00399775-MSOFTX3000 VoIP User Manual - (V100R006C05 - 02)

HUAWEI MSOFTX3000 Mobile SoftSwitch Center
V100R006C05
VoIP User Manual
Issue 02
Date 2008-04-20
Part Number 00399775
Huawei Proprietary and Confidential

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Huawei Proprietary and Confidential

VoIP User Manual Contents
Contents
About This Document.....................................................................................................................1

1 Introduction to VoIP .................................................................................................................1-1
1.1 Background.....................................................................................................................................................1-3
1.2 Basic Concepts of VoIP..................................................................................................................................1-3
1.2.1 SCTP Multi-Homing..............................................................................................................................1-4
1.2.2 BICC CMN............................................................................................................................................1-5
1.2.3 TrFO.......................................................................................................................................................1-6
1.2.4 IP/TDM Dual Bearers............................................................................................................................1-7
1.2.5 IP-based A Interface...............................................................................................................................1-8
1.2.6 IP-based Iu Interface..............................................................................................................................1-9
1.2.7 Speech Codec.......................................................................................................................................1-10
1.3 QoS Control Principle...................................................................................................................................1-11
1.4 The Networking of the VoIP Bearer Network..............................................................................................1-13
1.5 Interface Protocol Stack................................................................................................................................1-17
1.6 Network Security...........................................................................................................................................1-19
2 VoIP Planning.............................................................................................................................2-1
2.1 System Requirement.......................................................................................................................................2-2
2.2 Planning Principle...........................................................................................................................................2-3
2.2.1 Networking Planning..............................................................................................................................2-4
2.2.2 Bandwidth Planning...............................................................................................................................2-5
2.3 Traffic Planning over Each Interface for VoIP Networking Evolution..........................................................2-6
2.3.1 Calculating the IP Traffic of BICC Signaling and Number of BICC or M3UA Links over the Nc Interface
.........................................................................................................................................................................2-8
2.3.2 Calculating the IP Traffic of H.248 Signaling and Number of H.248 Links over the Mc Interface......2-9
2.3.3 Calculating the IP Traffic of BSSAP Signaling and Number of M3UA Links over the A Interface. .2-10
2.3.4 Calculating the Traffic over the Nb Interface......................................................................................2-11
2.3.5 Calculating BICC CICs........................................................................................................................2-12
2.3.6 Calculating the Numbers of Required WBSGs and WIFMs................................................................2-13
2.3.7 Calculating the Number of Required MHRUs.....................................................................................2-13
2.3.8 Calculating the Number of TDM Signaling Links...............................................................................2-14
2.3.9 Calculating the Number of E1 Circuits Required for the TDM Bearer Mode.....................................2-15
3 Newly Established VoIP Networking....................................................................................3-1
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Contents VoIP User Manual
3.1 Single-MGW Networking...............................................................................................................................3-5

3.1.1 Configuration Flow................................................................................................................................3-6
3.1.2 Interworking Data..................................................................................................................................3-7
3.1.3 Data Configuration on the MSOFTX3000.............................................................................................3-9
3.1.4 Data Configuration on the UMG8900..................................................................................................3-21
3.2 Multi-MGW Networking..............................................................................................................................3-30
3.2.1 Configuration Flow..............................................................................................................................3-31
3.2.2 Interworking Data................................................................................................................................3-32
3.2.3 Data Configuration on the MSOFTX3000...........................................................................................3-34
3.3 CMN Networking..........................................................................................................................................3-56
4 VoIP Networking Evolving from Network in Use..............................................................4-1

4.1 Single-MGW Networking...............................................................................................................................4-3
4.1.2 Interworking Data..................................................................................................................................4-6
4.1.3 Data Configuration on the MSOFTX3000.............................................................................................4-7
4.2 Multi-MGW Networking..............................................................................................................................4-16
4.3 CMN Networking..........................................................................................................................................4-28
4.3.3 Data Configuration MSOFTX3000......................................................................................................4-32
5 Routine Maintenance on the VoIP Networking.................................................................. 5-1

5.1 MSOFTX3000 Performance Entity Measurement..........................................................................................5-2
5.2 UMG8900 Performance Entity Measurement.................................................................................................5-6
6 VoIP Troubleshooting...............................................................................................................6-1
6.1 Collecting VoIP Fault Information.................................................................................................................6-4
6.2 Analyzing VoIP Fault......................................................................................................................................6-4
6.3 Troubleshooting VoIP Faults..........................................................................................................................6-5
6.4 Troubleshooting VoIP Faults Based on Alarms..............................................................................................6-6
6.5 Troubleshooting VoIP Faults Based on Performance Measurement..............................................................6-7
6.6 Troubleshooting VoIP Faults Related to Services..........................................................................................6-8
6.6.1 Call Setup Failure...................................................................................................................................6-9
6.6.2 One-Way Audio and No Audio Faults ................................................................................................6-13
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VoIP User Manual Contents
6.6.3 Speech Noise........................................................................................................................................6-14
7 Principles and Data Configuration of Bearer Network......................................................7-1

7.1 Basic Principles...............................................................................................................................................7-2
7.2 Reliable Networking Technologies for IP Network........................................................................................7-4
7.3 Example of VoIP Bearer Network Solution....................................................................................................7-5
7.3.2 Configuration Preparations....................................................................................................................7-7
7.3.3 Data Configuration.................................................................................................................................7-7
7.3.4 Configuration File................................................................................................................................7-13
8 VoIP Network Test....................................................................................................................8-1

8.1 Core Network Test..........................................................................................................................................8-2
8.1.1 IP Bearer Between Multiple MGWs (WCDMA) ..................................................................................8-2
8.1.2 IP Bearer Between Multiple MGWs (GSM) .........................................................................................8-2
8.1.3 IP-based Iu Interface..............................................................................................................................8-3
8.1.4 TDM/IP Dual Bearer..............................................................................................................................8-3
8.2 Bearer Network Test.......................................................................................................................................8-4
8.2.1 Checking System Operation When the Active Signaling Link of MSC Server Is Faulty......................8-5
8.2.2 Checking System Operation When the Standby Signaling Link of MSC Server Is Faulty...................8-5
8.2.3 Checking System Operation When the Active Signaling Link of MGW Is Faulty...............................8-6
8.2.4 Checking System Operation When the Standby Signaling Link of MGW Is Faulty.............................8-6
8.2.5 Checking System Operation When the Active Media Link of MGW Is Faulty....................................8-7
8.2.6 Checking System Operation When the Standby Media Link of MGW Is Faulty..................................8-7
8.2.7 Checking System Operation When CE1 Is Faulty.................................................................................8-8
8.2.8 Checking System Operation When CE2 Is Faulty.................................................................................8-8
8.2.9 Checking System Operation When the Link Between CE1 and CE2 Is Faulty.....................................8-9
8.2.10 Checking System Operation When the Link Between CE1 and AR1 Is Faulty..................................8-9
8.2.11 Checking System Operation When the Link Between CE2 and AR2 Is Faulty................................8-10
9 FAQ...............................................................................................................................................9-1
Index.................................................................................................................................................i-1
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VoIP User Manual Figures
Figures
Figure 1-1 The paths related to the SCTP association ........................................................................................1-4

