AS.

SIP Protocol
Definition Session Initiation Protocol (SIP), a multimedia communication pro-
tocol defined by IETF, is a application layer control protocol based
on texts and independent of the bottom protocols, used for estab-
lish, modify and terminate the two-party or multi-party multimedia
session over the IP network.
The Client-Server mode based on the text format is adopted in
SIP, which indicates the message syntax, semantics and code in
the text form. In this case, the Client originates requests and the
Server gives responses.
In the SoftSwitch network, SIP is applied among the SoftSwitch
device, SIP intelligent terminal and SIP server.
SIP-I is adopted among the SoftSwitch devices.
Primary Network Figure 12shows the primary network model of SIP
Model of SIP
FIGURE 12 PRIMARY NETWORK MODEL OF SIP
1. User Agents (UA)

As an entity that originates and terminates the session, UA
covers two functional entities
� User Agent Clients (UAC), the functional entity that origi-
nates the SIP transaction request.
� User Agent Server (UAS), the functional entity that receives
the SIP transaction request.
Both UAC and UAS can terminate a call.
2. Proxy Server
� Relates to Redirect Server and Proxy Server
� It works as an agent for other clients to transit and forward
the SIP messages. The message mechanism is similar to
that of UAC and UAS.
� It translates and processes the received request message
and then transfers it to other servers.
� It routes the SIP request and response.
3. Location Server

The location server is a database. It is used to store the current

location information of the terminal subscribers, and provide
the SIP Redirect Server or Proxy Server with possible location
information of the called party.
4. Redirect Server
� It connects with location server using LDAP protocol.
� It returns the new location of the called to the caller. The
caller can recall based on the new location.
� Different from the proxy server, the redirect server cannot
originate its own call.
� Different from the user agent, the redirect server does not
accept the call termination or terminate the call on its own
initiative.
5. Registrar Server
� It accepts the REGISTER request to complete the registra-
tion of the subscriber address.
� It supports the authentication function.
SIP Message SIP adopts the text code mode, whose message falls into request
message and response message. Each message contains message
header and message body.
1. Request message
The SIP message has six basic request messages, which are
extended in development, including Updata and Message.
Here come the six basic request messages:
� Invite
It is the message initially originated by the calling party,
indicating the capability of the calling party and inviting the
called party to participate in a session.
� Ack
After the called party gives a final response to the invite
message, the calling party will send the Ack message, in-
dicating that the final response is received.
� Options
When originating a call request, the calling party can query
the capability of the proxy server or the capability of the
called client through the Options message.
� Bye
It indicates terminating a session.
� Cancel
It is used to cancel an ongoing session request which does
not obtain the final response from the called party yet.
� Register
It is the registration information sent by the user end when
registering to the network.
2. Response message

After the server receives the request message from the client,
it will generate a series of response messages, indicating the
processing situation to the request. The response message
falls into six types.
� 1××: Provisional, indicating that the request has been re-
ceived and is being processed.
� 2××: Success, indicating that the action has been received
and it is understood and accepted.
� 3××: Redirection, indicating that is necessary to perform
the next action, usually, redirection.
� 4××: Client Error, indicating that the request has a syntax
error or it cannot be executed by the server.
� 5××: Server Error, indicating that the server cannot exe-
cute the request but the request is a valid request.
� 6××: Global Failure, indicating that all servers in the net-
work cannot execute the request.
In SIP, 1×× response is defined as a temporary response and
other five responses as a final response. When receiving a final
response, the calling party will send an Ack message to make con-
firmation. When receiving a temporary response, the calling party
can send a Cancel message to cancel the former message.
Message Format Usually, both request message and response message have mes-
sage header and message body, but some messages do not have
the message body, such as 100trying.
1. Message header
The message header contains the following:
� From: Address of the party originating the request
� To: Address of the party receiving the request
� Call-ID: It is a parameter to identify some request. Usually,
this parameter will not change in a call. Call-ID, From and
To ensures the uniqueness of a call together.
� Cseq: It is a parameter indicating transaction. This param-
eter ensures the sequence of different request messages in
a call.
� Contact: It notifies the peer end of the address to which
the next request message can be sent directly.
� Via: When the request message passes the proxy server,
the proxy server adds its address into the domain, indi-
cating the chosen route. Thus, the response message can
return to the calling party along the route.
� Content-type: It indicates the format of the message body,
for example, SDP or ISUP.
� Date: It indicates the time, which may include year, month,
date, hour, minute, second, and day of a week.
2. Message body
In SIP, the message body usually adopts SDP. SDP is in a pure
text format, so it can be embedded into SIP.
SIP-T has the SDP+ISUP message body. For the processing of
ISUP, keep the binary coding format of ISUP.

H.323 Protocol Stack

Definition The H.323 protocol a standard real-time multimedia communica-
tion protocol stack developed by theITU-T for the IP network. It
consists of a series of protocols such as call control, media coding,
management control and network security.
H.323, transferred at the bottom layer without QoS guarantee, is
applied to PSN to meet the requirements for performing the mul-
timedia communication. The main purpose is to implement the
audio-video communication among terminals over different net-
works.
H.323 Terms The H.323 protocol stack defines the components, protocols, and
procedures for providing multimedia communication in the net-
works including the IP network based on the packet switching.
H.323 defines four components: Terminal, Gateway, Gatekeeper,
and MCU. Using these four components, H.323 can support the
point-to-point or point-to-multipoint audio, video, and data com-
munication.
� Terminal
On the IP-based network, Terminal is a client. It should support
the following three functions:
� Signaling and control
These support H.245 (a complex protocol for channel usage
and channel capacity), H.225 (a call signaling receiving,
sending, and establishing protocol similar to Q.931), and
RAS (defined on H.225 and used for the terminal and GK
communication protocol).
� Real-time communication
It supports RTP/RTCP (a protocol that handles the audio
and video information packet in order)
� Encoding
It supports the compression before transmission and de-
compression after receiving. The encoding is inter-opera-
ble. Each H.323 terminal should support Q.711, and other
encoding formats are available options.
� Gateway (GW)
It provides the connection between packet switching networks
and circuit switching networks.
� Gatekeeper (GK)
It is optional in the H.323 system. If they exist, they have
some mandatory functions. GK performs four mandatory func-
tions covering the address translation, connection admission
control, bandwidth control, and domain management. More-
over, GK performs four optional functions covering the call
control signaling, call authentication, bandwidth management,
and call management. When there is the GK in the H.323 sys-
tem, before establishing a call, all the terminal users need to
register on the GK and obtain its permission.
� Multipoint Control Unit (MCU)

MCU supports more than three terminal users in performing

sessions. A typical MCU comprises a multipoint controller (MC)
and multiple (or none) multipoint processor (MP). MC performs
the control functions, such as the negotiation between termi-
nals, deciding the shared capacity for handling voice or video.
MP processes media streams in the session, such as voice mix
and voice/video switch.
SS7 Signaling
Overview NO.7 signaling is a common channel signaling. Multiple voice
channels share a signaling link. All the signaling links compose
the NO.7 signaling network.
Composition of NO.7 signaling network contains the following three elements:
No.7 Signaling
� Signaling Link
� Signaling Point (SP): is the node that can generate signaling
message.
� Signaling Transfer Point (STP): is in change of transferring the
received signaling messages to other nodes.
The node can be a SP or STP, or both.
3-Grade Structure Figure 13shows the 3-grade structure of No.7 signaling network.
of No.7 Signaling
FIGURE 13 3-GRADE STRUCTURE OF NO.7 SIGNALING NETWORK
� The 3-grade structure covers HSTP, LSTP, and SP.

� HSTP and LSTP are netty networks.
� HSTPs exist on the A and B planes. They are interconnected
in pairs.
� LSTP is connected to two HSTPs on the A and B planes. SP is
connected to two LSTPs.

No.7 Concepts The following are several concepts of No.7 signaling network.
� SP: is the start point or end point of the No.7 signaling mes-
sage.
� SP code: is the unique code to locate the SP in the signaling
network. It adopts the 14-bit or 24-bit code.
� STP: is the network node that completes the No.7 signaling
message transfer.
� Signaling link: is the No.7 data link connecting SP and STP.
� Signaling link set: is a set of parallel signaling links directly
interconnecting two or more STPs.
� Destination (signaling) Point Code (DPC): indicates the trans-
mission destination of the No.7 message. In our country, the
24-bit code is used.
� Originating (signaling) Point Code (OPC): indicates the trans-
mission source of the No.7 message. In our country, the 24-bit
code is used.
� Circuit Identification Code (CIC): specifies the trunk circuit oc-
cupied by the No.7 message to establish a connection. The
12-bit code is used.
� Signaling Link Code (SLC): indicates the code of the link be-
tween two neighboring SPs. The 4-bit code is used.
� Signaling Link Selection Code (SLS): indicates the link used by
the No.7 message. The lower 4-bit code of CIC is used.
Function model of Figure 14shows the function model of No.7 signaling.
No.7 Signaling
FIGURE 14 FUNCTION MODEL OF NO.7 SIGNALING
1. MTP1 is the signaling data link layer equivalent to L1 physical

layer of OSI. It works as the bi-directional transfer channel of

the data, including data transfer channel and signaling terminal

equipment. The data transfer channel uses 64kb/s.
2. MTP2 is the signaling link layer equivalent to L2 link layer, which
completed on the COMM board of the ZXJ10. MTP2 provides
the signaling link for the reliable signaling message transport
between two directly connected SPs. The functions of MTP2
cover the pegging and locating of the signaling unit, error de-
tection and correction, signaling link monitoring, and traffic
control.
3. MTP3 is a function of the signaling network. It is combined with
SCCP to the third function layer of OSI. At this layer, signaling
message processing and signaling network management are
performed.
� Signaling message processing covers message route selection,
message identification, and message allocation.
� Signaling network management is classified into signaling
traffic management, signaling link management, and signaling
routing management.
Signaling traffic management is responsible for the traffic con-
trol and the transfer under the fault or congestion. Signaling
link management covers resuming, starting, and exiting the
signaling link in the signaling network. Signaling route man-
agement is in charge of reliably switching messages on the
signaling route between SPs, including permitting, restricting,
forbidding, and transferring messages.
� Signaling Connection Control Part (SCCP)
� Telephone User Part (TUP)
� ISDN User Part (ISUP)
� Transaction Capability Application Part (TCAP)
� Operation and Maintenance Application Part (OMAP)
� Intelligent Network Application Part (INAP)
� Mobile Application Part (MAP)
V5.2 Protocol
Definition The V5.2 protocol is used for interconnection and interworking via
Signaling Gateway (SG) between the SoftSwitch network and the
access network that has the V5.2 interface. In the SoftSwitch sys-
tem, the Media Gateway (MG) can be used as an access device.
TheSG is connected to the SoftSwitch control device via V5UA.

V5 Interface Figure 15and Figure 16show the V5 interface functions.

Functions
FIGURE 15 FUNCTIONS OF V5 PROTOCOL
FIGURE 16 FUNCTIONS OF V5.2 INTERFACE
1. V5 interface Layer 1 (physical layer)

It provides the physical channel for information transmission.
2. Layer 2 of V5 interface (data link layer)

It provides reliable point-to-point transmission and an ideal

error-free path to the upper layer.
3. Layer 3 of the V5 interface (application layer)
The L3 protocols of the V5 interface include the PSTN proto-
col, control protocol (public control protocol and port control
protocol), link control protocol, BCC protocol and protection
protocol, of which the link control, BCC and protection proto-
cols are used only for V5.2 interfaces.
Related Protocols 1. PSTN protocol
The PSTN protocol of the V5 interface provides the procedures
related to the specific analog calls.
2. Control protocol
The control protocol of the V5 interface consists of two parts:
Common control and user port control.
The status indications of PSTN and ISDN user ports are based
on the principle of separate responsibilities of AN and LE. Only
the user port status related to call control can pass through the
V5 interface and influence the state machine in the LE.
In the working state, the LE uses the activation/deactivation
procedures specified in Recommendation I.430.
For ISDN-BA, additional states are required in the AN to:
� Support port activation of the permanent line (PL capabil-
ity) service;
� Maintain the digital section (DS) and user ports;
� Implement optional activation control as defined in ITU-T
Recommendation G.960.
3. Link control protocol
In the V5.2 interface, the following functions and requirements
are needed for each independent 2048 kb/s link:
� Layer 1 link status and relevant link identification of the
2048kb/s link;
� Blocking and unblocking of Layer 1 links through manage-
ment;
� Verification of the link consistency through link identifica-
tion;
� Coordination of the link control functions;
� Providing the link control protocol for the communication
between AN and LE so as to coordinate these functions at
both sides.
4. BCC protocol
The V5.2 BCC protocol provides LE with a means to request AN
to set up and release connections between the AN user port
and the specified V5.2 interface time slot. The BCC protocol
enables the V5.2 interface bearer channel to be allocated or
de-allocated by independent processes.
5. Protection protocol

The protection mechanism protects all active C channels as well

as the C path (itself) of the protection protocol used to control
the protection switching program.
The protection protocol does not protect the bearer channel
but allows re-configuration of the bearer channel in the case of
failure of its associated 2048 kb/s link.
Logical C channels will not be protected if there is only one
2048 kb/s link.
Recommendation Q.931
Definition Recommendation Q.931 provides the procedures for establishing,
holding and clearing the network connections between ISDN sub-
scribers and the network. These procedures comprise the message
transmission interface structures at the basic rate and the primary
rate on the D channel. The related protocol functions, methods
and interfaces with the other layers are included in Recommenda-
tion Q.930.
The content in the Recommendation is used for the control of
circuit switched connections, user-user signaling connections and
packet switched connections.
Functions of Recommendation Q.930/Q.931 provides the following functions:
Q.930/Q.931
� Processes Layer 3 messages, communicates with the call con-
trol and resource management entities of the switch and com-
pletes call processing.
� Communicates with the adjacent layers via primitives.
� Implements resource management (such as channel and call
reference) as necessary.
� Provides the required basic services and supplementary ser-
vices for users.
R2 Signaling
Definition R2 signaling comprises of circuit signal and register signal. The
definitions of circuit signal and register signal vary from country
to country.
1. Circuit signals, transported among circuit devices (repeater)
comprise of multiple circuit monitoring signals, monitoring
trunk status and connection execution control. Since each
trunk shall be equipped with a suite of circuit devices without
the global utilization, therefore, to reduce the cost, circuit
signals are simple and comparatively and the types are lesser.
2. Register signals, transported among registers, comprise of
routing signals and service signals, used for selecting routing,
callee and managing telephone network. Since registers are
public devices and the total amount is few, therefore, register
signals can be more complicated and the types are more.