Figure 1-2 Principles of the TrFO function..........................................................................................................1-7
Figure 1-3 Protocol stack of the A interface .......................................................................................................1-8
Figure 1-4 Protocol stack of the Iu interface .......................................................................................................1-9
Figure 1-5 IP CAC principle..............................................................................................................................1-11
Figure 1-6 MPLS principle.................................................................................................................................1-12
Figure 1-7 Jitter Buffer principle........................................................................................................................1-13
Figure 1-8 Local router networking mode..........................................................................................................1-14
Figure 1-9 Remote router networking mode......................................................................................................1-15
Figure 1-10 E1/POS access method...................................................................................................................1-16
Figure 1-11 GE/FE access method.....................................................................................................................1-16
Figure 1-12 Protocol stack of the Nc interface ..................................................................................................1-17
Figure 1-13 Protocol stack of the Mc interface..................................................................................................1-18
Figure 1-14 Protocol stack of the Nb interface..................................................................................................1-18
Figure 1-15 Protocol stack of the A interface ...................................................................................................1-19
Figure 1-16 Protocol stack of the Iu interface ...................................................................................................1-19
Figure 2-1 Dual plane networking........................................................................................................................2-5
Figure 2-2 Bearer network access model.............................................................................................................2-6
Figure 3-1 VoIP networking structure..................................................................................................................3-1
Figure 3-2 Procedure of the VoIP data configuration..........................................................................................3-3
Figure 3-3 Single-MGW VoIP networking..........................................................................................................3-5
Figure 3-4 Procedure of the data configuration on the single-MGW VoIP networking......................................3-7
Figure 3-5 Multi-MGW VoIP networking.........................................................................................................3-30
Figure 3-6 Procedure of the data configuration on the multi-MGW VoIP networking.....................................3-32
Figure 3-7 CMN VoIP networking....................................................................................................................3-57
Figure 3-8 Procedure of the data configuration on the CMN VoIP networking................................................3-58
Figure 4-1 Single-MGW networking...................................................................................................................4-3
Figure 4-2 Single-MGW VoIP networking..........................................................................................................4-4
Figure 4-3 Flow of the data configuration for the single-MGW VoIP networking.............................................4-6
Figure 4-4 Multi-MGW networking...................................................................................................................4-16
Figure 4-5 Multi-MGW VoIP networking.........................................................................................................4-17
Figure 4-6 Flow of the data configuration for the multi-MGW VoIP networking............................................4-19
Figure 4-7 TDM networking..............................................................................................................................4-29
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Figures VoIP User Manual
Figure 4-8 CMN networking..............................................................................................................................4-30

Figure 4-9 Flow of the data configuration for the CMN networking mode.......................................................4-31
Figure 6-1 Troubleshooting procedure ................................................................................................................6-2
Figure 7-1 Networking of the VoIP Bearer Network...........................................................................................7-6
Figure 8-1 Access bearer network of the core network........................................................................................8-4
vi Huawei Proprietary and Confidential Issue 02 (2008-04-20)

VoIP User Manual Tables
Tables
Table 1-1 The active and standby transmission paths and priority configuration................................................1-4
Table 2-1 Input information required for traffic planning over each interface....................................................2-7
Table 3-1 Office information data (single-MGW)...............................................................................................3-7
Table 3-2 Interworking data between the MSC server and other NEs (single-MGW)........................................3-8
Table 3-3 Office information data (multi-MGW)..............................................................................................3-32
Table 3-4 Interworking data between the MSC server and other NEs (multi-MGW).......................................3-33
Table 3-5 Interworking data between the MSC server and other NEs (CMN)..................................................3-59
Table 4-1 Interworking data of single-MGW VoIP networking..........................................................................4-7
Table 4-2 Interworking data of single-MGW VoIP networking........................................................................4-20
Table 4-3 Interworking data in the CMN networking mode..............................................................................4-32
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VoIP User Manual About This Document
About This Document
Purpose
This document describes the MSOFTX3000 Mobile SoftSwitch Center (short for
MSOFTX3000) VoIP networking, including the basic principles of VoIP, VoIP planning, newly
established VoIP networking, VoIP networking evolved from network in use, routine
maintenance on the VoIP networking, VoIP troubleshooting, and principles, and data
configuration of bearer network.
This document also provides procedures of the basic configuration and VoIP troubleshooting.
Related Versions
The following table lists the product versions related to this document.
Product Name Version
MSOFTX3000 V100R006C05
Intended Audience
The intended audiences of this document are:
l Mobile network maintenance engineers

l Marketing technical engineers
l Telecommunications management staff
Organization
This document consists of nine sections and is organized as follows.
Issue 02 (2008-04-20) Huawei Proprietary and Confidential 1

About This Document VoIP User Manual
Chapter Description
1 Introduction to VoIP This section describes the concepts and basic principles related
to VoIP.
2 VoIP Planning This section describes the VoIP specifications, QoS of the bearer
network and planning principle.
3 Newly Established This section describes certain types of typical VoIP networking
VoIP Networking and related data configuration that must be performed for newly
established VoIP networking.
4 VoIP Networking This section describes related data configurations that must be
Evolving from Network performed for VoIP networkings evolved from typical TDM
in Use networkings in use.
5 Routine Maintenance This section describes VoIP-related measurement units that the
on the VoIP Networking UMG8900 and the MSOFTX3000 must measure during the
routine maintenance.
6 VoIP Troubleshooting This section describes the basic methods of rectifying the VoIP
faults.
7 Principles and Data This section describes the principles and data configuration of
Configuration of Bearer the VoIP bearer network.
Network
8 VoIP Network Test This section describes the tests related to the IP bearer networks.
9 FAQ This section describes FAQs related to the IP bearer networks.
Conventions
Symbol Conventions
The following symbols may be found in this document. They are defined as follows.
Symbol Description
Indicates a hazard with a high level of risk which, if not

avoided, will result in death or serious injury.
DANGER
Indicates a hazard with a medium or low level of risk which,

if not avoided, could result in minor or moderate injury.
WARNING
Indicates a potentially hazardous situation that, if not

avoided, could cause equipment damage, data loss, and
CAUTION
performance degradation, or unexpected results.
2 Huawei Proprietary and Confidential Issue 02 (2008-04-20)

VoIP User Manual About This Document
Symbol Description
TIP Indicates a tip that may help you solve a problem or save
your time.
NOTE Provides additional information to emphasize or

supplement important points of the main text.
General Conventions
Convention Description
Times New Roman Normal paragraphs are in Times New Roman.
Boldface Names of files, directories, folders, and users are in boldface. For
example, log in as user root.
Italic Book titles are in italics.

Courier New Terminal display is in Courier New.
Command Conventions
Boldface The keywords of a command line are in boldface.
Italic Command arguments are in italic.
[] Items (keywords or arguments) in square brackets [ ] are

optional.
{ x | y | ... } Alternative items are grouped in braces and separated by

vertical bars. One is selected.
[ x | y | ... ] Optional alternative items are grouped in square brackets

and separated by vertical bars. One or none is selected.
{ x | y | ... } * Alternative items are grouped in braces and separated by

vertical bars. A minimum of one or a maximum of all can
be selected.
GUI Conventions
Boldface Buttons, menus, parameters, tabs, window, and dialog titles are in
boldface. For example, click OK.
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About This Document VoIP User Manual
> Multi-level menus are in boldface and separated by the ">" signs. For
example, choose File > Create > Folder.
Keyboard Operation
Format Description
Key Press the key. For example, press Enter and press Tab.
Key 1+Key 2 Press the keys concurrently. For example, pressingCtrl+Alt+A means the
three keys should be pressed concurrently.
Key 1, Key 2 Press the keys in turn. For example, pressing Alt, A means the two keys
should be pressed in turn.
Mouse Operation
Action Description
Click Select and release the primary mouse button without moving the pointer.
Double-click Press the primary mouse button twice continuously and quickly without
moving the pointer.
Drag Press and hold the primary mouse button and move the pointer to a certain
position.
Update History
Updates between document versions are cumulative. Therefore, the latest document version
contains all updates made to previous versions.
Updates in Issue 02 (2008-04-20)

Second release.
The description of traffic planning over each interface for VoIP networking evolution is added.
Updates in Issue 01 (2008-02-25)

Initial field trial release.
4 Huawei Proprietary and Confidential Issue 02 (2008-04-20)

VoIP User Manual 1 Introduction to VoIP
1 Introduction to VoIP
About This Chapter
This section describes the concepts and basic principles related to Voice over Internet Protocol
(VoIP).
1.1 Background
With the development of mobile communication and the increase of mobile subscriber numbers,
various mobile services are emerging. To meet the large capacity and low cost demand from
operators and follow the trend in the development of the carrier-class multimedia services, the
VoIP technology is introduced to the WCDMA CS domain. This technology enables the
transmission of voice services over the IP network in architecture of independent bearer and
control modules.
1.2 Basic Concepts of VoIP
The implementation of the VoIP technology depends on many features, such as SCTP multi-
homing, BICC CMN, TrFO, IP/TDM dual bearer, IP-based A interface, IP-based Iu interface,
and speech codec .
1.3 QoS Control Principle
The performance of the IP network is less satisfactory than that of the TDM network in the delay,
jitter and packet drop rate. These performance indexes are closely related to the voice quality.
When the delay, jitter or packet drop rate exceeds the normal level, the voice quality is affected
or even the call is disconnected. Therefore, the QoS (Quality of Service) control is very important
to the VoIP.
1.4 The Networking of the VoIP Bearer Network
The VoIP network introduces the IP bearer network to the traditional mobile network. The voice
data can be transmitted over the IP network to save the cost. A typical IP bearer network can
adopt the networking mode using routers to transmit voice data. The router networking mode
consists of the local router networking mode and remote router networking mode.
1.5 Interface Protocol Stack
In the VoIP network, the following network interfaces are involved in IP applications: Nc
interface, Mc interface, Nb interface, A interface, and Lu interface.
1.6 Network Security
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1 Introduction to VoIP VoIP User Manual
After the IP bearer network is introduced, you must pay attention to the security issue of the
mobile network.
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1.1 Background
With the development of mobile communication and the increase of mobile subscriber numbers,
various mobile services are emerging. To meet the large capacity and low cost demand from
operators and follow the trend in the development of the carrier-class multimedia services, the
VoIP technology is introduced to the WCDMA CS domain. This technology enables the
transmission of voice services over the IP network in architecture of independent bearer and
control modules.
Compared with the TDM technology used for bearing voice services, the VoIP technology has
the following advantages:
l The VoIP technology can save network bandwidth by using Voice Activity Detection
(VAD), compressed speech codec, and IP statistical-multiplexing technology. The VoIP
networking architecture is favorable for the flatten networking.
l The VoIP technology can serve both realtime services and non-realtime services.
l The cost of the IP network is higher than that of the TDM network during initial stage for
constructing the VoIP network. When the network scale is enlarged, the advantages of the
IP network are more obvious. With the application of advanced technologies, such as, VAD
and Transcoder Free Operation (TrFO), the network cost can be reduced greatly.
1.2 Basic Concepts of VoIP

The implementation of the VoIP technology depends on many features, such as SCTP multi-
homing, BICC CMN, TrFO, IP/TDM dual bearer, IP-based A interface, IP-based Iu interface,
and speech codec .
1.2.1 SCTP Multi-Homing

The SCTP multi-homing management can associate multiple IP transmission paths by setting
up SCTP associations between multiple reachable IP addresses on the local and peer ends. In
this way, the system can provide reliable network-class transmission services.
1.2.2 BICC CMN
If a node supports the BICC signaling and it provides related bearer control functions, this node
is called a service node. If the node cannot provide bearer control functions, this node is called
a call mediation node (CMN). In the actual networking, a site can serve as a service node and a
CMN at the same time.
1.2.3 TrFO
The audio codec can cause lossy compression during voice encoding and decoding. The voice
quality is reduced and transmission delay is extended during speech encoding and decoding.
TrFO can be used to process voice without using the TC during voice transmission. This function
can provide a peer to peer voice transmission of high fidelity and low transmission delay.
1.2.4 IP/TDM Dual Bearers
The all-IP network evolving process refers to the process that the TDM network evolves to the
IP network. The IP network and TDM network coexist in this process. The inter-MSC and intra-
MSC TDM/IP dual bearers help to implement smooth evolution from a TDM network to an IP
network.
1.2.5 IP-based A Interface
The IP-based A interface is essential to all IP network construction.

1.2.6 IP-based Iu Interface

The IP-based Iu interface refers to that the Iu interface adopts IP transmission. After IP-based
Iu interface is set up, the signaling and bearer plane between the CN and RNC can transmit data
through the IP bearer.
1.2.7 Speech Codec
Because the different protocols and bearer methods are used between different networks, the
MGW needs to convert the format of the services streams with speech codec. In the VoIP
network, the UMG8900 supports the Adaptive Multi-Rate (AMR) and G.711 at present.
1.2.1 SCTP Multi-Homing

The SCTP multi-homing management can associate multiple IP transmission paths by setting
up SCTP associations between multiple reachable IP addresses on the local and peer ends. In
this way, the system can provide reliable network-class transmission services.
The SCTP multi-homing management supports two or four SCTP links in one SCTP association
at present. To ensure the interface stability, the configuration of two SCTP links in one SCTP
association is preferred. The following section describes the configuration of two SCTP links.
After the SCTP association is established, you must define a preferred path for each SCTP end
point. This path is used for transmitting SCTP packets in normal cases. At the receiving end, the
multi-homing path management function verifies whether the SCTP association for the incoming
SCTP packets is configured before processing the SCTP packet. The paths that are related to
the SCTP association when the dual planes of the SCTP multi-homing are enabled are shown
in Figure 1-1.
Figure 1-1 The paths related to the SCTP association