Circuit Signal Circuit signals of R2 signaling are in three forms: DC circuit sig-
nal, in-band single frequency pulse circuit signal and digital circuit
signal.
1. DC circuit signal
DC circuit signal is mainly used over the solid line trunk of
electromechanical exchange. At present, the domestic local
call network uses the SPC switches; therefore, the DC circuit
signal is not used now. No details on DC circuit signal are given
here.
2. In-band single frequency pulse circuit signal
In the automatic toll call network, when the inter-office trans-
mission system uses carrier, microwave, satellite circuit of Fre-
quency Division Multiplexing (FDM), the inter-office circuit sig-
nals use audio signal, that is, in-band single frequency pulse
circuit signal.
3. Digital circuit signal
Circuit signals are used for monitor the seizure, release and
block state of trunks. Meanings of digital circuit signals are
listed in Table 5.
TABLE 5 MEANING OF DIGITAL LINE SIGNAL
Forward Signal Backward Signal
Code Meaning Code Meaning
Office status af=0 Offhook Onhook ab=0 Offhook

of calling state state of the state of the
party (seizure) called party called party
of the
calling
party
af=1 Onhook ab=1 Onhook

state (dis- state (dis-
connec- connection
tion) of backwards)
the call- of the called
ing party party
Status bf=0 Normal Office bb=0 Idle state

(Fault) status of
bf=1 Faulty called party bb=1 Seized or
blocked
state
Re-ringing cf=0 Re-ringing Operator cb=0 Operator

or forced or forced response response
clear of the clear ringing ringing
operator of the
operator
cf=1 No cb=1 No operator

re-ringing response
or forced ringing
clear
of the
operator

Register Signal The Multi Frequency Compelled (MFC) mode is adopted for register
signals of R2 signaling. The signals fall into two kinds: forward sig-
nals and backward signals. Both of them are successive. Forward
signals are used for transferring addresses and control instruc-
tions, while backward signals are used for acknowledgement and
control. When a digit is sent, the forward signal can stop sending
unless the backward acknowledgement is received. Similarly, the
receiving end stops backward signals unless the forward signals
are detected stopped sending.
MFC register signals fall into forward signals and backward signals.
Here, forward signals are divided into two groups: Group I and
Group II. Backward signals are divided into another two groups:
Group A and Group B. Signals in Group A are acknowledgement
signals of Group I, while signals in Group B are acknowledgement
signals of Group II. Meanings of each signal group are listed in
Table 6.
TABLE 6 MEANING OF SIGNALS
Forward Signal
Group Name Meaning Capacity
I KA Caller type 10/15
KC Long-distance connection 5
type
KE Long-distance (local) 5
connection type
Digital Digit 1~0 10

signal
II KD Call service property of 6

originator
Backward Signal
A A signal Loopback control 6

acknowledgement of
digit collection status
and connection status
B B signal Callee status 9
China No.1 Signaling

Definition The China No.1 signaling is the CAS signaling. Based on the signal
contents, the signals can be divided into circuit signals and register
signals.
� Circuit signals are transported among circuit devices. Most of
them are circuit monitoring signals.

� Register signals are transported among registers. Registers

are publicly used and types of signals transported by register
are more compared with circuit signals.
Circuit Signal The circuit signal can be presented in three forms:
� DC circuit signal
DC circuit signal is mainly used over the solid line trunk.
� In-band single frequency pulse circuit signal
This kind signal mainly used over the carrier circuit. It is ap-
plied to the FDM circuit devices.
� Digital circuit signal
Digital circuit signal mainly used over the PCM TD transmission
circuit. It is transferred via TS16 of TDM system frame struc-
ture. From one through third digits are used. The signals fall
into forward signals and backward signals. Byte meanings of
the circuit signal are listed in Table 7.
TABLE 7 MEANING OF NO.1 SIGNALING LINE SIGNAL
Af 0 Offhook (seizure)
1 Onhook (disconnection)
Bf 0 Normal
1 Faulty
Cf 0 Re-ringing or forced clear of the

operator
1 No re-ringing or forced clear of the

operator
Ab 0 Called offhook
1 Callee onhook
Bb 0 Idle
1 Seized or blocked state
Cb 0 Operator response ringing or call

reaches called
1 No operator response ringing or call

reaches called
Register Signal MFC signal: the register signal is sent out by a register in an ex-
change office and received by a register of another exchange of-
fice. It is used to automatically connect the control circuit. To
guarantee the fast transfer and anti-interference capability, the
register adopts the MFC mode. Therefore, the signal is called MFC
signal.
MFC signals fall into the following kinds:
� Forward signals: Group I and Group II

� Backward signals: the response backward signals fall into

Group A and Group B.
Group I and Group A are in pair, while Group II and Group B are
in pair.
Meanings of register signals are listed in Table 8.
TABLE 8 MEANING OF REGISTER SIGNAL
Forward Signal
I KA Caller type 15
KC Long-distance connection 5
type
KE Long-distance (local) 5
connection type
Digital Digit 1~0 10

signal
II KD Call service property of 6

originator
Backward Signal
A A signal Loopback control acknowl- 6

edgement of digit collection
status and connection status
B B signal Callee status 6

Chapter 4
Service Provision
Table of Contents
Overview..........................................................................49
Primary Voice Services .......................................................51
Primary PSTN Voice Services and Supplementary Ser-
vices................................................................................51
IP Centrex ........................................................................57
IP Public Phone .................................................................63
Traditional Intelligent Services.............................................66
F3G Intelligent Services .....................................................68
Hybrid Number Assignment Service .....................................69
WAN Centrex Services........................................................70
Number Portability (NP)......................................................71
Pre-paid Service ................................................................71
PHS Hotel Service..............................................................72
Callee CRBT ......................................................................72
Caller CRBT ......................................................................72
Non-Personalized CRBT ......................................................73
One Number Multiple-Phone................................................73
IP-based Value-Added Service .............................................74
SoftDA Service ..................................................................76
Multimedia Service ............................................................77
Web Call Service ...............................................................78
Web 800 Service ...............................................................78
Web Conference Service .....................................................80
Unified Messaging Service (UMS) .........................................82
Overview
ZXSS10 SS1b has the powerful service provision capability. The
system uses the multi-layer network architecture, as shown in Fig-
ure 17. It has the following features:
� The service provisioning is independent of network and media.
Once a service is generated, it can be shared by the whole
network.
� When a new service is introduced by the operator, the resilient
creation environment of service logic help the operator free of
expensive hardware upgrade cost along with other techniques
and transportation cost.
� Services can be resilient provided by developers, operators,
software companies licensed by operators and the third-party
developers.

� Compared with traditional service provisioning, operators can

introduce new services much more quickly with lower cost.
FIGURE 17 SERVICE PROVISIONING MODE OF SOFTSWITCH

NETWORK
Following is the description of above figure.

� SG: Signaling Gateway
� MG: Media Gateway
� Parlay GW: Parlay Gateway
� IAD: Integrated Access Device
ZXSS10 SS1b supports four service provisioning methods:
1. ZXSS10 SS1b directly provides primary PSTN services, sup-
plementary services, IP Centrex services, point-to-point video
and multi-party video services.
2. Interconnected with the existing Service Control Point (SCP),
ZXSS10 SS1b implements traditional intelligent services.
The SoftSwitch control equipment interconnects with the SCP
by two means:
i. Interconnect with the SCP by means of standard No.7
through SG.
ii. Based on the TCP/IP connection, interconnect with the SCP
by means of extended INAP.
3. Through support of the application server, the IP based Value
Added Services (VAS), intelligent services and a variety of mul-
timedia services are implemented.
Besides, A WEB Server is added to the network for service
registration and selection platform of IP users and as the ser-
vice user window. In specific applications, the standard service
API interface of the application server can be provided to the
third-party attendants, who keep developing diversified new
services for users. The application server can also be leased to

Chapter 4 Service Provision
enterprise group users to provide them with customized ser-

vices or to network attendants as VAS platform.
4. VIP customer (VIC) system
VIC attendant console system can provide the access of multi-
ple services, such as the VIP voice, data, and stream media. In
this way, VIP customer system integrates functions of various
consoles and adds some new functions. VIP customer system
is mainly used by group users, such as hotels, schools, facto-
ries, and government offices.
Primary Voice Services

The primary voice service refers to the service of which the major
purpose is to ensure normal conversation between subscribers. It
is a service collection instead of a specific service. It includes voice
services such as primary call services, primary PSTN voice ser-
vices, supplementary services and traditional intelligent services.
Here, primary call services refer to local, national and international
automatic incoming call and outgoing call services of various termi-
nals. The terminal type includes ordinary phone accessed via the
IAD, Soft Phone and various IP Phones based on H.248, MGCP,
NCS, SIP and H.323. Primary call services are provided by the
SoftSwitch control equipment independently.
Primary PSTN Voice

Services and Supplementary
Services
ZXSS10 SS1b can provide all primary PSTN voice services and sup-
plementary services specified in YDN 065-1997 General Technical
Specifications for Telephone Switching Equipment by the MPT and
its supplementary specifications. The supplementary services are
shown in Table 9.
TABLE 9 SUPPLEMENTARY SERVICES
Service Description
Call for- Automatically forward calls to a phone to the temporarily

warding specified phone set. According to the actual needs of the
subscriber, four services can be provided: call forwarding
on busy, call forwarding on no reply, call forwarding
unconditional (forwarding of all calls) and call forwarding
on subscriber offline.

Service Description
Toll pro- After registering call forwarding, a subscriber can use this
hibited in service so that calls are not forwarded to toll numbers.
call for- This service needs to be used in conjunction with call
warding forwarding.
Forward- After registering call forwarding, a subscriber can use this

ing of in- service so that incoming calls other than tolls are forwarded
coming while incoming tolls are intercepted. This service needs to
toll pro- be used in conjunction with call forwarding.
hibited
Call When a subscriber calls another and the called party

parking is busy, the call is recorded. Within 20 minutes, if
on busy the subscriber wants to call that subscriber again, the
subscriber just needs to pick up the phone and wait for 5
seconds without dialing any number.
Calling When the called party is busy, the caller can hang up
back on and wait. Once the phone of the called party is idle, an
busy automatic callback can be made.
Call It is permitted that a subscriber in conversation is notified

waiting of an incoming call. The subscriber can select to accept or
ignore this waiting call. After accepting it, the subscriber
can switch conversation between these two callers. In this
case, one caller will be held to listen to the waiting music.
The subscriber can also disconnect a call and concentrate
only on the other.
Absent When there is an incoming call but the subscriber is out,

sub- the exchange can provide voice service to reply the call for
scriber the subscriber so as to avoid repeated dialing of the caller.
service
Do not When the subscriber does not want to be interrupted by

disturb any incoming call, the exchange can reply all incoming calls
without affecting his outgoing calls.
Do not With this service, the subscriber can be exempt from the
disturb interference of outside calls within 12 hours (adjustable
within by the equipment buyer). Moreover, outgoing calls of the
the set subscriber will not be restricted.
time
Malicious When being annoyed by an unknown malicious call, the

call subscriber can query the number of the malicious call via
tracing the exchange simply by key pressing.
Instant It is supplementary for malicious call tracing. With this

mali- service, all incoming calls will be recorded in the alarms of
cious call the equipment buyer.
tracing
Multiple The analog subscriber is provided with multiple numbers

DNs for just one phone. In addition, there are different rings
for different numbers.
Transfer When a subscriber is in conversation with the called party,

service the called party can be held and another subscriber can
be dialed so that the latter two subscribers can converse
with each other.

Service Description
Three- When two subscribers are in conversation, another

way con- subscriber can be dialed without interrupting the
nection conversation between these two subscribers. Thus,
the three-party conversation or conversation with two
subscribers respectively is available.
Confer- The SoftSwitch control equipment provides the service of

ence call simultaneous conversation of more than three parties.
When the telephone conference service is used, the chair
subscriber can dial multiple subscribers (60 at most)
continuously so that they can participate in the conference
by pressing the hook.
Virtual In the data of the SoftSwitch control equipment, virtual

confer- conference access codes can be set, including the
ence password, maximum access number, start and end time.
When such a conference is used, subscriber A dials the
virtual conference number and inputs the password when
prompted to create a virtual conference room. Then, the
subscriber is notified to be the first in the conference
room through voice. Subscriber B picks up the phone,
dials the virtual conference number after hearing the dial
tone, and inputs the password as prompted for access to
the conference room. Then, he can hear the notification
that he is already in the conference room. Subscriber A
and subscriber B can converse with each other. In this
case, other subscribers can dial the virtual conference
number to enter the conference room and participate in
the conference.
IP Refer to the IP Centrex topic for details.

Centrex
Abbrevi- It is a service that uses a 1–2 bytes user-defined code to

ated di- substitute for a telephone number with many digits. The
aling service can be used to dial either a local number or a
national/international toll number.
Caller In the case of an incoming call, the LCD of the called

identifi- party’s phone can automatically display and store the
cation incoming number.
Call- It is prohibited to display the phone number of the

ing line subscriber for the called party. The priority of this service
identi- is higher than the calling number display service.
fication
restric-
tion
Immedi- The subscriber can set a frequently-used phone number as

ate hot- the hotline (only one hotline is permitted). After application
line of this service, the subscriber will be automatically
connected to the fixed number upon hooking off.
Delayed The subscriber can set a frequently-used phone number

hotline as the hotline (only one hotline is permitted). With this
service, the subscriber will be automatically connected to
the phone number if not dialing any number within the
specified time (5 seconds) after hook-off.

Service Description
Call With this service, if the called party has registered do

override not disturb, absent subscriber or call forwarding service,
normal ringing is available too.
Emer- This function permits emergency calls on the call restricted

gency phone set. The service is valid for all subscribers and
call over- requires no special application.
ride
Sec- After subscriber A activates this service, all incoming

retary calls to subscriber A will be first connected to another
service specified subscriber B. After being filtered by subscriber B,
the calls can be forwarded to subscriber A again through
transfer or handled by subscriber B directly. Subscriber B
can correspond to multiple subscriber As. Subscriber B
provides the override function for the secretary service of
subscriber A, that is, calls from subscriber B to A will be
directly connected to subscriber A.
Alarm The phone set can ring automatically according to the

clock preset time.
service
Missing If a subscriber has registered the missing number service,

number and there is any missing call, the subscriber will hear "You
have XX missing calls. Please dial #*99 for information"
after hook-off.
Forced With this service, the phone set of the subscriber will have
release the forced release function.
service
Auto- This service should be used together with outgoing call

matic restriction. After applying for outgoing call restriction, the
locking subscriber needs only dial *#66* S1S2S3S4* plus the
in out- outgoing number to dial the restricted numbers. He need
going not dial #54*KS1S2S3S4# to cancel the call restriction.
call re- The call restriction is still valid after hook-off.
striction
Outgoing This service is used to restrict some outgoing calls on this

call re- phone. It enables the subscriber to restrict some specific
striction outgoing calls of the phone with some simple operations
when leaving the phone set so as to avoid unauthorized
use of the phone.
Restric- The incoming calls of the subscriber with destination code

tion over restriction activated are not restricted. A maximum of
specified ten destination codes can be defined in the restriction
called list of each subscriber. The destination code can be the
destina- prefix, country code, area code, office code, special service
tion code number or DN (with the maximum length of 40 digits). The
emergency calls of the subscriber will not be restricted.
The proportion of such subscribers is 10%.