Active path
IP1 IP3
Local Peer
end Standby path end
IP2 IP4
The preferred path uses the first IP address in the IP address list of both the local and peer end.
The preferred path of the SCTP association at the client uses the first IP address in the destination
IP address list as the destination IP address and the first IP address in the local IP address list as
the local IP address. After the server receives the primitive through the SCTP association, you
must set the source and destination IP addresses of the preferred path of the SCTP association.
The transmission path is selected based on the priority of the available paths. The path with the
highest priority is selected as the preferred path. The rest paths are selected in an ascending order
based on the serial numbers of the local addresses. Table 1-1 shows the active and standby
transmission paths and priority configuration.
Table 1-1 The active and standby transmission paths and priority configuration
Path Active and Standby Path Priority
[IP1,IP3] Active High
[IP2,IP4] Standby Low

Note the following for transmission paths:

l When the current transmission path is unavailable, the path with the highest priority is
selected from available paths. If no available path is found and the current transmission
path remains unchanged, the transmission link is disconnected.
l When a transmission path with higher priority than the current path is available, the
transmission path with higher priority is adopted.
1.2.2 BICC CMN

If a node supports the BICC signaling and it provides related bearer control functions, this node
is called a service node. If the node cannot provide bearer control functions, this node is called
a call mediation node (CMN). In the actual networking, a site can serve as a service node and a
CMN at the same time.
The BICC CMN supports the following functions:
l The CMN function is controlled by the license.
The license can be used to specify whether the MSOFTX3000 supports the CMN function.
If the incoming and outgoing office directions for a call are BICC office directions, and the
license for supporting the BICC CMN function is enabled, the system can check whether
the call is a BICC CMN call. If the license is disabled, the call cannot be a BICC CMN
call.
l The number of the BICC CMN calls is controlled by the license.
The license can be used to control the maximum number of BICC CMN calls that is
supported by the MSOFTX3000. If the number of BICC CMN calls in the system reaches
the maximum number specified by the license, the subsequent BICC CMN calls are
disconnected.
l The CMN function can be configured in the sub-route table.
When the incoming and outgoing office directions for a call are BICC office directions,
and the license for supporting the BICC CMN function is enabled, the system can judge
whether a call is a BICC CMN call based on the preferred CMN configuration in the sub-
route table.
l The BICC CMN call release function defined in ITU-T1902.4 is supported.
According to ITU-T1902.4, the system must transparently transmit the received REL and
RLC messages during a BICC CMN call. When receiving the REL message, the system
cannot release the call in call segments immediately. The system can release the call
resources only after it receives the inter-office RLC message.
l The COT message processing function defined in ITU-T1902.4 is supported.
According to ITU-T1902.4, the system must transparently transmit the call continuity check
indicator in the received IAM messages during a BICC CMN call. The system cannot
activate the T8 timer. When receiving a COT message, the CMN must send the COT
message to the backward office.
l The reset message processing function defined in ITU-T1902.4 is supported.
The system must transparently transmit the received RSC and RLC messages during a
BICC CMN call. When the system receives the RSC message indicating the BICC CIC
reset, the system must send the RSC message to the peer office. When the BICC CMN
receives the RLC message from the peer office, the BICC CMN sends the RLC message
as the response of the RSC message.
l The processing of APP with the length over 255 bytes is supported.

During a BICC CMN call, the system supports the call that is configured with Application
Transport Parameter (APP) of the length over 255 bytes.
l The performance measurement features related to the BICC CMN are supported.
To enable performance measurement for BICC CMN calls, the BICC CMN performance
measurement unit is added to the system. In addition, the indexes related to BICC CMN
calls are added to the performance measurement units for incoming/outgoing trunk office
direction, trunk office direction and CPU utilization.
l The rerouting function upon failure is supported.
During a BICC CMN call, the previously selected sub-routes are excluded when the system
reselects the routes based on the cause value of routing failure.
l The calling number normalization function is controlled by the related software parameters.
During a BICC CMN call, the system determines whether to normalize the calling number
in the international number format (8613**) into a calling number in the national number
format (13**).
NOTE
The BICC trunk groups must be the CMN trunk groups when the MSOFTX3000 supports the CMN
function, or the CMN function will be failed.
1.2.3 TrFO
The audio codec can cause lossy compression during voice encoding and decoding. The voice
quality is reduced and transmission delay is extended during speech encoding and decoding.
TrFO can be used to process voice without using the TC during voice transmission. This function
can provide a peer to peer voice transmission of high fidelity and low transmission delay.
On the legacy radio network (access network) side, AMR-coded signals are transmitted at the
highest rate of 12.2 kbit/s; on the legacy core network side, signals are transmitted at the rate of
64 kbit/s. To ensure successful signal processing, transmission rate must be adapted through a
transcoder. During coding/decoding, the transcoding operations of the voice code decrease the
voice quality, and the transmission delay is increased. In the UMTS network, the transcoding
operations performed through a transcoder are not required. Therefore, the voice quality is
improved because both the UE and the network support the end-to-end calls. The Out-of-Band
Transcoder Control (OoBTC) function is applicable to both the calls between the mobile
networks and the calls between the mobile network and the other networks. The system maintains
the consistency between the codec of the caller and the codec of the callee by using the OoBTC
function, which is called the TrFO feature.
When a call is set up, TrFO negotiates the coding and decoding rules on both sides and adopts
the coding and encoding type and mode that are supported on both sides. In this way, the TC's
processing is not required any more during voice transmission. Figure 1-2 shows the principles
of the TrFO function.

Figure 1-2 Principles of the TrFO function
(3)
(1)
(1) (2)
(1) (2)
Caller Callee
1. The caller sends the speech codec algorithms supported to the called MSC.
2. The callee sends the speech codec algorithms supported to the called MSC.
3. The called MSC compares the speech codec algorithms provided by both sides and selects
the codec type and mode that are supported on both sides. Then the called MSC sends the
information as the final negotiation result to the calling MSC.
4. The caller and callee start voice transmission based on the codec type and mode that is
provided through negotiation.
1.2.4 IP/TDM Dual Bearers

The all-IP network evolving process refers to the process that the TDM network evolves to the
IP network. The IP network and TDM network coexist in this process. The inter-MSC and intra-
MSC TDM/IP dual bearers help to implement smooth evolution from a TDM network to an IP
network.
Inter-MSC IP/TDM Dual Bearers

The inter-MSC IP/TDM dual bearers refers to that the system supports both the IP bearer network
and TDM bearer network in one office direction. That requires the system to support broadband
BICC trunk and narrowband ISUP trunk in the same office direction.
The inter-MSC IP/TDM dual bearers have the following policies on selecting the bearer type:
l The system selects the bearer type based on the call priority. The system selects the TDM
trunk group in preference to provide good conversation quality for the subscribers with
high priority.
l The system selects the bearer network based on the type of incoming bearer network. If the
bearer type of the outgoing trunk group is consistent with the bearer type of the incoming
trunk group carried by the route selection message, the transmission efficiency is higher.
l The system selects the outgoing bearer type based on the intra-office bearer type. If the
bearer type of the outgoing trunk group is consistent with the intra-MSC bearer type
(required for gateway path selection), the transmission efficiency is higher.
l The system selects the outgoing bearer type based on the IP priority percentage and TDM
priority percentage. The sum of the IP priority percentage and TDM priority percentage
must be 100%.

Intra-MSC IP/TDM Dual Bearers

The intra-MSC IP/TDM dual bearers refers to that the system supports both the IP bearer network
and TDM bearer between different MGWs under the same MSC server.
In the case of intra-office hybrid networking, a bearer type can be selected between TDM and
IP based on the configured percentages or preference policy when the system processes a service.
The intra-MSC IP/TDM dual bearers have the following route selection policies:
l In the case of preference-based bearer selection, the preferred bearer type is chosen in
preference for the service.
l In the case of percentage-based bearer selection, a bearer is chosen based on the configured
percentages of the two bearer types and the comparison between the actual selection times
of the two bearer types.
The preference-based bearer selection policy is higher in priority than the percentage-based
selection mode. Namely, if both the preference-based bearer selection policy and the percentage-
based bearer selection exist, the preference-based bearer selection policy is used.
To improve dual backup between the two bearer types and reduce call loss in inter-MGW bearer
selection, the other bearer type will be selected automatically for call connection, if it is found
that a TDM bearer cannot be selected according to, for example, the TDM circuit status, or if
an IP bearer cannot be selected according to, for example, the IP QoS flow control indication.
1.2.5 IP-based A Interface

The IP-based A interface is essential to all IP network construction.
The IP-based A interface refers to the IP-based user plane. The IP-based protocol of the signaling
plane is supported. The general scheme is as follows:
l The BSC does not provide the TC function. The Transcoder & Rate Adaptation Unit
(TRAU) frames are terminated directly. The TC is moved to the MGW.
l The A interface does not support the protocol stack of the user plane (UP).
l The 2G GSM single rate codec of the Nb interface uses the UP transparent mode.
Figure 1-3 shows the protocol stack used when the data on the A interface is transmitted through
the IP bearer network.
Figure 1-3 Protocol stack of the A interface
BSSAP
SCCP
M3UA RTP
SCTP UDP
IP IP
Signaling plane Bearer plane
The IP-based A interface has following advantages:

l The peer to peer TrFO can be provided. No TC is configured on the call link.
l Transmission resources are saved: The IP network uses statistical multiplexing. During
accessing, the bandwidth allocation has no granularity size limitation as the TDM and can
be used based on your requirements. When the compressed codec is transmitted, the IP
based A interface function can efficiently reduce the occupancy of the bandwidth.
l Maintenance cost is reduced: After the core network is operating over IP bearer network,
the IP based A interface and IP based BSS enable the types of networks to be maintained
to normalize to one type. This can decrease the technical requirements of maintenance
personnel and reduce the operation expenditure (OPEX).

The IP-based Iu interface refers to that the Iu interface adopts IP transmission. After IP-based
Iu interface is set up, the signaling and bearer plane between the CN and RNC can transmit data
through the IP bearer.
For the IP-based Iu interface to be successfully implemented, the RNC and MGW must support
the features of the IP-based Iu interface. This feature does not support the scenario of the mixed
networking of Mini-flex ATM and IP networking.
When the IP-based Iu interface is set up, the MSC server, RNC and MGW provide the following
functions:
l The MSC server exchanges the IP addresses of the RNC and MGW between the RNC and
the MGW. The MSC server also sets up links with the signaling plane of the RNC.
l The RNC sets up links with the signaling plane of the MSC server. The RNC also allocates
radio resources and sets up the bearer plane with the MGW.
l The MGW needs to obtain the IP address of the RNC and UDP port number during the UP
initiation. The MGW can send the IP packets from the RNC successfully.
Figure 1-4 shows the protocol stack when the data on the Iu interface is transmitted through the
IP bearer network.
Figure 1-4 Protocol stack of the Iu interface
RANAP
SCCP IuUP
M3UA RTP
SCTP UDP
IP IP
When the IP-based Iu interface is set up, the networking between the MSC server and RNC and
that between the MSC server and MGW is changed. The networking is as follows:
l The RANAP supports IP transmission. The RNC and MSC server are connected directly.
l The RANAP supports IP transmission. The MGW and MSC server are connected directly.

l The RANAP supports the mixed networking using ATM and IP bearer network.
1.2.7 Speech Codec

Because the different protocols and bearer methods are used between different networks, the
MGW needs to convert the format of the services streams with speech codec. In the VoIP
network, the UMG8900 supports the Adaptive Multi-Rate (AMR) and G.711 at present.
AMR
The AMR supports voice transmission of different rates. There are eight compression methods.
The corresponding rates are as follow.
Speech Codec Source Codec Bit Rate
AMR_12.20 12.20 kbit/s (GSM EFR)
AMR_10.20 10.20 kbit/s
AMR_7.95 7.95 kbit/s
AMR_7.40 7.40 kbit/s (IS-641)
AMR_6.70 6.70 kbit/s (PDC-EFR)
Besides the eight transmission rates, the AMR provides functions including SCR, VAD, CNG,
and lost frame compensation. In this way, the AMR can ensure the speech quality with various
methods.
G.711
The voice signal is transmitted as the analog signal in the digital communication system. This
process is composed of three steps: sampling, quantizing and coding. The signal obtained
through sampling is dispersed in the time line. The amplitude of the signal is continuous. To
convert the signal into the digital signal, the sampling values must be converted into the dispersed
values. The amplitude with infinite continuous signal can be converted into finite dispersed
signal. This is the cause of quantizing. The methods for quantizing can be classified into linear
quantizing and non-linear quantizing.
The G.711 standard provides a non-linear quantizing method, that is, to use unequal quantizing
difference (interval). For the signal with small amplitude difference in dense classification, use
small difference for quantizing. For the signal with great amplitude difference in scattered
classification, use great difference for quantizing. The interval of quantizing is decreased when
the signal amplitude decreases. The signal-to-noise ratio of the signal amplitude within wide
dynamical range can meet the requirements.

1.3 QoS Control Principle

The performance of the IP network is less satisfactory than that of the TDM network in the delay,
jitter and packet drop rate. These performance indexes are closely related to the voice quality.
When the delay, jitter or packet drop rate exceeds the normal level, the voice quality is affected
or even the call is disconnected. Therefore, the QoS (Quality of Service) control is very important
to the VoIP.
The QoS covers the unidirectional transmission delay, packet drop rate and jitter.
The major problem that causes the decrease of QoS can be:
l The IP bearer network is faulty. (Faulty equipment, ports, transmission connections, or

logical links are found.)
l The load of the IP bearer network is unbalanced.
The system needs to perform QoS control concerning equipment, services, subscribers, and
reducing jitter.
Equipment-Based QoS Assurance Technology - CAC
The Connection Admission Control (CAC) can restrict the subscriber access based on the total
number of registrations, total number of calls and call rate. When the total number of the
registrations, total number of calls or call rate reaches the threshold, the system restricts
subscriber access to the network. The system discards packets and prohibits the subscriber from
originating an access request. For the subscriber who has already originated the access request,
its service is not affected. When the total number of registered subscribers, total number of calls,
subscriber registration rate, or call rate returns to the normal level, the system allows the
subscribers to connect the network.
The CAC technology can be used to control the network load, reduce network damage, and
ensure QoS in the mobile network. Figure 1-5 shows the CAC principle.
Figure 1-5 IP CAC principle
(4) (1)
(4)
MSC Server MSC Server
(2) (3) (2) (3)
MGW MGW