Service Description
Connec- This service allows a subscriber to call only specified

tion of called destination codes, and his incoming calls will not be
specified restricted. Up to 10 destination codes can be defined in
called the destination code list. The destination code can be of
destina- prefix, country code, area code, office code, special service
tion code number or DN (with the maximum length of 40 digits). The
emergency calls of the subscriber will not be affected. The
percentage of the subscribers with such service is 10%.
Restric- This service allows a subscriber to restrict the incoming

tion to calls of specified calling destination codes, but his outgoing
specified calls will not be restricted. Up to 10 destination codes
calling can be defined in the calling destination code list. The
destina- destination code can be of prefix, country code, area
tion code code, office code, special service number or DN (with the
maximum length of 40 digits). The emergency calls of the
subscriber will not be restricted. The percentage of such
subscribers is 10%.
Connec- This service restricts the incoming calls to only the specified
tion of numbers, and the outgoing calls of the subscriber are not
specified be restricted. Up to 10 destination codes can be defined
calling in the list of designated destination code connections.
destina- The destination codes may be prefix, country code, area
tion code code, office code, special service number or DN (with the
maximum length of 40 digits). The emergency calls of
subscribers will not be restricted by the call destination
connection service. The percentage of such subscribers
is 10%.
Overrid- Subscribers in need can apply to the exchange carrier

ing sub- for this service in advance. With this service, a service
scriber reservation flag is set in the background database, which
functions for caller identification and malicious call tracing.
Voice an- Subscribers in need can apply to the exchange carrier

nounce- for this service in advance. With this service, a service
ment for reservation flag is set in the background database. When
failed a call fails, the corresponding failure voice can be sent to
calls the subscriber.
Not in A service reservation flag is set in the background

service database. Any subscriber dialing this number will hear a
voice prompt.
Number A service reservation flag is set in the background database

reserva- of the exchange. Any subscriber dialing this number will
tion after hear the voice prompt: "Sorry, the subscriber has applied
tempo- for number reservation after temporary release".
rary re-
lease
Call filter Based on the calling number or incoming trunk identifier,

with certain callers or calls from a certain incoming trunk can
black/whitebe prohibited or allowed to go to certain destination codes.
list A destination code may be of prefix, national code, area
code, exchange number, special service number or DN. The
maximum length is 20 digits.

Service Description
Setting With this service, the subscriber can be activated only

as com- after the telecom exchange assigns the subscriber with
puter at- the corresponding rights. Independent use of this service
tendant is meaningless.
console
Authen- 1. Authenticate calls of subscribers under a carrier. The

tication authentication content includes authorities in using the
function network and various services.
2. Authenticate calls from networks of other carriers so
as to judge whether such subscribers have the right
to use services of this network, including the national
full automatic service and international full automatic
service or other services of this network.
Inter- 1. 1. For local subscribers, intercept prohibited calls

ception and send the corresponding recorded announcements
function according to subscriber attributes and the dialed
number.
2. Intercept inter-network calls after checking the
black/white list.
3. Intercept calls after authentication.
4. Intercept incoming calls with incomplete calling
numbers or without calling numbers.
Polarity Notify the gateway and IAD to provide the corresponding

reversal polarity reversal pulse for public phone billing.
function
Inter- Intercept the call when the called number is changed, the
ception route is blocked temporarily, or the subscriber misoperates.
service Transfer the call to an answer-agent device to reduce
unnecessary connections. This service is free of charge and
requires no special application.
## Au- One can hear the number of the subscriber reported by the
tomatic system through dialing ##.
number
report
117 Time announcement

time an-
nounce-
ment
Anon- Based on the national standards, to perform three kinds

ymous of caller identification display restriction, the SoftSwitch
call classifies the anonymous calls into three:
A (temporary display restriction): The caller has the right
to restrict the identification display via a certain operation
when originating a call. Generally, the caller identification
display is permitted.
B (temporary display permission): The caller has the right
to permit the identification display via a certain operation
when originating a call. Generally, the caller identification
display is restricted.
C (permanent display restriction): The caller identification
display is permanently restricted. If the called is a override
user, the called can identify the caller regardless of any
kind of caller identification display restriction.

Service Description
Choice For preset subscribers, the SoftSwitch control equipment

of carrier provides the capability of free selection or prohibition of
based on free selection according to the network interconnection
the set- rules.
ting by
the sub-
scriber
or the
call
Service When the MSG series gateway works as the media server
tone on of the SoftSwitch, one can dial an access code and a
trial service tone number on a telephone to listen the service
tone loaded to the gateway. It is convenient to find out and
test the service tone.
IP Centrex
Functions of IP ZXSS10 SS1b can provide the IP Centrex service. Besides the
Centrex functions of the traditional Centrex service, it also provides exten-
sion subscribers with new services that only the SoftSwitch sys-
tem has to meet requirements for data communication. In terms
of network, it can provide some functions unavailable on the tradi-
tional switches. It supports CENTREX application modes crossing
exchanges, regions, a variety of equipment and terminals.
The IP Centrex service is a high-quality service provided by the
SoftSwitch control equipment, which is applicable to enterprise
users at various scales. It can replace the traditional PBX or be
used to enhance the traditional telephone service of the exist-
ing PBX. Besides the traditional "*#" plus number service appli-
cation mode, ZTE SoftSwitch control equipment can also provide
WEB-based services and self-help management. Subscribers can
subscribe or cancel services via the WEB page, which is very con-
venient.
ZXSS10 SS1b supports intra-group subscribers registering the full
network intelligentized services, such as calling party CRBT and
called party CRBT.
Figure 18 shows the service provisioning mode of IP Centrex.

FIGURE 18 SERVICE PROVISIONING MODE OF IP CENTREX
The IP Centrex of ZTE SoftSwitch control equipment can provide

all functions of traditional Centrex service, as listed in Table 10.
TABLE 10 FUNCTIONS OF IP CENTREX
Service Description
Intra-group calls Subscribers within the same group can directly

dial the extension numbers. The called extensions
can be set with special ringing tones.
Inter-group 1. Inter-group subscribers can directly call

incoming calls extension subscribers within the group. The
called extensions provide ordinary ringing.
Moreover, extensions can be set with external
call restriction.
2. External calls can be transferred through the
attendant. With the incoming call transfer
function, calls can be transferred from any
extension to another. If the transfer fails, the
original calls should be transferred back.
3. External calls can be automatically transferred
via the computer attendant console.
Intra-group Extensions can be connected to inter-group

outgoing calls numbers after the external access code is dialed
without transfer by attendants. Extensions can
be restricted in inter-group call origination. After
dialing the external indicating number, subscribers
can either dial numbers continuously or perform
dual-stage dialing (dial the subsequent number
after hearing the dial tone). The specific mode can
be set as requires.
Call origination Every extension can be set to prohibit call

or acceptance acceptance or origination.
prohibited

Service Description
Phone set locking
Incoming call For intra-group or inter-group calls, subscribers

transfer can transfer incoming calls by means of pressing
the hookflash.
Intra-group Extensions within the same group can pick up

pick-up incoming calls to any other extension of the group.
Assigned pickup When extension A is ringing, extension B of the

call same group can pick up the incoming calls to
extension A.
Secretary service A secretary station can be designated to pick up

calls to superiors and call back.
Secretary station Generally, superiors of an enterprise do not want

service to accept all calls. With this service, calls to them
will be first transferred to the secretary station,
and then the secretary will ask the superior for
instructions whether to accept the call (in this
case, callers will hear music).
Attendant Centrex subscriber groups can be set with

attendants. The indicating number that an
inter-group subscriber used to call the Centrex
subscriber group can be connected to the
attendant of this group. Intra-group subscribers
can dial a specified number to call the attendant.
Attendant Login Calls to the simple attendant console are queued

/Cancellation by the queuing machine. When the console logs
in or exits, SS shall notify the VIC system which
notifies the queuing machine to queue or cancel
the queuing.
Computer voice It can provide subscribers with voice prompts

attendant console during automatic call handling, and subscribers
perform various service functions as prompted.
Emergency Attendants are provided with the emergency

override override function. With this function, the
following services of intra-group extensions can be
overridden and the extension subscribers can be
dialed directly. The overriding services include call
transfer, absent subscriber service, do not disturb
and incoming call restriction on extensions.
Missing call query With this service, the system will give the number
of missing calls in voice announcements when
subscribers pick up phones. Subscribers can query
the time and number of missing calls according
to voice navigation.
Closed user group Only intra-group calls are allowed. Such

subscribers cannot call the ones outside the group.
Abbreviated Refer to the Primary PSTN Voice Services and

dialing Supplementary Services topic for details.
Caller Display the number of incoming call on the LCD

identification of the phone.

Service Description
Hotline Provide immediate hotline and delayed hotline.
Outgoing call Add a level to the user rights: inter-group

screening outgoing call screening.
Do not disturb Refer to the Primary PSTN Voice Services and

Supplementary Services topic for details.
Alarm clock Refer to the Primary PSTN Voice Services and

service Supplementary Services topic for details.
Interception When a subscriber calls a vacant number, the

service called number has changed, the route is blocked
temporarily or the subscriber uses the service
inappropriately, such calls will be intercepted
automatically and forwarded to the answer-agent
device.
Call waiting Refer to the Primary PSTN Voice Services and

Malicious calls 1. The calling number can be traced within 30

tracing seconds after the caller hangs up.
2. If the caller does not hang up, the called can
continue to talk to the caller after tracing the
calling number.
3. The calling number can be either intra-group
or inter-group.
Forwarding Refer to the Primary PSTN Voice Services and

unconditionally Supplementary Services topic for details.
Forwarding on Refer to the Primary PSTN Voice Services and

busy Supplementary Services topic for details.
Call forwarding Refer to the Primary PSTN Voice Services and

on no reply Supplementary Services topic for details.
Recall on busy Refer to the Primary PSTN Voice Services and

Three-way Refer to the Primary PSTN Voice Services and

connection Supplementary Services topic for details.
Call parking on Refer to the Primary PSTN Voice Services and

busy Supplementary Services topic for details.
Absent subscriber Refer to the Primary PSTN Voice Services and

service Supplementary Services topic for details.
Conference call Refer to the Primary PSTN Voice Services and


Service Description
Mixed group The group data is configured on one or more than

(SHLR) one SHLR, the lengths of full numbers and short
numbers are consistent. For the local users in
the group, SoftSwitch can translate full numbers
and short numbers based on the VLR data, and
carry out the service connection control on the
translated full numbers. For users of other offices
in the group, SoftSwitch can send the SRI request
of the short number to SHLR, and then SHLR
returns the corresponding full number to the
SoftSwitch. Based on the full numbers, SoftSwitch
carries out the service connection control.
Mixed group The group data is configured on SHLR and SCP.

(SHLR and SCP) When the SoftSwitch queries the short number
routing from SHLR, SHLR returns the access code
+ short number if there is no the corresponding
full number. SoftSwitch triggers it to SCP after
receiving the returned message.
No outgoing When a user in the group call a user out of the

group prefix group, no outgoing group prefix might be carried
(SHLR configures a group without the outgoing
group prefix with C, D, or E).
Group selected by This is the continuous selection or optimum

two tandems selection for the group selected by two tandems.
Because the status of users between two tandems
does not synchronize, the two tandems might
select the same user, and therefore, another
tandem fails.
Small switch When working as a caller, the caller numbers of

group (with pilot the terminal office group are the same one, that is,
line number) the pilot line. When working as a called, only the
pilot line is the called number. When the tandem
office handles the pilot line, the tandem office can
make a call to the pilot line or accept the call made
by the pilot line even the pilot line is busy.
Cross-Domain An attendant, in the “Insert access code + Called

attendant console number” mode, notifies the SoftSwitch of inserting
insertion the specified user. If the insertion is successful,
three parties perform the triple talks and select
whether listen to the insert prompt tone.
Cross-Domain An attendant, in the “Listen access code + Called

attendant console number” mode, notifies the SoftSwitch of listening
listening the specified user.
Cross-Domain When user A activates services such as Do

attendant console Not Disturb, forwarding unconditionally, and
emergency incoming call restriction, the attendant can
override override the mentioned services and call user A
in the “Emergency override access code + called
number” mode.
Cross-Domain When user A is talking, the attendant can break

attendant console down the undergoing talk in the “Forced release
forced release access code + called number” mode. At the same
time, user A directly talks to OPR.

Service Description
Alarm clock SoftSwitch need notify the attendant console

information of the following information of users in the
notification group: Register alarm clock, Cancel alarm clock,
(notify the Successful wake-up, Failed wake-up, Wake-up
attendant frequency. In thus a way, the attendant can obtain
console) the alarm clock situation and manually perform
the failed wake-up.
Subscriber Status An attendant can query the user status in the

Query processable group, including Busy, Free, and
Block. SoftSwitch need tell whether the attendant
allows the query of the user status and returns
the current status.
Login and logout Calls to the simple attendant console are uniformly
notification of the queued by the automatic call distributor (ACD);
simple attendant therefore SoftSwitch need notify the attendant
console console of logins and logouts. Then, the attendant
console notifies the ACD of in-line or out-of-line.
Incoming/Out- When services such as incoming/outgoing simple

going simple at- attendant console, simple attendant console
tendant console release and call forwarding are performed,
and simple at- SoftSwitch need notify the attendant console to
tendant console update the queue status.
release notifica-
tion
Call back on After the automatic attendant console is activated,

automatic a user dials the pilot line to access the automatic
attendant console attendant console and then dials the extension
forwarding failure number. If the extension has no response or
is busy, the call goes back to the console and
prompts the failure.
Conversion If a caller calls the voice attendant console but

from automatic does not dial for a long time (by default the
attendant console duration is 15 seconds), the call is transferred to
to standard manual attendant console.
attendant console
on not dialing
Announcement When waiting queues of all the attendant consoles

playback on in a group are totally busy, this announcement is
attendant console played to notify the next caller the busy situation.
total busy
Call back of the Based on the configuration, when a user out of the
incoming call group dials the full number of a user in the group,
the call under one of the following conditions is
directly transferred to the attendant console:
1. Dialing a full number in the group, the call is
directly transferred to the manual attendant
console.
2. Dialing a full number in the group, Do Not
Disturb is activated by the user in the group.
3. Dialing a full number in the group, the user in
the group restricts the incoming call.

Service Description
Attendant User A calls the attendant and demands to put

pre-occupation through user B in the group. If the user B is busy,
the attendant clicks the pre-occupation button,
and user A listens to the ring back tone, then OPR
exits the call. As soon as user B is free, B side
automatically rings. User B can directly pick up
the phone to talk with user A.
CRBT Users in the group can subscribe the CRBT service.

When ringing, the caller can listen to the CRBT.
IP Public Phone
ZXSS10 SS1b can provide IP public phone services. According to
different billing methods, we can divide IP public phone services
into two implementation types: compact and standard.
The compact IP public phone service uses the traditional meter
in billing. The corresponding user port of IAD or AG is connected
with the meter. When a conversation ends, the SoftSwitch control
equipment will tell IAD or AG to generate a polarity reversal pulse
to provide for the connected meter for billing. This mode needs
no configuration of extra device at the network side. Being sim-
ple and quick in implementation, it is applicable to self-employed
business operators. However, the compact IP public phone is only
applicable to ordinary phones accessed via the IAD/AG and cannot
be applied to intelligent terminals and visual terminals. In addi-
tion, its billing algorithm management is difficult and the charging
duration is incorrect. Therefore, it is recommended to apply the
standard billing mode to large-scale IP phones.
Figure 19 shows how the compact IP public phone is implemented.