1. The MSC server maps the allowed transmission bandwidth to the BICC CIC based on
bandwidth planning. Then the MSC server starts to control the amount of transmitted data.
2. The MSC server sends the quality detection event to the MGW. When the MGW detects
that the data traffic exceeds the threshold, the new data transmission request is rejected.
When the data traffic returns to normal, the system permits data transmission requests.
3. The MGW reports the quality detection event to the MSC server.
4. The MSC server selects the routes and conducts flow control based on the quality detection
result reported by the MGW.
Priority-Based QoS Assurance Technology

This technology is used to sort data based on subscribers and service types and set different
priorities for them. The transmission quality for the service with the high priority is assured. The
service quality of the subscribers with the high priority and important data is assured.
The priority-based QoS assurance technology uses the Multiprotocol Label Switching (MPLS).
The MPLS technology integrates the IP switching and ATM switching principles. It can be used
to sort the groups with the same forwarding method into one group. The system can sort data
based on source address, destination address, source port, destination port, protocol type, and
VPN. The groups are identified with the fixed labels. The label forwarding table is created on
the label forwarding router. When the router at the entrance of the MPLS network receives a
packet, the router determines the forwarding method and adds a label to this packet. In this way,
a packet with the MPLS label is created. The label forwarding router matches the label contained
in the packet with that in the label forwarding table. Then, the router exchanges the labels and
forwards the packets. Figure 1-6 shows the principle.
Figure 1-6 MPLS principle
MGW BR BR MGW
CR CR
IP Package MPLS Label IP Package
Jitter Reduction - Based QoS Assurance - Jitter Buffer Technology

In the IP network, the system selects the transmission path for the IP packets based on the network
status. Therefore, the IP packets cannot be sent to the receiving end based on the sending
sequence. The neighboring IP packets cannot be received at the same interval, thus resulting in
delay and jitter, and further affecting the voice quality. In Jitter Buffer mode, a jitter buffer is
added at the receiving end. When the packet arrives at the jitter buffer, the system does not
forward the packet to the receiving end. After a certain interval, the packets in the jitter buffer
are sent to the receiving end based on the sequence in which the packets are constructed before
sending. In this way, the jitter is reduced and the original sequence of the packets is maintained.
Figure 1-7 shows the principle.

Figure 1-7 Jitter Buffer principle
20ms 20ms 20ms 20ms 20ms 20ms
A B C A B C A B C
RTP time
10 30 50 10 30 50 10 30 50
stamp
Jitter Buffer
1.4 The Networking of the VoIP Bearer Network

The VoIP network introduces the IP bearer network to the traditional mobile network. The voice
data can be transmitted over the IP network to save the cost. A typical IP bearer network can
adopt the networking mode using routers to transmit voice data. The router networking mode
consists of the local router networking mode and remote router networking mode.
Local Router Networking Mode

In this networking mode, all the core network components are configured with routers locally.
The core network components are connected to the proximate routers directly through FE/GE
(Fast Ethernet / Gigabit Ethernet). The core network components are then interconnected through
the IP bearer network composed of the routers. Figure 1-8 shows the networking.

Figure 1-8 Local router networking mode
PSTN RNC
Remote MGW Router

RNC equipment Central
room A equipment
Router Router room
IP
Router MSC
Server
Router
RNC Router
FE
RNC E1
MGW
Remote POS/E1/GE
equipment room B GE
PSTN
The advantages and disadvantages of the local router networking mode are as follows:
l Advantages
The network can be expanded easily to provide multiple services.
The network can be evolved to the future IMS network seamlessly.
The IP bearer network has powerful access capability and provides a number of IP
interfaces. The core network components can connect the proximate IP bearer network
easily.
This mode is powerful for large scale network.
l Disadvantages
The Capital Expenditure (CAPEX)/OPEX is high.
The QoS, reliability and security technologies for this network are complex.
The local router networking mode can be adopted in the following conditions:
l The network has large capacity.
l Many network elements are configured. (More than 10 MSC servers and 50 MGWs are
configured.)
l Multiple services are provided.
l The all IP network construction is required.
Remote Router Networking Mode

In this networking mode, the E1s/Passive Optical Splitters (POS) of core network components
are connected to the SDH transmission network directly. The core network components are

connected to the remote routers through the SDH transmission network. The FE/GE interfaces
of core network components are connected to the SDH transmission network through Multi-
Service Transmission Platform (MSTP), and then connected to the remote routers. The core
network components are interconnected through the remote routers. Figure 1-9 shows the
networking.
Figure 1-9 Remote router networking mode
Remote
RNC equipment MGW
room A
Central
equipment
room
Remote
equipment SDH
room B
RNC MSC
Server
Router
RNC
FE
RNC E1
MGW POS/E1
Remote
equipment room
C
In this networking mode, the access methods are classified as E1/POS access method and GE/
FE access method.
l E1/POS access method
When the E1/POS access method is used, the core network components and routers must
provide E1/POS interfaces. The core network components and routers are connected
through the SDH transmission network directly. Figure 1-10 shows the E1/POS access
method.

Figure 1-10 E1/POS access method
SDH
Router
MGW
SDH
Router
MGW
E1
POS
When the E1 access method is used, the core network components must support the IP
over E1 feature. The router provides a few E1 ports, so the access capability is limited.
When the POS access method is used, the network is costly because the POS interface
is expensive.
l GE/FE access method
When the GE/FE access method is used, the MSTP equipment is provided in the SDH
transmission network. The GE/FE interfaces of core network components are connected to
the SDH transmission network through the MSTP equipment. The routers are also
connected to the SDH transmission network through the MSTP equipment. In this access
method, the interfaces of routers and core network components are less expensive and the
TDM resources of the SDH transmission network can be used efficiently. Figure 1-11
shows the GE/FE access method.
Figure 1-11 GE/FE access method
SDH
MGW MSTP MSTP Router
GE/FE
E1/POS
The advantages and disadvantages of the remote router networking mode are as follows:
l Advantages
The network can be expanded easily to provide multiple services.
The network can be upgraded to the distributed IP bearer network seamlessly.

The CAPEX/OPEX is low.

The QoS, reliability and security technologies for this network are simple.
l Disadvantages
The NE must be connected to the SDH transmission network through the Ethernet. The
MSTP equipment or the router is required to provide the POS/E1 interface.
The central router provides limited interfaces, so it cannot support a large scale network.
The remote router networking mode can be adopted in the following conditions:
l The network capacity is small.

l The central router can provide sufficient IP interfaces for the networking.
1.5 Interface Protocol Stack

In the VoIP network, the following network interfaces are involved in IP applications: Nc
interface, Mc interface, Nb interface, A interface, and Lu interface.
Nc Interface
The Nc interface is used to transmit data between the following NEs:
l MSC server and MSC server

l GMSC server and MSC server
l CMN and (G)MSC server
l CMN and CMN
The Nc interface complies with the BICC protocol. Figure 1-12 shows its protocol stack when
the Nc interface is used in the IP network.
Figure 1-12 Protocol stack of the Nc interface
BICC
M3UA
SCTP
IP
Mc Interface
The Mc interface is used to transmit data between the following NEs:
l MSC server and MGW

l GMSC server and GMGW
The Mc interface complies with the H.248 protocol. Figure 1-13 shows its protocol stack when
the Mc interface is used in the IP network.