FIGURE 19 COMPACT IP PUBLIC PHONE
The standard IP public phone service belongs to the prepaid ser-

vice of the SoftSwitch control equipment, whose subscribers are
prepaid calling subscribers. During operation, it is necessary to
configure corresponding equipment at the network and subscriber
sides. Its architecture is shown in Figure 20.
The network side or the public phone management center consists
of three parts: the management server, database and manage-
ment position. Generally, the database and management server
share one PC server (hardware) platform; while the management
position is an ordinary PC installed with management software.

FIGURE 20 STANDARD IP PUBLIC PHONE
The management server can be taken as an AAA sever, which in-

teracts with the SoftSwitch control equipment via the (Remote Au-
thentication Dial-In User Service) RADIUS interface. When sub-
scribers use the IP public phone service, their calls will be sent to
the management server of the public phone management center
by the SoftSwitch control equipment via the RADIUS interface for
authentication, authorization, and accounting. In addition, such
management functions as real-time disconnection will be imple-
mented for these calls according to the available balance of cus-
tomers.
The management server also provides the data storage function.
It can save the data configuration information, charging config-
uration information and generated accounting information of the
public phone bar. It implements centralized management for op-
eration positions of various business halls, distributes information
(e.g. real-time CDR transmission) to various operation positions,
and responds to the operations of operation positions (e.g. pre-
paid deposit setting and billing query statistics).
The management position provides such centralized management
functions as data configuration, accounting setting and operation
statistics for public phone bars. It functions as follows:
1. Support the login of a super administrator.
2. Supports management and setting of public phone bars and
their equipment
3. Make and query business statistics of all public phone bars and
provides reports for settlement.

4. Support unified charge rate and charge algorithm, as well as

different charge rates and algorithms for different bars.
5. Monitor the operation of phone bars in real-time.
At the subscriber side, it is necessary to configure the operation
position and the corresponding terminals. The operation position
interacts with the central management server, implementing some
client management functions (e.g. prepaid deposit setting and
instant charging). In addition, it controls the IAD at business halls
via the standard SNMP interface and performs such functions as
phone locking/unlocking and phone status display.
The standard IP public phone is applicable to terminals of any
types, including ordinary phones accessed via the IAD/AG, and
intelligent terminals and visual terminals, thus facilitating imple-
mentation of unified management. It can be used widely in busi-
ness halls providing integrated telecom services.
One IP public phone management center of the basic configuration
can manage 64 IP public phone bars simultaneously.
Traditional Intelligent
Services
Interworking with the traditional IN equipment SCP, ZXSS10 SS1b
provides all traditional intelligent services, as listed in Table 11.
TABLE 11 TRADITIONAL INTELLIGENT SERVICES
Service Description
Account card It allows subscribers to make calls on any phone

service without paying cash. The charge is only recorded
in a specified account.
Prepaid service It is an account card service with the calling

number as the card number.
Telephone Generally, a manufacturer applies for an AD

advertisement telephone number and then plays advertisement
service to users calling this number. After dialing this
number and listening to the advertisement,
ordinary users can dial the phone number they
want to according to the voice prompt. In this
case, the call is free of charge or the user pays
only part of the charge (The other part is paid
by the manufacturer).
Centralized A service subscriber can provide a free phone

payment by the number for callers. Then, the charge of all
called calls of this number will be paid by the service
subscriber.

Service Description
All-in-one service The all-in-one service consists of caller

authentication single-stage dialing IP telephone
service, dual-stage dialing IP telephone service,
direct single-stage dialing toll service, dual-stage
dialing toll telephone service and account card
network access service etc.
Number portability It allows a subscriber to remain the phone

service number as it is when the geographic location
changes or the carrier providing telecom services
changes.
Universal personal With a single Personal Telecommunication

telecommunication Number (PTN), the subscriber can have access
to any network, and originate/receive any kind
of calls across multiple networks.
Telephone payment It refers to such commercial payment acts as

service paying the electricity charge, water charge or
call charge via the telephone.
Televoting It is the polling service via telephone networks.
VPN service It allows service subscribers to set up their own

private networks in use of the public network.
These private networks can have their own codes
and override regions or countries.
Telephone Subscribers can conduct such operations as

securities trading and querying securities or modifying
passwords via any dual tone telephone.
Receiving end When a subscriber of a telecom network (e.g.

network access a fixed phone subscriber) makes a toll to a
service subscriber of another telecom network e.g. a
mobile subscriber), the current originating end
network access is changed to receiving network
access.
Number change The number change notification service provides

notification service subscribers with number change prompts.
Toll grooming on When there is a toll for a local subscriber, and

busy the subscriber is in conversation with another
local subscriber, a prompt is inserted into the
conversation, reminding the former subscriber
that there is a toll. The called subscriber can
select to continue the talking or to receive the
toll in precedence.
Telephone payment The telephone payment service enables

service subscribers to perform bill query and payment
via the telephone.
Bank card It allows subscribers to make calls at any phone

telephone without paying cash. In this case, the charge is
recorded in the specified account and the bill is
paid through the bank card.

Service Description
Called payment The caller dials the specified access code of called
service payment service, and then the called number.
The system gives a voice notification to the
called to tell whether the called accept the call.
If the called agrees to pay the call (determined
through the voice interaction), the system puts
through the called. This bill is on the account
of the called.
Display of POTS After the terminal office user subscribes

terminal bill of the the instant bill display service on the SHLR,
terminal office SoftSwitch triggers the SCP service when the
terminal office user initiates the call. SCP
balances the bill of this call. After the call, SCP
requests the SoftSwitch for an attempt call that
carries the bill information along with the caller
number. SoftSwitch sends the IAM call request
with which the caller number is bill information
to the terminal office user. The terminal displays
the bill information as the caller identification.
F3G Intelligent Services

Interconnecting with the IN equipment and SHLR, ZXSS10 SS1b
provides unique ZTE services besides all traditional intelligent ser-
vices. The service mode is called fixed network 3G service.
The notability of the fixed network 3G intelligent services is the in-
troduction of SHLR, where subscriber data or part subscriber data
is saved.
Due to the co-existence of SoftSwitch network and PSTN (LS ex-
change), the introduction of SHLR separates the Directory Number
(DN) from the routing function completely. The introduction of Lo-
cation Routing Number (LRN) leads to two numbers:
1. DN, also called logic number, is a unique number assigned to
the subscriber by the carrier. It is also the number the sub-
scriber tells others. When the subscriber acts as the calling
party, DN is the calling number displayed for the called party.
When the subscriber acts as the called, the DN is the number
dialed by the calling party. DN is also the number used by the
carrier for subscriber identification and billing.
2. LRN, also called subscriber physical number or subscriber line
number, is a routing number assigned by the operation to the
subscriber. The number is not publicly known. It is an internal
network number.
The number corresponds to a port or a subscriber line of a
fixed switch. The switching equipment of the local network
can address and locate the subscriber through this number.
Due to the introduction of SHLR and the separation of DN and LRN,
the calling process has taken place great changes. The changes
are as follows.

1. In making subscriber data or in subscriber’s transition, SHLR

inserts the subscriber attributes, and the LRN and DN data into
the SS.
2. The LS subscriber adopts the immediate hotline or ordinary call
to direct all subscribers’ calls to SS + SG. The calling number
in the sent message is the calling LRN, and the call number is
the called DN.
3. Embedded with the VLR function, SS corresponds to LRN and
DN according to the HLR insertion command. After receiving
the LS call, SS converts the DN according to the calling LRN,
and the call authentication process is performed.
4. In the conversion of SS, if the location of the calling subscriber
is found changed, supplementary service operation is made,
or the data is not reliable, a location update request will be
originated to the SHLR to regain the correspondence between
the DN and the LRN.
5. SS is converted to LRN according to DN, and the call is directed
to LS where the called LRN is according to the called LRN.
6. In the conversion of SS, if the data of the called subscriber
does not exist or the data is not reliable, a routing request will
be originated to the SHLR to get the correspondence between
DN and LRN.
If the signaling link between SS and SHLR is faulty, SS cannot
query callee outside its domain on SHLR. By data configuration,
the following can be selected: Connect the callee by original
callee number or release the call.
SS is added with the Connect by original number if SHLR is
faulty on-off option. By default, the flag is checked. If SS finds
that SHLR is unreachable, it will perform the number analysis ac-
cording to the original number to connect the call. If the flag is
not checked, SS releases the call directly if SHLR is unreachable.
Hybrid Number Assignment

Service
Hybrid number assignment is to make the number resource plan-
ning for subscribers in the local networks, including the NGN, PSTN
and PHS networks. DN the subscriber dials has no direct relation
with the service network and home location. One number can be
used in the PHS and fixed networks freely.
With the development of number resource scarcity and the hy-
brid number assignment service, the usage rate of numbers will
increase for a full usage of number segments.
The subscriber data of hybrid number assignment are usually
maintained in SHLR. DN (or MSISDN) is fixed for subscribers. LRN
(or MSRN) is updated according to the actual network location.
The ZTE intelligent solution plans unified SHLR to store the sub-
scriber data of the NGN, PSTN and PHS networks in a centralized
way. Therefore, unified hybrid number assignment can be made
in the NGN, PSTN and PHS networks.

The implementation principle of ZTE fixed network 3G hybrid num-

ber assignment is shown in Figure 21. The calls of subscribers
participating in hybrid number assignment must be all transited to
the TG or MS, with all subscriber data stored in the SHLR.
FIGURE 21 HYBRID NUMBER ASSIGNMENT APPLICATION
WAN Centrex Services

Based on the local Centrex service, the WAN Centrex service ex-
tends to a wider application scale. The subscriber group can over-
ride multiple local switches and different service networks, while
the data maintenance is performed in SHLR. Compared with the
service development in IN network, both maintenance and imple-
mentation of WAN Centrex are more convenient. As a result, the
alternative routing of massive voice channels is avoided. More-
over, the Centrex service can organize a virtual network for the
PHS subscribers and fixed network subscribers, implementing the
communication among them via dialing the PHS subscriber num-
ber or short number of the fixed network subscriber.
Besides the whole suite of traditional PSTN Centrex services such
as closing subscriber group, answering for same group member,
specified answering and attendant console, WAN Centrex provides
Web-based management and service self-assistance. For exam-
ple, the administrator can set different authorities for every tele-
phone over the Web page. In some cases, the local call authority
is given for the workplace telephone, the long-distance call au-
thority is given for functional departments and offices of leaders,
the international call authority is given to the foreign affair depart-
ment, or ordinary staff is only given the right to make internal calls.
Besides the setting of dialing authority, the charge threshold can
be set to each phone. In this sense, enterprises can effectively
control the business cost. All functions can be easily set by the
administrator over the Web page. Besides, over the Web page,
employees can query the conversion history and communication
expenses, customize and cancel services.

Number Portability (NP)

The NP service in the 3G fixed network is similar to the unified Per-
sonal Number (PN) in the existing network, but the implementa-
tion method differs greatly. The unified PN in the existing network
adopts the SCP for implementation, thus leading to inconveniency
and alternative routes. The use of SHLR implements PN in the lo-
cal network in a real sense, and is free of alternative routes.
In the local network, as the logical number of the subscriber has
nothing to do with his physical location. He can move freely with
PN within the scope of the local network, not restricted by area
planning. Once the subscriber makes service registry on the phone
set, and the SHLR will change the LRN corresponding to the sub-
scriber logical number. In this way, the PN is implemented, and
the subscriber attributes are moved to the new location automat-
ically. Thus, the service management is convenient.
This service is applicable to persons on constant move, which en-
ables them to have a unique number. It is used in conjunction
with the prepayment service, preventing malicious delinquency.
It is also applicable to the store that often change the location but
hope to keep the same contact number. In the case of frequent
city reconstruction, it solves large-scale PN problem.
During service development, the original number of a subscriber
can be moved from one place to another, with no need to change.
That is, the number has nothing to do with the location of the
subscriber.
Pre-paid Service
For this service, subscriber must first prepay an appropriate
amount at subscription (or charge the number account by pur-
chasing the pre-paid phone card). When a call is connected, the
subscriber account decides whether to accept or refuse the call.
The call is charged real-timely in the process of conversation and
the call charge is deducted from that prepayment. When the
prepaid amount is used up, the system plays the announcement
for the subscriber to remind of the payment for continuous service
provisioning.
Traditionally, the caller is required to dial the access code of the
pre-paid service or the caller switch is required to identify that the
caller is a pre-paid service subscriber. In this sense, the access
code of the pre-paid service is automatically added to the callee
number. Since SHLR is being used, the caller switch can trigger
the pre-paid service very easily.

PHS Hotel Service

PHS Hotel service, often used for a hotel, is intended to use the
PHS subscriber and the fixed subscriber to create a virtual internet,
which is uniformly managed and charged in the real-time mode by
the traffic console of a fixed switch. The PHS has the same number
as its bound fixed phone. No matter whether the PHS subscriber
or the fixed subscriber originates a call, the same caller number is
displayed. For a hotel, the long number or hotel exchange number
plus room short number is optional. As other subscriber calls a
number, the two phone sets can ring at the same time. If one
hooks off, the other one stops ringing.
The hotel traffic console can manage the fixed subscriber and the
PHS subscriber in the rooms in a hotel, including modifying au-
thority, using service, activating part of new services and querying
services.
Thus, as a customer lives in a hotel, he/she can rent a public net-
work PHS terminal. The PHS authority can flexibly be controlled
at the hotel side. The rental PHS terminal adopts the real-time
charging. When the call ends, its CDR will be sent to the traffic
console at the hotel side in the real-time mode so as to carry con-
veniently out the settlement between the hotel and the subscriber.
Callee CRBT
The CRBT service is set by the callee. When a call goes to the
callee, before pick-up, the caller is played with a piece of music,
advertisement or message recorded by the callee. At the mo-
ment, what the caller heard is not the flat beep but the CRBT set
in the system by the callee. The personalized CRBT is preset by
the callee. It might be the ring tone or music offered by the sys-
tem, voice file recorded in advance by the callee via the IN system
or voice file downloaded in advance by the callee to the IN sys-
tem. At present, the CRBT service is already offered by the PHS
network. With the help or addition of the CRBT system, the fixed
network subscriber can register the value-added ring back tone
customization, meeting the personalized requirements and pro-
moting the business income of the operators. If SHLR manages
the subscriber data of the local network, the CRBT service infor-
mation registered by the callee can be stored in SHLR. When the
callee is called, the CRBT service attributes or CRBT service access
code can be returned to SS, which triggers the service to the ser-
vice platform.
Caller CRBT
Caller CRBT allows users to set their preferred CRBT. When the
caller makes a call and the called rings, the caller can hear the set
CRBT instead of the CRBT of called. The CRBT service information
subscribed by the caller is stored in the SHLR. When the specified

called is called, the CRBT service attributes or access code is re-

turned to the SoftSwitch. SoftSwitch triggers the service to the
service platform.
At present, caller CRBT is realized in the following two modes:
� SCP control mode
� Switch dual-call mode
Caller CRBT and called CRBT share the same processing mode,
only the service access codes are different.
Non-Personalized CRBT
Except the caller CRBT and called CRBT, ZXSS10 SS1b can also
provide the non-personalized CRBT service. This service is applied
to the user group that requires CRBT but not frequently changes
the tone. Non-personalized CRBT is offered by the ZXMSG 9000
media server.
One Number Multiple-Phone

The One Number Multiple-Phone services mean a public number
can be shared by multiple subscriber numbers. Here, the multiple
subscriber numbers are selected in turn. Ringing one by one or
ringing at the same time can be selected. The One Number Mul-
tiple-Phone services cover One Number Two-Phone, One Number
Three-Phone and Same Number Service. The One Number Two-
Phone subscriber owns a fixed telephone and a PHS phone at the
same time, that is, the fixed telephone and the PHS phone share
the same number. When a call comes to the number, both ring.
When a call goes from them, only one number is presented. The
detailed service description is as follows: One Number Two-Phone
means one number (unified number) is configured to two physical
terminals (PHS phone and PSTN phone). For the subscriber, there
are two physical terminals but only one number is given. One
Number Two-Phone has the following functions:
1. When a call comes in, the two physical terminals configured
with the One Number Two-Phone number ring at the same
time. The subscriber can pick up any of the physical termi-
nal. None of the pick-up is charged.
2. When a call goes out, the subscriber can make the call on any
of the terminal. The caller number presented for the callee is
the One Number Two-Phone number. The number is charged.
3. If one of the two One Number Two-Phone physical terminals
makes a call going to another terminal by dialing the One Num-
ber Two-Phone number, another terminal rings and can be an-
swered. The caller number presented for the callee is the One
Number Two-Phone number. The number is charged.
4. If one of the two One Number Two-Phone physical terminals is
in a conversation, another terminal can make a call or accept
the incoming call.

IP-based Value-Added
Service
The NGN-based unified service platform (Product model: ZXUP10)
is a platform developed by the ZTE according to the new ser-
vice generation specifications in the NGN system. Using the Open
Service Access (OSA) standard architecture for a reference, the
ZXUP10, based on the establishment of the unified and open ser-
vice platform, separates the service deployment and the bearer
network as the independent part. This move is favorable for the
third party service provider to participate in developing the ser-
vice; it can promote the interworking of multiple networks and the
rapid deployment of new customization services. Additionally, this
can provide service for the PSTN, NGN and 3G subscribers. The
ZXUP10, ZTE NGN unified service platform, can externally provide
the open, standard, PARLAY Application Program Interface (API).
ZXSS10 SS1b can interwork with the ZTE application server
ZXUP10 to provide the IP-based value-added service listed in
Table 12.
TABLE 12 IP-BASED VALUE-ADDED SERVICES
Service Description
Clicking Subscribers can originate calls by clicking phone

dialing numbers on the WEB page.
WEB 800 It is a kind of WEB-based called payment service.

In this case, calls are originated from PCs while the
information input and selection are via the WEB page.
Service subscribers manage their own information
through Web pages, and administrators manage
service subscribers through Web pages.
WEB It is a WEB-based conference telephone service. The

conference conference information is displayed on the WEB page.
The conference chairperson makes a reservation
for the conference via the WEB page and conduct
real-time management on the conference. There are
two modes for conference members to participate
in the conference: convergence and divergence. In
addition, conference requests are submitted to the
chairperson via the WEB page.
Accounting It allows subscribers to make calls at any phone

card without paying cash. In this case, the call charge is
recorded in the specified account
Unified It means to digitalize the voice information, electronic

message information (e.g.: e-mails) and text information
(e.g. fax) and save them in the same format. The
information can be sent with any one in the following
tools: e-mail, fax, voice mail, MS and telephone. After
being transformed and transmitted, the information
will be saved in the central mailbox of the system
unifiedly. While receivers can connect to the system
server with any tool at any moment and place
conveniently to them and acquire needed information.

Service Description
One number The subscribers can access any network and

links you originate/receive any kinds of calls across multiple
networks with the ONLY personal communication
number (PTN). According to requirements of
subscribers, the personal communication number can
be translated into a specific communication number
and connect incoming calls to a specified place without
being affected by geographical locations.
CRBT After registering the CRBT service, subscribers can

trigger this service and play RBT for the calling party
when receiving a call.
Voice Visitors can leave messages for system subscribers and

mailbox the latter can hear these messages. In addition, this
service allows subscribers to manage their personal
bulletins and let any visitor to hear their voices.
Subscribers are allowed to conduct personal setting,
including setting the login password and welcome
words.
The ZTE unified service platform hardware comprises the Parlay

gateway and the Application Server (App Server). Additionally, the
TTS server and the media server can be configured as required.
1. Parlay gateway
PARLAY gateway interacts with the SoftSwitch control equip-
ment via SIP and allocates appropriate protocols to calls from
the SoftSwitch control equipment. It can also directly control
the media server, provide calls with various media resources
needed in the services and charge the relevant services. In
addition, the PARLAY gateway can provide the API interface
for the application server (it can be the application server of a
third party).
2. Application server
The application server can run various intelligent services. The
selection of mini computers or servers is based on the logic
complexity of services. The application server is responsible for
the logic execution and management of various value-added
services and intelligent services. It interacts with the PARLAY
gateway via PARLAY/CORBA Application Programming Inter-
face (API).
3. Media server
Media server is an independent device that provides specialized
media resources in the SoftSwitch system. It is also an impor-
tant device in the Next Generation Network (NGN). It provides
the media processing function in primary and value-added ser-
vices, including collection and coding of DTMF signals, gen-
eration and sending of signal tones, sending of recorded an-
nouncements, conference and conversion between coding and
decoding algorithms, and other resources functions as well as
communication function and management maintenance func-
tion.
Application server of the third party can use API provided by
ZXUP10 for service development. It can also interact with the
PARLAY gateway directly via PARLAY/CORBA and then interact

with the SoftSwitch control equipment via the PARLAY gateway.

Moreover, it can support independent service development and
interact with the SoftSwitch control equipment directly via SIP.
SoftDA Service
In the NGN, the most obvious feature is that the subscriber is
reachable though IP address. In such favorable situation, the ZTE
has developed the SoftDA service on the basis of the Parlay gate-
way software platform.
The SoftDA service has utilized the IP to improve the service ca-
pacity of previous common telephones. Replying on voice appli-
cation, the service, based on the SIP, can set up individual portals
to provide all types of data assistant services, with the data being
supplementary.
The SoftDA service includes Click To Dial (CTD), click to confer-
ence, Click To Fax (CTF), network number directory, call records
and message summary, intelligent call, intelligent routing, Pres-
ence, application sharing and other functions. It can provide the
subscriber with a whole digital communication platform.
The subscribers of the SoftDA service are the medium or high-end
ones who possess some knowledge of computers and network ap-
plication. Aiming at the major customers and the commercial sub-
scribers, this service needs more favorable communication envi-
ronment to improve its efficiency. The SoftDA service is popular-
ized in the group users as the data Virtual Private Network (VPN).
The SoftDA service includes:
1. Login number
When logging in to the SoftDA service, the subscriber can bind
a number to a PC, a telephone set, or a mobile phone. The
subscriber will make or receive calls at that number.
2. Status display
After login to the SoftDA service, a subscriber can view the
status information (online, busy, off, answering a call, and so
on) of his friends; likewise, his friends can view his status in-
formation.
3. Calling a friend
The subscriber can make a call to one of his online friends. The
called party can answer the call at the number bound at the
time of login.
4. Making calls
The subscriber can dial a fixed-line number or a mobile phone
number with the dial-pad on the SoftDA interface.
5. Conference call
The subscriber can convene a teleconference with several se-
lected friends or called parties.
6. One number links you (ONLY)

A subscriber can activate the ONLY function and integrate sev-

eral numbers into one strategically. As a result, if an incoming
call arrives when the subscriber is off, the call will be forwarded
to a mobile phone or other telephone set as set by the sub-
scriber.
7. CRBT
A subscriber can set customized ring-back tone. When making
a call to the subscriber, the caller will hear the ring-back tone
set by the subscriber instead of the monotonous beep sound.
8. Instant message
Two online SoftDA service subscribers can start the instant con-
versation, namely, send and receive instant messages in the
form of text.
9. File transfer
Two online SoftDA service subscribers can transfer files each
other.
10. Call information record and summary
A subscriber can view his own call information in a certain pe-
riod of time in the past, comprising received calls and missing
calls.
11. Unified Message Service (UMS)
A subscriber may adopt the UMS service to communicate with
other subscribers. The UMS service can be employed in differ-
ent forms, like mail, short message and telephone.
12. Video calling (optional)
The SoftDA service supports video calling function. Two parties
in a conversation can see each other.
Multimedia Service
Multimedia service integrates image, voice and data, like video-
conference, two-point video communication, voice interworking
(between videoconferencing terminal, telephone, handset and
other voice terminals), video office collaboration and remote mon-
itoring. Multimedia services as new strategic services launched by
the telecom operators come next after voice and data services.
In the IP network, the provision of multiple multimedia applica-
tions such as videoconference, two-point video communication,
interworking with the voice terminal, video office collaboration has
become the goal of the communication network development. The
SoftSwitch platform is the best network platform to reach such
goal.
Taking ZXSS10 SS1b as the core, multimedia service can be real-
ized via SHLR and MCU as follows:
� Multiparty video conference
� Two-point video communication
� Voice interworking service

� All-in-one digital assistant service
Web Call Service

The Web Call service provided by ZXUP10 is the Web-based call
service, which is the call originates from the Web page. Using this
service, subscribers can perform the PC To Phone and PC To PC
functions. As a subscriber originates a call by clicking the called
number on the Web page and the call connects, the calling party
and the called party enter the conversation status. While main-
taining the call service for the subscribers, the Web Call service of
the ZXUP10 also provides powerful and flexible billing modes, for
example, deduction from account card, discount by time section.
The Web Call service can provide video call function, and set dif-
ferent charging rates according to the video functions.
The description of the Web Call service features is as follows:
1. Primary call
A subscriber can originate a call to a fixed-line phone or mobile
phone, or soft terminal on the Web.
2. Video call
A subscriber can originate a video call to a video phone or soft
terminal to support the video function. A subscriber can switch
off or switch on the video function. The system adopts different
charging policies according to the possibility of using the video
function by the subscriber in the conversation.
3. Call reservation
A subscriber can enter the card number and password on the
Web page to reserve a call. Both the calling number and the
called number are common telephone numbers; the call time is
set by the subscriber. At a given point, the system will respec-
tively call these two telephone sets. Thus, the call between
such two telephones occurs.
4. Familiarity number
A subscriber can set a familiarity number. During dialing this
number, he or she can enjoy certain preferential charging rate.
5. Fax
A subscriber can easily send the fax via the Web page.
Web 800 Service

The Web 800 service is a free phone service provided by the
ZXUP10.
It’s a new multimedia service representing the charging perfor-
mance. Its major feature is that some service subscribers pay for
the calls connected to them instead of the calling subscribers. The
service is originated from a PC. Through Web pages, information

is entered and selected. Moreover, a call is finished. Service sub-

scribers manage their own information through Web pages, and
administrators manage service subscribers through Web pages.
1. Use of service
A subscriber can select a 800 service subscriber that he needs
and department number through Web pages enters his own
telephone number, and click to implement the function. The
subscriber can also enter numbers of 800 service subscribers
that he knows to implement the function.
2. Charging
Service subscribers instead of calling subscribers are charged
for the service.
3. Service data management
Service subscribers can manage their own attributes through
Web pages. For example, they can modify routes or change
passwords.
4. Subscriber management
Administrators can manage the data of service subscribers
through Web pages, which include generation of subscribers,
modification on subscriber data, and removal of subscriber
data.
5. Unique number
It means that service subscribers can register only a Web 800
number if they have multiple public telephone numbers. The
calls dialing this only number can be connected to different des-
tinations according to the requirements of service subscribers.
For example, the call can be connected to different destina-
tions according to the originating area.
6. Call forwarding on busy/no answer
A subscriber can define multiple destination numbers. When a
destination number is busy in case of a call on busy/no answer,
the call is forwarded to next destination number.
7. Barring for calling
It means the service subscriber can accept only the calls from
some areas while barring those from other areas; or he can
bar the calls from some areas while accepting those from other
areas.
8. Select the destination location according to department num-
ber
It means that service subscribers can correspond to telephone
numbers of different destinations according to different depart-
ment numbers.
9. Selecting the destination by the caller location
It means the calls to the service subscriber are routed by caller
location.
10. Time route
It means that a service subscriber can set the time period to
correspond to the destination telephone number. In a time
period, the calls to the service subscriber will be connected to

the destination telephone number that corresponds to the time

period.
11. Temporary route
Service subscribers can set the temporary route for a destina-
tion telephone. Calls to the telephone will be transferred to the
set temporary number.
12. Password access
Service subscribers can register password for incoming calls.
Only after entering correct incoming call passwords, calling
subscribers can originate calls normally. Each service sub-
scriber has a management password. To enable the manage-
ment function, the subscriber needs to enter the correct man-
agement password.
13. Management function
Service subscribers can manage their own attributes through
telephone, for example, they can add, delete, modify, query,
prohibit, or permit a number, activate, deactivate, prohibit, and
permit a list, modify a temporary route, or change the pass-
word.
Web Conference Service

The Web conference, developed by the ZXUP10, is a new ser-
vice aiming at multiple networks. It is also a Web-based confer-
ence telephone service. Invoking the Parlay API provided by the
ZXUP10 to develop multimedia conference for realizing the service
logic, the subscriber can attend the Web conference via soft ter-
minal, common telephone set, SIP hard terminal, handset and so
on.
The conference chairman reserves conferences through Web pages
and manages conferences in the real time. Conference information
is displayed through Web pages.
Conference members attend conferences in the convergence mode
or divergence mode.
The conference members can also hold sub-conferences during a
conference. They request for sub-conference through Web pages
and submit the request to the chairman. Sub-conferences can be
held only after the chairman approved the request.
The sub-conference function enables conference attendants to dis-
cuss by groups. It is very resilient and convenient.
1. Prepaid function
Web conference can be a kind of prepaid service.
The sum is recorded on the conference card. The card bearer is
not entitled to reserve and convene a conference if there is not
enough balance on the card. The conference charges are com-
posed of the following parts: telephone conversation charges,
conference charges, and conference reservation charges. The
card bearer can query the conference card, and recharge the
conference card with a rechargeable card.