Figure 1-13 Protocol stack of the Mc interface
H.248
SCTP
IP
Nb Interface
The Nb interface is used to transmit data between the following NEs:
l MGW and MGW
l MGW and GMGW
The Nb interface complies with RTP. When the Nb interface is used in the IP network, the speech
codec uses AMR2 12.2 kbit/s. Figure 1-14 shows its protocol stack.
Figure 1-14 Protocol stack of the Nb interface
NbUP
RTP
UDP
IP
When the softswitching network uses the IP network to bear the voice data, the flatten networking
mode is used at the bearer layer. The media stream between any MGWs is transmitted over the
private IP bearer network directly. The media stream is not forwarded by any other mediate
MGW.
A Interface
The A interface is used to transmit data between the following NEs:
l BSC and MSC
l BSC and MGW
l BSC and MSC server
The A interface complies with the BSSAP protocol. Figure 1-15 shows its protocol stack when
the A interface is used in the IP network.

Figure 1-15 Protocol stack of the A interface
BSSAP
SCCP
M3UA RTP
SCTP UDP
IP IP
Iu Interface
The Iu interface is used to transmit data between the following NEs:
l RNC and MSC server

l RNC and MGW
The Iu interface complies with the RANAP protocol. Figure 1-16 shows its protocol stack when
the Iu interface is used in the IP network.
Figure 1-16 Protocol stack of the Iu interface
RANAP
SCCP IuUP
M3UA RTP
SCTP UDP
IP IP
1.6 Network Security

After the IP bearer network is introduced, you must pay attention to the security issue of the
mobile network.
To ensure the security of the mobile services, the mobile signaling, traffic and gateway must use
the private IP network. The private IP network must be isolated by the firewall.
The firewall can be used for network isolation. Either the independent firewall or the built-in
firewall of the MGW can be used. The built-in firewall can filter the packets based on the source
IP address, destination IP address, source UDP port number, destination UDP port number, and
protocol type. Note that when the built-in firewall of the MGW is started, the performance of
the MGW is decreased by 30%.

VoIP User Manual 2 VoIP Planning
2 VoIP Planning
About This Chapter
This section describes the requirement for the QoS of the bearer network in the VoIP networking
mode, planning principle and traffic planning over each interface for VoIP networking evolution.
2.1 System Requirement

The data is transmitted over the IP bearer network. The quality of the IP bearer network is closely
related to the quality of the mobile service. To ensure the normal service operation, the quality
of the IP bearer network must meet the specified requirement.
2.2 Planning Principle
This section describes some principles to ensure the safe and reliable network operation during
VoIP network planning.
2.3 Traffic Planning over Each Interface for VoIP Networking Evolution
During VoIP networking evolution, both the traffic over each interface and number of wideband
signaling links based on the IP bearer mode must be planned. At the same time, when the original
TDM signaling links remain unchanged after network evolution, the number of links for capacity
expansion based on the TDM bearer mode must also be planned.

2 VoIP Planning VoIP User Manual
2.1 System Requirement

The data is transmitted over the IP bearer network. The quality of the IP bearer network is closely
related to the quality of the mobile service. To ensure the normal service operation, the quality
of the IP bearer network must meet the specified requirement.
The QoS of the voice services concerns the delay, jitter and packet loss ratio. The QoS of the IP
bearer network of the voice services is defined as follows:
DATA Unidirectional Packet Loss Ratio Jitter (ms)

Delay (ms)
Media <50 (Recommended) 1% (Consider VP 1

<400 (maximum) services)
Signalling 100 0.1% (Consider 10

high traffic)
Requirement to IP bearer network of speech service

To ensure the QoS of the entire network, the QoS of the partial networks must be under control.
In this way, the normal service operation is assured. According to the QoS indexes of the voice
services, the QoS indexes of the IP bearer network must satisfy the following requirements:
Network Class Unidirectional Packet Loss Ratio Jitter

Delay
Good 5 0.01% 1
Normal 50 0.1% 10
Bad 200 5% 40
For example, the network quality is as follows:

l The delay is less than 80 ms at the access network side.
l The overheading of the caller and callee is 150 ms at the access network side.
l The overheading of the caller and callee MGWs is 50 ms. (The duration for packing is 20
ms.)
According to the permitted unidirectional delay of the voice services defined in the 3GPP, the
average value 250 ms (the average of 50 ms and 400 ms) is used. The delay of the IP bearer
network must be less than 50 ms (250-150-50=50).
The core network components apply the Jitter Buffer technology, so the jitter can be reduced by
20 ms. If the jitter of the IP bearer network is less than 20 ms, the jitter of the voice services can
be reduced to 1 ms by using the Jitter Buffer technology in the core network components.
The core network components provide the packet loss compensation function. The core network
components can eliminate the impact of random packet loss through packet loss compensation

and redundancy transmission function. When the packet loss ratio of the IP bearer network is
less than 5%, the core network components can eliminate the impact of packet loss. Therefore,
the packet loss ratio can be less than 3% as required.
Requirement to IP bearer network of speech codec

The QoS of the IP bearer network is different when there are different speech codec. The operator
can get the bandwidth with different bandwidth coefficient related to the different speech codec.
The bandwidth coefficient related to the different speech codec are as follows.
Speech codec Compressing Period of Bandwidth Bandwidth

speech rate packing (ms) coefficient of coefficient of
(kbps) the IP bearer the Ethernet
network
G.711a/G.711 64 20 1.25 1.41
G.729a/G.729b 8 20 0.38 0.54
G.732.1 6.3 30 0.27 0.37

(6.3kbit/s)
G.732.1 5.3 30 0.25 0.36

(5.3kbit/s)
2.2 Planning Principle

This section describes some principles to ensure the safe and reliable network operation during
VoIP network planning.
During network planning, the traffic model of the VoIP and the calculation method for interface
traffic are the same as those of the TDM networking planning. Comparing with the TDM
network, the VoIP network has the following differences in planning:
l For the VoIP network, the IP addresses of the ports for transmitting signaling of NEs,
service data and maintenance data must be planned globally. You must reserve IP addresses
based on the full configuration of the softswitching equipment and service usage.
l In the VoIP network, the MGW uses the IP-based AMR. The protocol stack is different
from that used in the TDM networking.
l The sufficient bandwidth must be reserved. You must reserve 20% of outgoing bandwidth
for the core network, and 100% of bandwidth for the IP bearer network.
l The bandwidth between nodes is configured according to the shortest path.
2.2.1 Networking Planning

This section describes the private network for bearing circuit switched domain data, service
network and maintenance network isolation, and dual plane networking in the VoIP networking
mode. You need to follow these principles to ensure the network security and reliability.
2.2.2 Bandwidth Planning
You must consider the actual service forwarding ability and service transmission ability of the
digital communication component during the bandwidth planning. In this way, you can enable
the network to provide sufficient bandwidth for normal service operation.