2. Conference chairman
As a subscriber holding the conference card, the conference
chairman is entitled to reserve the conference, to modify and
query the reserved conference.
According to the responses to the conference notification, the
conference chairman can determine whether to convene a con-
ference and manage the ongoing conference.
3. Conference member
Conference members refer to those invited to attend a confer-
ence by the chairman. A conference member can attend the
conference in the convergence mode or divergence mode.
4. Administrator
There are two types of administrators: system administra-
tor and common administrator. The system administrator can
manage common administrators, and common administrator
manages conferences.
5. Conference participating mode
Conference members attend conferences in the convergence
mode or divergence mode. The convergence mode means
when it is time to begin the conference, the conference mem-
bers dial the conference access codes, and input the confer-
ence numbers and conference passwords to access the con-
ference. Divergence mode means when it is time to begin the
conference, the system calls the members automatically, who
will access the conference after hooking off. In a conference,
these two modes can coexist.
6. Conference reservation
The conference chairman can reserve conferences. Fill in the
conference start time, end time, number of attendees, con-
ference members. For a conference convened in a divergence
mode, the telephone numbers of the conference members
should be filled in; for a conference convened in a convergence
mode, the conference number and the conference password
should be announced to the conference members. The reser-
vation result can be obtained after the chairman submits the
reservation.
7. Real-time management of chairman
After a conference starts, the chairman can manage the con-
ference in the real time. The chairman can see the dynamic
conference status, who are present and who are absent. The
chairman is entitled to delete someone who is present in the
conference by clicking on the web page, or add an attendee to
the conference by completing the member information, clicking
it and calling the attendee in a divergence mode. The attendee
will be added to the conference after going off-hook.
8. Displaying the conference status dynamically
The chairman can see the dynamic progress of the conference
and perform corresponding operations. The conference mem-
bers can see the status of the conference and request to join
or exit the sub-conference.
9. Sub-conference function

Conference members can request to join or exit the sub-con-

ference.
The application can be submitted through Web pages. The
application will be displayed after submission on the chairman’s
Web page. Thus, the chairman will accordingly execute the
next operation. The chairman is entitled to add some member
to the sub-conference or delete him or her.
10. Password access
When being added to the conference in the convergence mode,
the conference member is authenticated with the conference
password. This facilitates the conference convening and en-
sures that the attendees come at invitation.
After exiting halfway, the conference member can join the con-
ference again in the convergence mode.
11. Multiple call attempts
For subscribers joining the conference in the divergence mode,
the chairman can make multiple call attempts on busy or no
reply, so as to ensure his/her presence.
12. Address book
The chairman can create his own address book, and add com-
mon attendees to the book. When scheduling a conference,
the chairman can directly select the conference members from
the address book.
Unified Messaging Service

(UMS)
The UMS service provided by the ZXUP10 is intended for individual
subscribers. It offers a unified data platform for various terminals,
integrating voice, data, and fax services and eliminating commu-
nication barriers caused by the differences between terminals. It
digitizes information in different forms, such as voice information,
electronic information (email, for example), and characters (fax,
for example), and stores them in a unified form, enabling the com-
munication between subscribers of different terminals.
The information can be sent with any of the following tools: e-mail,
fax, voice mail, Mobile Station (MS) and telephone. After being
transformed, the information will be saved in the central mailbox
of the system. Receivers can connect to the system server with
any tool (from telephone, PC, fax, and MS), at any moment, and
in any place convenient to obtain the needed information.
1. Primary function
A subscriber can use any of the following terminals: telephone
set, fax machine, email terminal, WWW browser, WAP termi-
nal, SMS terminal.
The system can generate information in any of the following
forms: WAV file, TIFF file, TEXT file, emails, and short mes-
sages.

2. Information storage
� Leaving a telephone message
The existing voice mailbox function keeps unchanged. This
comprises actively leaving a message by the subscriber,
leaving a message in the auto answer mode and leaving a
message by bulletin owner.
� Email service provision UMS
It provides subscribers with email service. With the email
box provided by the system, a subscriber can receive email
messages forwarded by other email servers or send email
messages through the WWW.
� Sending a fax
The fax service provided by the existing voice mailbox
keeps unchanged. It can send, receive, and store faxes.
� WWW browser
A subscriber can send mails through the Web service of the
UMS system.
� Short message
The short message function is implemented through con-
nection with the Short Message Center (SMC). In general,
short message media are used for media notification and
they cannot be stored.
3. Message notification
� Message notification by telephone
When a voice, fax or Email message is added to the UMS
mailbox of a subscriber, the system will inform the mailbox
owner of the new message by calling the designated tele-
phone number of the owner.
� Message notification by email
When a voice or fax message is added to the mailbox of a
subscriber, the system will send the notification message
in the email mode to the Email address designated by the
mailbox owner.
� Message notification by pager
mailbox of a subscriber, the system will inform the mailbox
owner of the new message by originating the pager notifi-
cation.
� Message notification by short message
mailbox of a subscriber, the system will send the notifica-
tion message in the short message mode to the handset
designated by the mailbox owner.
4. Message retrieval
� Message retrieval by telephone
This mode can be used to retrieve voice mail messages and
Email.

� Message retrieval by WWW browser

This mode can be used to retrieve all types of messages.
With regard to Email, the POP3/SMTP protocol can be di-
rectly connected to the Email server to receive/send Email
messages. For voice retrieval, the Web Server is used to
visit the database of subscriber messages. After the voice
mail messages are obtained, they can be downloaded via
the PC or played. The fax message can be downloaded via
PC and be displayed.
� Message retrieval by Fax
This mode can be used to retrieve the stored fax messages
in the voice mailbox and the stored Email messages in the
UMS mailbox.
� Message retrieval by Email
This mode is used to send the voice or fax as an attachment
to the Email address designated by the subscriber.
� Message retrieval by short message
� In this mode, the voice can be sent in the voice short
message mode to the handset designated by the mailbox
owner; the fax or Email can be transformed into text to
send to the designated handset.

Chapter 5
Networking Configurat
ion
Table of Contents
Overview..........................................................................85
Networking Mode...............................................................85
Toll Network Solution .........................................................86
Local Tandem Office Solution...............................................87
Local Access Solution .........................................................88
Hybrid Networking Mode.....................................................94
System Configuration.........................................................94
Overview
This chapter describes networking modes to exemplify connection,
solution and detailed configurations of the product so that one can
understand the networking capability, service capability and envi-
ronment adaptation capability of the product.
Networking Mode
ZXSS10 SS1b supports various networking modes, which is
characteristic of sound expansibility, powerful system processing
capability and complete product series. It can provide multiple
networking modes according to different demands and network
environment. Therefore, it can meet requirements for not only
large-scale networking in telecom operation but also networking
of medium/small enterprises or even residential communities.
As a conclusion, ZXSS10 SS1b can provide networking modes from
C0 to C5, such as DC0, DC1 backbone network, DC2 (local-toll)
tandem and local network (Class 5).

Toll Network Solution

ZXSS10 SS1b can provide networking mode of the backbone
network (Class 4), which is an early application mode of the
SoftSwitch network. Figure 22shows the networking mode of
backbone network.
The typical configuration of toll network is ZXSS10 SS1b + ZXMSG
9000.
FIGURE 22 BACKBONE TOLL NETWORK SOLUTION (CLASS 4)
In the networking, ZXSS10 SS1b along with ZXMSG9000 works as

the toll office.
� As the control device, ZXSS10 SS1b is in charge of the call
control, resources allocation, billing management, and signal-
ing protocol control.
� ZXMSG9000 works as SG and TG at the same time to inter-
communicate with the local network through SS7.
� Between the ZXSS10 SS1b devices, the SIP-I protocol is used
to transfer the ISUP messages.
� The uniform service platform is used to serve users of the
whole network with various intelligent services. The standard
SIP interface is used between the service platform and the
SoftSwitch. The service platform uses the standard Parlay to
quickly provide new services. In order to protect the existing
intelligent network investment, the SS7 is used through SG to
interwork with the existing intelligent network. In this way, the
original intelligent network is inherited.

Chapter 5 Networking Configuration
Local Tandem Office

Solution
With the development of technology, traditional operators pay
more attention to NGN and mobile 3G. Traditional operators have
rich experiences of running the PSTN network and broadband
services, therefore have much PSTN and broadband resources.
They must ponder over the following points:
� To develop the existing network and converge the approaching
NGN and mobile 3G technologies.
� To make the fixed network “enjoy” the service platforms of
NGN and mobile 3G.
� To seamlessly move the fixed network users to the NGN and
mobile 3G networks.
� In terms of the mobile 3G network, how to take full advantage
of the existing merits of the fixed network at the very begin-
ning, and how to improve the service quality.
� The uniform service platform is used to serve users of the whole
network with various intelligent services.
ZTE offers the local tandem office solution, namely, 3G network
building solution of the fixed network, as shown in Figure 23.
FIGURE 23 LOCAL TANDEM OFFICE SOLUTION (CLASS 5 SOLUTION

1)
The above solution can provide the local/toll telecom service in

the backbone data network, and obtain the interworking among
the PSTN, IN, and services.
1. As the control device, ZXSS10 SS1b is in charge of the call

2. If SS7 is used by PSTN, TG is in charge of the trunk access

of SS7, while SG is in charge of the adaptation of SS7 and
converting the circuit bearer into IP bearer.
If the PRI signaling system is used by PSTN, MG (TG) uniformly
processes the voice media and signaling. SG can be omitted.
In Figure 23, ZXMSG9000 and ZXMSG7200 work as SG and TG
respectively and also work as gateway office.
3. The uniform service platform is used to serve users of the
4. In the local tandem networking mode, data of users of PSTN,
PHS, and SoftSwitch can be stored to SHLR. SHLR is respon-
sible for uniformly manage the data attributes of users in the
whole network. The core control device of SoftSwitch inter-
works with SHLR through standard MAP protocol or MAP+ pro-
tocol to obtain the user information.
� When the MAP protocol is used, data of users of PSTN, PHS,
and SoftSwitch can be stored to SHLR. SHLR is responsible
for uniformly manage the data attributes of users in the
whole network. The trigger mode of the intelligent service
in original networks such as PSTN is converted from num-
ber segment trigger to subscription trigger. In this way,
new services hardly being provided by existing networks
are realized, especially services such as number porting,
mixed number launch, and fixed network CRBT.
LS users adopt the instant hotline or common calling to
direct calls of all the users at SS+SG/TG.
SS has the built-in VLR function. It couples LRN and DN
according to the instruction of HLR. After receiving the call
of LS, VLR converts the caller LRN into DN and performs
the call authentication.
� When the MAP+ protocol is used, both primary and supple-
mentary services (including Centrex group services) of LS
users are realized by the terminal office.
New services of terminal office users, such as One Num-
ber Links You and CRBT, are triggered by SS to the new
provincial service platform through querying the subscrip-
tion information from SHLR. Both caller and called subscrip-
tion information are queried from SHLR. If both caller and
called have the intelligent services, a normal call is com-
pleted after multiple query flows.
Local Access Solution

Local network networking (Class 5) means that the SoftSwitch net-
work directly provides subscribers with interfaces and services.

According to network resources and environment, following solu-

tions can be provided:
� (Small Office Home Office) SOHO Solution
� Community Solution
� Intelligent terminal Solution
The applicable scope is listed in Table 13.
TABLE 13 ALL SOLUTIONS
Solu- Home Desk Corri- Com- School Com-

tion /SOHO dor munity mer-
cial
Build-
ing
SOHO ü ü ü ü ü
Solution
Com- ü ü ü
munity
Solution
Intel- ü ü ü ü
ligent
Termi-
nal Sol-
ution
Primary Mode of Figure 24 shows the primary mode of local networking. The typical
Local Networking configured devices cover ZXSS10 SS1b + ZXMSG9000 + MSAN +
IAD or intelligent terminal.
FIGURE 24 PRIMARY MODE OF LOCAL NETWORKING (CLASS 5

SOLUTION 2)

1. As the control device, ZXSS10 SS1b is in charge of the call

2. ZXMSG9000 works as SG and TG at the same time and works
as gateway office, interworking with other PSTN networks.
3. ZXMSG9000/7200 and MSAN can directly provide the access
of the Z interface subscriber, ISDN digital subscriber, V5 sub-
scriber and DSL subscriber. They can also access the ISDN PBX
group via the ISDN interface.
4. The uniform service platform is used to serve users of the
5. The access via IAD provides the integrated voice and data so-
lution for homes, corridors and offices.
6. It directly accesses the SIP/H.248 terminal.
7. In the local tandem networking mode, data of users of PSTN,
PHS, and SoftSwitch can be stored to SHLR. SHLR is respon-
sible for uniformly manage the data attributes of users in the
whole network. The core control device of SoftSwitch can in-
terwork with SHLR through standard MAP protocol or MAP+
protocol to obtain the user information.
SOHO Solution The mode of ZXSS10 SS1b + TG + SG + ZXSS10 I500/600 is
adopted in this solution. The condition is that users only have
C5 cable instead of twisted pair resources. It is mainly applied in
homes/SOHO, corridors, schools and commercial buildings. The
solution is very simple. The users can make the best use of the
existing network resources and realize the access of subscribers
quickly.
Users can select ZXSS10 I500 or ZXSS10 I600 as required, to
provide the pure voice or voice/data-integrated service. Moreover,
users can select different types for use in residences, corridors or
offices.
In the networking example shown in Figure 25, adopt the family
voice/data-integrated solution provided by ZXSS10 I601. In ad-
dition, multi-port IAD600 series products can be used in corridors
or offices.

FIGURE 25 DATA AND VOICE INTEGRATED SERVICE
To utilize the multi-port IAD500 product, a networking example

for pure voice solution used at corridors or offices is provided, as
shown in Figure 26.

FIGURE 26 PURE VOICE SOLUTION FOR OFFICES (CORRIDORS)
Community The mode of ZXSS10 SS1b + ZXMSG7200 is adopted in this so-

Solution lution. It requires the twisted pairs and is mainly applied in com-
munities, corporations and commercial buildings. The advantages
of this solution lie in its low cost. Via MG, it can provide the ac-
cess of several hundreds to several thousands of ordinary tele-
phone subscribers. The networking diagram is shown in Figure
27. Here, ZXMSG7200 provides trunk and subscriber access at
the same time.

FIGURE 27 COMMUNITY SOLUTION
Intelligent The ZXSS10 SS1b + ZXSS10 B100/B200 + IP broadband phone

Terminal Solution set mode is adopted in this solution. It requires C5 cables and is
mainly oriented to users who use the resource directly. The so-
lution is very simple. Users can make the better use of network
to provide value-added service and intelligent services. This is a
brand-new access mode, which directly uses such intelligent ter-
minals as IP Phone, PC and IP Fax to provide local services. It is
applicable to the situation in which users have widely distributed
C5 cable resources. This solution can be adopted in advanced com-
mercial buildings and luxurious houses. The networking example
is shown in Figure 28.

FIGURE 28 INTELLIGENT TERMINAL SOLUTION
Here, the ZXSS10 B100/B200 broadband access gateway can have

other routing device proxies, but ZXSS10 B100/B200 can perform
address conversion and subscriber access control.
For the local networking solutions above, if a call takes place be-
tween subscribers of a SoftSwitch network, the network does not
require communicating with PSTN, So TG and SG can be omitted.
Hybrid Networking Mode

The hybrid networking mode consists of the toll office networking
mode, the local tandem office networking mode and the local net-
work networking mode.
System Configuration
From the perspective of the reliability of the telecom equipment,
the basic configuration principles are as follows: the 1+1 backup
mechanism is applied to major boards, power modules, System
Control Card (SC) and GNIC/Network Interface Card (NIC, via data
configuration); the quantity of the SPC in SC depends on the net-
work capacity; here, the 1+1 backup is also applicable.

According to subscriber capacity, it can be configured with 1~13

SPCs, which provides the processing capacity of two million sub-
scribers or 200 thousand trunks at most (capacity varies depend-
ing on the traffic model).
The configuration quantity of SPCs should be ultimately deter-
mined according to the actual traffic model and in consideration of
such factors as network security and load balancing, when the load
of CPU is 55%, the processing capability of SPC is 46 CAPS/board.
The ZXSS10 SS1b SoftSwitch control equipment can be configured
with 2~4 NICs. Since NIC supports the 1+1 backup mode, there
are two configuration modes for the card: two NIC cards and four
NIC cards. The configuration principles are:
Every 10 SPCs is configured with a pair of NICs. Each SS shelf can
be configured with up to two pairs of NICs. NIC boards in the NIC
pair are in the active/standby partnership. If SS1b is configured
with more than one shelf (exclusive of one shelf situation), an
extra pair of NICs is required to be configured on the main control
shelf to be used for NM.
In addition, if the GNIC board with multiple FE interfaces is
adopted, GNICs work in pair. Each GNIC provides four FE in-
terfaces whose working mode is Trunk, externally providing one
IP address. If SS1b adopts the GNIC board, in spite of the
configuration mode of SS1b, its network interface board shall be
configured as follows: two GNICs + four NICs (one pair is for NM
and the other for charging). When there are two SS1bs in the
networking, an extra pair of NICs shall be configured to transfer
the heartbeat information between them. That is, the network
interface board of SS1b shall be configured as follows: a pair
of GNICs + three pairs of NICs (one pair for NM, one pair for
charging and the rest for heartbeat information).
A configuration example is listed in Table 14and Table 15.
TABLE 14 ZXSS10 SS1B HARDWARE CONFIGURATION LIST 1
SN Name Code Unit Quantity Remarks
1 12U shelf PCS 1
2 19-inch ZX209.01J-2 PCS 1

server
rack
3 Backplane BSSB2 PCS 1
4 Network NIC PCS 2~4 To be

interface configured
card in pairs
5 System SC PCS 2
control
card
6 System SPC PCS 2

protocol
processing
card

7 System SSN PCS 1

switching
network
card
8 System SSNI PCS 1

switching
network
card
interface
9 Transport TIC PCS 1

interface
card
10 Data PCS 1
mainte-
nance
man-
agement
server
11 Billing PCS 2
server
12 Power unit SPOWB PCS 1
13 Power dis- SPWBP PCS 1

tribution
box
14 Fan tray SPWBF PCS 1
TABLE 15 ZXSS10 SS1B HARDWARE CONFIGURATION LIST 1
1 12U shelf PCS 1
2 19-inch ZX209.01J-2PCS 1
server
rack
3 Back- BSSB2 PCS 1

plane
4 Network NIC PCS 2~4 To be

interface config-
board ured in
pairs
5 System SC PCS 2
control
card

6 System SPC PCS 1~13

protocol
process-
ing card
7 System SSN PCS 2

switching
network
card
8 System SSNI PCS 1

switching
network
card
interface
9 Inter- CSN PCS 2

shelf
cascading
board
10 Inter- CSNI PCS 1

shelf
cascading
interface
board
11 Transport TIC PCS 1

interface
card
12 Data PCS 1
mainte-
nance
man-
agement
server
13 Billing PCS 1
server
14 Power SPOWB PCS 2

unit
15 Power SPOWP PCS 1

distribu-
tion box
16 Fan tray SPWBF PCS 1


Chapter 6
Technical Specifications
Table of Contents
Overview..........................................................................99
Technical Specifications ......................................................99
Busy Hour Call Attempts (BHCAs) ........................................99
Maximum Subscriber Capacity........................................... 100
Signaling Processing Capability.......................................... 100
Protocol Processing Capability ........................................... 100
Call Authentication .......................................................... 101
Charging Performance ...................................................... 103
Time Monitoring .............................................................. 103
Reliability and Availability ................................................. 103
Physical Specifications...................................................... 104
Overview
This chapter describes ZXSS10 SS1b in terms of its technical spec-
ifications such as system processing capability, charging perfor-
mance, time monitoring, load capability, reliability and availability
and its environmental requirements.
Technical Specifications
As a large-capacity SoftSwitch control equipment, a single shelf
of the ZXSS10 SS1b supports 2000K (2 Million) BHCAs, and eight
cascading shelves supports a maximum of 16,000K ( 16 Million)
BHCAs, while eleven cascading shelves supports up to 22,000K (
22 Million) BHCAs.
Busy Hour Call Attempts

(BHCAs)
The maximum digital trunks and subscriber capacity of single SS1b
shelf are decided by the specific traffic model. When the load of

CPU is 55%, the processing capability of SPC is 46 CAPS/board.

The SPC boards are configured in pair and in the loadsharing/active
standby each other working mode partnership. Each pair of SPCs
provides the processing capability of 92 CAPS.
Maximum Subscriber
Capacity
The maximum digital trunks and subscriber capacity of single SS1b
shelf are decided by the specific traffic model.
When the load of CPU is 55%, the processing capability of SPC is
46 CAPS/board. The SPC boards are configured in pair and in the
loadsharing/active standby each other working mode part-
nership. Each pair of SPCs provides the processing capability of
92 CAPS.
Signaling Processing
Capability
� ZXSS10 SS1b supports up to 1024 local signaling points.
� A shelf of the ZXSS10 SS1b supports up to 1500 64k or 100
2M No.7 signaling links.
Protocol Processing
Capability
� The SIP/SIP-T/BICC protocol is adopted for interpretability
among the ZXSS10 SS1b SoftSwitch control equipment.
� The SIGTRAN protocol is used between the ZXSS10 SS1b and
signaling gateway.
� The H.248 protocol is used between the ZXSS10 SS1b and
trunk gateway.
� The H.248 or MGCP protocol is used between the ZXSS10 SS1b
and media gateway.
� The SIP protocol is used between the ZXSS10 SS1b and appli-
cation server.
� The MGCP protocol is used between the ZXSS10 SS1b and
MGCP terminal.

Chapter 6 Technical Specifications
� The SIP protocol is used between the ZXSS10 SS1b and SIP
terminal.
� The H.323 protocol is used between the ZXSS10 SS1b and the
existing IP telephony network.
� The MAP/MAP+ protocol are used between the ZXSS SS1b and
the smart home location register (SHLR).
� The existing intelligent network application (INAP) can be used
to enable the ZXSS10 SS1b to be interoperable with the exist-
ing intelligent network.
� It supports standard protocols such as V5.2, V5UA, IUA, ISDN
DSS1, ISUP, TUP, M3UA, and SCTP to support signaling trans-
mission and adaptation, as well as all types of user access in-
terfaces on the existing communication networks.
� For the H.248 protocol, ZXSS10 SS1b supports text coding.
� The supported No.7 signaling protocols include TUP, ISUP,
MAP, CAP, and INAP.
Call Authentication
The black/white list provided by the ZXSS10 SS1b can be used to
prohibit or allow certain calling parties or incoming calls from a
certain incoming trunk to go to certain destination codes based on
the calling number or incoming trunk identifier.
ZXSS10 SS1b defines the black/white list of calling numbers in
the outgoing/incoming trunk group and judge whether calls from
a caller in the black/white list can be connected through the trunk
group. A calling number can be a toll area code and part or all of
a user number.
ZXSS10 SS1b defines the black/white list of called numbers in the
outgoing/incoming trunk group and judge whether calls from a
called party in the black/white list can be connected through the
trunk group. A called number can be a service code, prefix of the
toll call, and part or all of a user number.
ZXSS10 SS1b can set the black/white list of calling/called numbers
globally.
ZXSS10 SS1b can intercept calls that have insufficient calling num-
ber digits, no digits or excessive digits according to different call
sources.
ZXSS10 SS1b authenticates the forwarded calls based on the
“original called number”.
ZXSS10 SS1b supports 500 million users on the black and white
lists simultaneously (external authentication server is required).
Up to 0.5 million numbers can be authenticated.
Table 16lists the call types that support the call authentication.

TABLE 16 CALL TYPES
SN Call Type Description
1 Local calls on the Calls in the SoftSwitch control area

current SS
2 Calls to the suburb Calls to the suburb and countryside in the

and countryside on same SoftSwitch control area
the current SS
3 National toll calls Calls to different area code in the different

on the current SS SoftSwitch control areas
4 International toll Calls to different country area code in the

calls on the current different international SoftSwitch control
SS areas
5 Local calls on the Calls to the PSTN through the trunk

outgoing trunk SS
6 Calls to the suburb Calls to the PSTN suburb and countryside

and countryside on number segment on the outgoing trunk SS
the outgoing trunk
SS
7 National toll calls Calls to PSTN over numbers with different

(intra-region) on area codes (area codes of caller and callee
the outgoing trunk belong to the same national area)
SS
8 National toll calls Calls to PSTN over numbers with different

(inter-region) on area codes (area codes of caller and callee
the outgoing trunk belong to the different national areas)
SS
9 International Calls to PSTN (gateway offices of other

toll calls on the countries) over numbers with different
outgoing trunk SS country codes via trunk
10 Charged special Dial the charged special service number in

service the SoftSwitch control area
11 Business group IPCentrex call authority

authority
12 Inter-group Intra-group authority

authority
13 Calls of high rate Calls of high rate in the SoftSwitch control

area
14 Local calls and Call other SoftSwitch device over the IP

network calls on network
the IP outgoing SS
15 National toll calls Calls to SoftSwitch with different area codes

on the IP outgoing via IP network
SS
16 International calls Calls to SoftSwitch with different country

on the IP outgoing codes via IP network
SS

Chapter 6 Technical Specifications
SN Call Type Description
17 Smart calls Call of the smart service invoked by

SoftSwitch
18 URL calls Call established through the URL address
Charging Performance
The primary charging performance specifications are as follows:
� Charging error rate: < 10-4
� Charging accuracy: 100 ms
� Accuracy rate: ≥99.96 %
� Bill processing capability: ≥10,000,000 CDRs /month
� Charging data storage capability: ≥10,000,000 CDRs (in case
of charging database ≥ 15G and file storage space ≥ 6G)
� Charging data buffer capability ≥ 1,000,000 CDRs
Time Monitoring
Specifications of time monitoring and loading capability are as fol-
lows:
� Time supervision for no dialing after hook-off: 10 seconds
� Time limit for no dialing in digit interval: 20 seconds (ad-
justable)
� Time limit for long time no reply of local calls: 60 seconds
(adjustable)
� Time limit for long time no reply of toll calls: 90 seconds (ad-
justable)
� Time limit for long time no reply of international toll calls: 120
seconds (adjustable)
� Time limit for howler tone: 40 seconds (adjustable)
� Time limit for busy tone: 15 seconds (adjustable)
Reliability and Availability

ZXSS10 SS1b is a carrier-class product. All important boards sup-
port active/standby hot backup and hot swapping, meeting the
99.999% (five 9s) reliability requirement of carrier-class equip-
ment. The maximum fault time in a year is less than 5.3 minutes.

Physical Specifications
� Dimensions
Cabinet dimensions: 600mm × 800 mm × 2,000 mm (W × D
× H)
� Weight
Empty cabinet: 350 kg

Chapter 7
Environmental
Requirements
Table of Contents
Power Supply and Power Consumption ............................... 105
Grounding Requirements .................................................. 106
Antistatic Requirements.................................................... 107
Anti-Electromagnetic Interference...................................... 108
Dust-proof Requirements.................................................. 108
Temperature and Humidity Requirements............................ 108
Noise ............................................................................. 109
Lighting ......................................................................... 109
Atmospheric Pressure....................................................... 109
Air Pollution .................................................................... 109
Power Supply and Power

Consumption
DC: -57V to -40V
The maximum single-shelf power consumption is calculated by
power consumption sum of all configured boards/0.8. Power con-
sumption of SoftSwitch boards and units are listed in Table 17.

TABLE 17 POWER CONSUMPTION OF SOFTSWITCH BOARDS
Board Average consumption per 8 hours (W)
SSC 24
SPC 22
NIC 22
SSN 18
TIC 1
SSNI 1
CSN 20
CSNI 2
GNIC 18
SPOWB 55
SPWBP 15
SPWBF 55
Power unit 30
Alarm box 20
Grounding Requirements
The grounding in the equipment room should meet the following
requirements:
1. Grounding modes
The grounding mode of the telecommunication office where
the SoftSwitch control equipment is located should comply with
the joint grounding principle, that is, the working ground, pro-
tection ground, and building lightning protection ground of all
communication equipment share one group of grounding con-
ductors, as shown in Figure 29.

Chapter 7 Environmental Requirements
FIGURE 29 RACK GROUNDING MODE
2. Sectional area of the grounding cables

The ground cable should be good conductor (copper). Do not
use plain conductor.
3. Grounding resistance
Generally, the resistance for the centralized grounding should
be less than 5 ohm.
Antistatic Requirements
� Generally, anti-static movable floor is required in the equip-
ment room where the SoftSwitch control equipment is
installed. The height of the movable floor should be between
300 and 330 mm. The resistance value of the unit raised
floor system should be in accordance with the Technical
Requirements for Raised Floor in Computer Rooms.
� If no movable floor is available, the conductive floor should be
used instead (resistance: 1.0x107 to 1.0x1010 ohm).
� The conductive or raised floor must be grounded and can be
connected to the grounding device with a current-limiting re-
sistor and a connection cable. The resistance of the current-
limiting resistor shall be 1 M ohm.

Anti-Electromagnetic
Interference
� The electric field intensity shall not be greater than 300mV/m.
� Magnetic intensity shall not be greater than 11 GS.
Dust-proof Requirements
� Dust with diameter over 5 mm shall be less than or equal to
3×104 granules/m3.
� The dust particles shall be non-conductive, non-magnetic and
non-corrosive.
Temperature and Humidity

Requirements
Table 18lists the temperature and humidity requirements.
TABLE 18 TEMPERATURE AND HUMIDITY REQUIREMENTS
Item Temperature (°C) RH (%)
Long-Term Short-
Working Term Short-
Condition Working Long-Term Term
(Note 1) Condition Working Working
(Note 2) Condition Condition
Scope 15°C to 0°C to 45°C 40% to 65% 20% to 90%

30°C
1. Under the long-term working environment of the device, the

measurement values of temperature and humidity refer to the
values measured 2 m above the floor and 0.4 m before the
equipment (measurement is done when there is no protective
board in the front or at the back of the rack).
2. The short-term working condition means that the continuous
running period is no more than 48 hours and the accumulated
running period in a year is no more than 15 days.

Chapter 7 Environmental Requirements
Noise
Indoor noise shall not be greater than 70 dB.
Lighting
� Direct sunlight should be avoided to prevent aging and distor-
tion of such components as circuit boards caused by long-time
exposure.
� Average luminance should be 150lx to 200lx with no glare.
� Emergency lighting is available.
Atmospheric Pressure
1. Operating: 1.08 × 105 Pa to 5.1 × 104 Pa
2. Storage: 1.08 × 105 Pa to 1.2 × 104 Pa
Air Pollution
� The equipment shall not be exposed to corrosive air, such as
sulfur dioxide and oxide containing ammonia.
� The equipment shall not be exposed to smoke such as oil flux
and soot.


Figures
Figure 1 MULTI-LAYER ARCHITECTURE OF ZXSS10 SS1B ......... 2

Figure 2 CONNECTION OF ZXSS10 SS1B ..............................18
Figure 3 ZXSS10 SS1B CABINET..........................................19
Figure 4 HARDWARE LOGICAL RELATIONSHIP OF ZXSS10
SS1B ...............................................................19
Figure 5 SOFTWARE STRUCTURE OF SOFTSWITCH CONTROL
EQUIPMENT ......................................................20
Figure 6 TYPICAL APPLICATION OF MAIN PROTOCOLS ............27
Figure 7 COMMANDS OF H.248 PROTOCOL............................29
Figure 8 RELATIONSHIP AMONG ENDPOINT, CONNECTION,
CALL, AND GATEWAY .........................................30
Figure 9 APPLICATION MODEL OF SIGTRAN PROTOCOL ..........32
Figure 10 STRUCTURE OF SIGTRAN PROTOCOL STACK ...........32
Figure 11 LOCATION OF M3UA ............................................34
Figure 12 PRIMARY NETWORK MODEL OF SIP........................35
Figure 13 3-GRADE STRUCTURE OF NO.7 SIGNALING
NETWORK ........................................................39
Figure 14 FUNCTION MODEL OF NO.7 SIGNALING .................40
Figure 15 FUNCTIONS OF V5 PROTOCOL...............................42
Figure 16 FUNCTIONS OF V5.2 INTERFACE ...........................42
Figure 17 SERVICE PROVISIONING MODE OF SOFTSWITCH
NETWORK ........................................................50
Figure 18 SERVICE PROVISIONING MODE OF IP CENTREX ......58
Figure 19 COMPACT IP PUBLIC PHONE .................................64
Figure 20 STANDARD IP PUBLIC PHONE................................65
Figure 21 HYBRID NUMBER ASSIGNMENT APPLICATION..........70
Figure 22 BACKBONE TOLL NETWORK SOLUTION (CLASS 4) ...86
Figure 23 LOCAL TANDEM OFFICE SOLUTION (CLASS 5
SOLUTION 1) ....................................................87
Figure 24 PRIMARY MODE OF LOCAL NETWORKING (CLASS
5 SOLUTION 2) .................................................89
Figure 25 DATA AND VOICE INTEGRATED SERVICE ................91
Figure 26 PURE VOICE SOLUTION FOR OFFICES
(CORRIDORS) ...................................................92

Figure 27 COMMUNITY SOLUTION........................................93

Figure 28 INTELLIGENT TERMINAL SOLUTION .......................94
Figure 29 RACK GROUNDING MODE ................................... 107

Tables
Table 1 Chapter Summary .................................................... i

Table 2 PRODUCT STANDARDS ............................................12
Table 3 COMMANDS OF H.248 PROTOCOL .............................29
Table 4 COMMANDS OF MGCP PROTOCOL .............................31
Table 5 MEANING OF DIGITAL LINE SIGNAL ..........................45
Table 6 MEANING OF SIGNALS ............................................46
Table 7 MEANING OF NO.1 SIGNALING LINE SIGNAL ..............47
Table 8 MEANING OF REGISTER SIGNAL ...............................48
Table 9 SUPPLEMENTARY SERVICES .....................................51
Table 10 FUNCTIONS OF IP CENTREX ...................................58
Table 11 TRADITIONAL INTELLIGENT SERVICES ....................66
Table 12 IP-BASED VALUE-ADDED SERVICES ........................74
Table 13 ALL SOLUTIONS....................................................89
Table 14 ZXSS10 SS1B HARDWARE CONFIGURATION LIST 1 ...95
Table 15 ZXSS10 SS1B HARDWARE CONFIGURATION LIST 1 ...96
Table 16 CALL TYPES ....................................................... 102
Table 17 POWER CONSUMPTION OF SOFTSWITCH BOARDS... 106
Table 18 TEMPERATURE AND HUMIDITY REQUIREMENTS ...... 108


List of Glossary
AAA
- Authentication, Authorization, Accounting
-
Ack
- Acknowledgement
-
ACL
- Access Control List
-
AG
- Access Gateway
-
API
- Application Program Interface
-
ARP
- Address Resolution Protocol
- IP
AS
- Application Server
-
ASP
- Application Server Process
-
ATM
- Asynchronous Transfer Mode
-
BHCA
- Busy Hour Call Attempts
-

BICC
- Bearer Independent Call Control
-
CAP
- CAMEL Application Part
- CAMEL
CDR
- Call Detail Record
-
CIC
- Circuit Identification Code
-
CLI
- Command Line Interface
-
COMM
- Communication board
-
CPU
- Central Process Unit
-
CSN
- Center Switch Network (Card)
-
CSNI
- Center Switch Network Interface (Card)
-
DC
- Direct Current
-
DN
- Directory Number
-

Glossary
- Distinguished Name
-
DPC
- Destination Point Code
-
DS
- DiffServ
-
DSS1
- Digital Subscriber Signaling No.1
- 1
DTMF
- Dual Tone Multi-Frequency
-
EMS
- Enterprise Management System
-
FDM
- Frequency Division Multiplexing
-
FTP
- File Transfer Protocol
-
GBIC
- Giga Bitrate Interface Converter
-
- Gigabit Interface Converter
-
GK
- Gatekeeper
-
GNIC
- Gigabit Ethernet Network Interface Card
-
GRE
- General Routing Encapsulation
-

GUI
- Graphic User Interface
-
GW
- Gateway
-
H.248
- ITU-T Rec. H.248 Gateway Control Protocol
- ITU-T H.248
H.323
- ITU-T Rec. Packet-based Multimedia Communications Systems
- ITU-T
HLR
- Home Location Register
-
HSTP
- High Signal Transfer Point
-
HTTP
- Hyper Text Transfer Protocol
-
IAD
- Integrated Access Device
-
IAM
- IP Access Module
- IP
ICMP
- Internet Control Message Protocol
-
IETF
- Internet Engineering Task Force
-
IN
- Intelligent Network
-

Glossary
INAP
- IntelligentNetworkApplicationPart
-
IP
- Internet Protocol
- Ineternet
IPDC
- Internet Protocol Device Control
-
IPv6
- IP version 6
- IP6
IS-IS
- Intermediate System-Intermediate System
-
ISDN
- IntegratedServiceDigitalNetwork
-
ISUP
- ISDN User Part
- ISDN
ITU-T
- InternationalTelecomUnion-Telecommunicationstandardization
-
IUA
- ISDN User Adaptation Layer
- ISDN Q.921
LDAP
- Lightweight Directory Access Protocol
-
LRN
- Location Routing Number
-
LS
- Local Switch
-

LSTP
- Low Signal Transfer Point
-
M3UA
- MTP3-User Adaptation layer protocol
- MTP
- Message Transfer Part 3 User adaptation layer
- 3
MAP
- Mobile Application Part
-
MCU
- Multipoint Control Unit
-
MEGACO
- Media Gateway Control
-
MFC
- Multi-Frequency Control
-
- Multiple Frequency Compelling signal
-
MG
- Media Gateway
-
MGC
- Media Gateway Controller
-
MGCP
- Media Gateway Control Protocol
-
MP
- Module Processor
-
MPLS
- Multi Protocol Label Switching
-
MPPP
- Main Processor to Peripheral Processor
-

Glossary
- Multilink Point to Point Protocol

-
MS
- Media Server
-
MSG
- Media Service Gateway
-
NAS
- Network Access Server
-
NAT
- Network Address Translation
-
NCS
- Network-Based Call Signaling Protocol
-
NGN
- Next Generation Network
-
NIC
- Network Interface Card
-
NTP
- Network Time Protocol
-
OMAP
- Operation Maintenance and Administration Part
-
OPC
- Originating point code
-
OPR
- Operator
-

OSA
- Open Service Access
-
OSPF
- Open Shortest Path First
-
PBX
- Private Branch Exchange
-
PHS
- Personal Handy-Phone System
-
PLMN
- Public Land Mobile Network
-
PPP
- Point-To-Point protocol
-
PRA
- Primary Rate Access
-
PRI
- Primary Rate Interface
-
PSTN
- Public Switch Telephone Network
-
RADIUS
- Remote Authentication Dial-In User Service
-
RIP
- Routing Information Protocol
-
RIPng
- Routing Information Protocol Next Generation for the IPv6
- IPV6

Glossary
RMON
- Remote Monitoring
-
RSVP
- Resource Reservation Protocol
-
SC
- System Control Card
-
SCCP
- Signalling Connection Control Part
-
SCN
- Switched Circuit Network
-
- Signaling Communication Network
-
SCP
- Service Control Point
-
SCTP
- Stream Control Transmission Protocol
-
SDP
- Session Description Protocol
-
SFP
- Small Form-factor Pluggable
-
SG
- Signal Gateway
-
SGCP
- Simple Gateway Control Protocol
-
SIGTRAN
- Signaling Transport
-

SIP
- Session Initiated Protocol
-
SIP-I
- SIP with encapsulated ISUP
- ISUP SIP
SIP-T
- SIP for Telephones
- SIP
SMTP
- Simple Mail Transmission Protocol
-
SNMP
- Simple Network Management Protocol
-
SoftDA
- Soft Digital Assistant
-
SOHO
- Small Office Home Office
-
SP
- Signal Point
-
SPC
- System Protocol Control (Card)
-
SPOWB
- SoftSwitch Power Box
-
SPWBF
- SoftSwitch Power Box of Fan
-
SPWBP
- SoftSwitch Power Box of Part
-
SS
- SoftSwitch
-

Glossary
SS7
- Signalling System No.7
-
SSC
- SoftSwtich System Control (Card)
-
SSN
- System Switching Network (Card)
-
SSNI
- System Switching Network Interface (Card)
-
SSP
- Service Switch Point
-
STP
- Signaling Transfer Point
-
TCAP
- Transaction Capabilities Application Part
-
TCP
- Transmission Control Protocol
-
TCP/IP
- Transmission Control Protocol/Internet Protocol
-
TDM
- Time Division Multiplexing
-
TE
- Terminal Equipment
-

TFTP
- Trivial File Transfer Protocol
-
TG
- Trunk Gateway
-
TIC
- Transport Interface Card
-
TIFF
- Tagged Image File Format
-
TTS
- Text To Speech
-
TUP
- TelephoneUserPart
-
UA
- User Agent
-
UAC
- User Agent Client
-
UAS
- User Agent Server
-
UDP
- User Datagram Protocol
-
UMS
- Unified Messaging Service
-
URL
- Uniform Resource Locator
-
URPF
- Unicast Reverse Path Forwarding
-

Glossary
VIC
- Very Important Customer
-
VOIP
- Voice Over IP
- IP
VPLS
- Virtual Private LAN Service
- LAN
VPN
- Virtual Private Network
-
VPWS
- Virtual Private Wire Services
-
WAN
- WideAreaNetwork
-
WAP
- Wireless Application Protocol
-
WWW
- World Wide Web
-

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sjzl20091120-ZXSS10 SS1b (V2.0.1.06.1) - Product Description Manual - EN

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ZXSS10 SS1b

SoftSwitch Control Equipment

Copyright © 2006 ZTE CORPORATION.

The ultimate right to interpret this product resides in ZTE CORPORATION.

Revision No. Revision Date Revision Reason

Serial Number: sjzl20091120

About This Manual.............................................. i

Purpose This manual describes the ZXSS10 SS1b SoftSwitch product in

Chapter 1, System This chapter describes the ZXSS10 SS1b

Chapter 2, System This chapter describes the ZXSS10 SS1b

Chapter 3, Interfaces This chapter describes the connection of

Chapter 4, Service This chapter describes the service provision of

Chapter 5, This chapter describes ZXSS10 SS1b in terms

Chapter 6, Technical This chapter describes ZXSS10 SS1b in terms

Related The following documentation is related to this manual:

Confidential and Proprietary Information of ZTE CORPORATION i

� ZXSS10 SS1b SoftSwitch Control Equipment Hardware De-

ii Confidential and Proprietary Information of ZTE CORPORATION

Confidential and Proprietary Information of ZTE CORPORATION 1

FIGURE 1 MULTI-LAYER ARCHITECTURE OF ZXSS10 SS1B

Standard interfaces are used for communication between layers.

2 Confidential and Proprietary Information of ZTE CORPORATION

Confidential and Proprietary Information of ZTE CORPORATION 3

multiple customer premises networks based on private net-

Core Transfer Layer

4 Confidential and Proprietary Information of ZTE CORPORATION

regard to the IPR and supports rich protocols. The platform

Confidential and Proprietary Information of ZTE CORPORATION 5

6 Confidential and Proprietary Information of ZTE CORPORATION

services can be easily introduced. Structurally, the Parlay

Confidential and Proprietary Information of ZTE CORPORATION 7

therefore protect the customer investment and system ex-

8 Confidential and Proprietary Information of ZTE CORPORATION

center flexibly and conveniently. The ZXSS10 SS1b pro-

Confidential and Proprietary Information of ZTE CORPORATION 9

� There are three service provision modes: the SoftSwitch

10 Confidential and Proprietary Information of ZTE CORPORATION

Confidential and Proprietary Information of ZTE CORPORATION 11

In the above various applications, the ZXSS10 SS1b, as the control

TABLE 2 PRODUCT STANDARDS

H.248 IETF RFC3525

MGCP IETF RFC2705

12 Confidential and Proprietary Information of ZTE CORPORATION

SIP IETF RFC3204

And related supplementary

SIP-T IETF RFC3372

SIP-I ITU-T Q.1912.5

H.323 ITU-T H.323include H.225.0, H.245,

NCS ITU-T J.169

ETSI TS 101 909-9

Confidential and Proprietary Information of ZTE CORPORATION 13

China Talecom PHS MAP Protocol

As same as the supplementary

14 Confidential and Proprietary Information of ZTE CORPORATION

NO.1 China No.1 comply international

Confidential and Proprietary Information of ZTE CORPORATION 15

This page is intentionally blank.

16 Confidential and Proprietary Information of ZTE CORPORATION

Confidential and Proprietary Information of ZTE CORPORATION 17

FIGURE 2 CONNECTION OF ZXSS10 SS1B

The description of each part is as follows:

18 Confidential and Proprietary Information of ZTE CORPORATION

FIGURE 3 ZXSS10 SS1B CABINET