2.2.1 Networking Planning

This section describes the private network for bearing circuit switched domain data, service
network and maintenance network isolation, and dual plane networking in the VoIP networking
mode. You need to follow these principles to ensure the network security and reliability.
Private Network for Bearing Circuit Switched Domain Data

The private network for bearing Circuit Switched domain data can use the logical isolation
method and a private IP bearer network.
l Using logical isolation
When the logical isolation method is used, you can use the current IP network as the bearer
network without creating a private IP network. The bearer network can work with the bearer
network of the Internet. The CS services can be isolated through logical isolation methods
such as Virtual Private Network (VPN).
This method features low cost but poor network isolation effect.
l Creating a private IP bearer network
When using this method, you need to create a new IP backbone bearer network for mobile
services. This bearer network is isolated from the IP network bearing Internet services
physically.
The network isolation effect of this method is good and network security can be assured
easily.
Service Network and Maintenance Network Isolation

The service network is the IP network bearing media streams and signaling streams. The
maintenance network is the IP network bearing network management and charging information.
The two networks must be isolated physically, especially at the IP access layer.
One independent IP bearer network bears the service data and the other independent IP bearer
network (such as the internal private DDN) bears the maintenance data. If the private DDN is
not provided, you can create an independent VPN for central network management in the IP
backbone network. In this way, the network management data and service data can be isolated
logically. The network management data of different locations is exchanged through this VPN.
Using Dual Plane Networking

This networking method can avoid single node failure in the IP network and improve the network
reliability. The IP interfaces of all core network components and digital communication
components must use the dual uplink networking. The core network component is connected to
two digital communication components. The uplink of each digital communication component
is connected to another two digital communication components. Figure 2-1 shows the
networking.

Figure 2-1 Dual plane networking
NM center
MSC Server
LAN Switch
Plane A
Plane B
NM center
MGW MGW
The active/standby configuration mode or load sharing mode can be used between the two planes.
To ensure the QoS (no call drop is found; the connected calls are not affected; a new call can be
set up normally.) after switchover (as the switchover arose by BFD) due to failure, the bandwidth
of each plane must be sufficient to bear the traffic of the entire network. Therefore, the bandwidth
of the entire network must double the actual bandwidth.
2.2.2 Bandwidth Planning

You must consider the actual service forwarding ability and service transmission ability of the
digital communication component during the bandwidth planning. In this way, you can enable
the network to provide sufficient bandwidth for normal service operation.
Planning Based on Actual Bandwidth

The actual bandwidth is the actual service forwarding ability of digital communication
components and links. It is not the standard bandwidth that is provided by the ports. The actual
bandwidth concerns the ability of the digital communication component, IP packet length, port
convergence ratio of interface board, Trunk ability, and QoS mechanism. During bandwidth
planning, you must configure interface boards, ports and links based on the actual bandwidth.
Reserved bandwidth must be configured during bandwidth planning. You can calculate the
reserved bandwidth based on the algorithm for trunk circuits of the PSTN and links of Signaling
System Number 7.
Planning Based on Service Traffic

Figure 2-2 is an example of the bearer network access model. You must consider the subscriber
number, toll call ratio and call model during bandwidth planning.

Figure 2-2 Bearer network access model
CR
CR
CR
CR
CR
BR BR
AR AR AR AR
MGW MGW MGW MGW
l Bandwidth between the MGW and access router (AR)

The outgoing bandwidth between the MGW and AR is determined by: the subscriber
numbers, toll call ratio (inner and outer province calls), local call ratio of remote MGWs,
and call model. The total bandwidth of the ARs is the sum of the bandwidth of connected
MGWs.
l Bandwidth between the AR and bearer router (BR)
The outgoing bandwidth between the AR and BR is determined by: the subscriber number,
toll call ratio (inner and outer province calls) in MGWs, and call model. The total bandwidth
of the BRs is the sum of the bandwidth of connected ARs.
l Bandwidth between the BR and center router (CR)
The outgoing bandwidth between the BR and CR is determined by: toll call ratio (inner
and outer province calls) in MGWs and call model. The total bandwidth of the CRs is the
sum of the bandwidth of connected BRs.
l Bandwidth between CRs
All CRs are interconnected. The outgoing bandwidth between CRs is determined by:
subscriber capacity, call traffic ratio between multi-areas and call model.
2.3 Traffic Planning over Each Interface for VoIP

Networking Evolution
During VoIP networking evolution, both the traffic over each interface and number of wideband
signaling links based on the IP bearer mode must be planned. At the same time, when the original
TDM signaling links remain unchanged after network evolution, the number of links for capacity
expansion based on the TDM bearer mode must also be planned.

Table 2-1 shows the input information required for traffic planning over each interface.
Table 2-1 Input information required for traffic planning over each interface
Required Information Remarks
Capacity of the target network The designer can better understand the scope
of the design.
Busy-hour traffic model of each office on the To calculate the traffic over each interface, it
original network to be optimized is required to use the busy-hour traffic model
of each office on the original network to be
optimized.
Busy-hour traffic statistics of each office on To calculate the traffic over each interface, it
the original network to be optimized is required to use the busy-hour traffic
statistics of each office on the original
network to be optimized.
Rate of traffic over each interface To calculate the traffic over each interface, it
is required to use the rate of traffic over each
interface.
Structure of traffic on the target network The designer can better understand the traffic
between network elements to be calculated
after network optimization.
Structure of signaling on the target network The designer can better understand the
signaling traffic between network elements to
be calculated after network optimization.
Planning the Traffic over Each Interface and Number of Wideband Signaling Links
Based on the IP Bearer Mode
The following parameters are required for planning the traffic over each interface and number
of wideband signaling links based on the IP bearer mode:
l Traffic over each signaling interface and number of wideband links based on the IP bearer
mode
Traffic over each interface and number of signaling links between MSC servers based
on the IP bearer mode
Traffic over each interface and number of signaling links between the MSC server and
the MGW based on the IP bearer mode
Required WBSGs and WIFMs
l Traffic over each interface based on the IP bearer mode
Required BICC CICs
Traffic over the Nb interface based on the IP bearer mode
Required MHRUs

Planning the Links for Capacity Expansion Based on the TDM Bearer Mode
The following parameters are required for planning the links for capacity expansion based on
the TDM bearer mode:
l Number of signaling links for capacity expansion based on the TDM bearer mode
l Number of E1 circuits for capacity expansion based on the TDM bearer mode
2.3.1 Calculating the IP Traffic of BICC Signaling and Number of BICC or M3UA Links over
the Nc Interface
This section describes how to calculate the IP traffic of BICC signaling and number of BICC or
M3UA links over the Nc interface.
2.3.2 Calculating the IP Traffic of H.248 Signaling and Number of H.248 Links over the Mc
Interface
This section describes how to calculate the IP traffic of H.248 signaling and number of H.248
links over the Mc interface.
2.3.3 Calculating the IP Traffic of BSSAP Signaling and Number of M3UA Links over the A
Interface
This section describes how to calculate the IP traffic of BSSAP signaling and number of M3UA
links over the A interface.
2.3.4 Calculating the Traffic over the Nb Interface
This section describes how to calculate the traffic over the Nb interface.
2.3.5 Calculating BICC CICs
This section describes how to calculate BICC CICs.
2.3.6 Calculating the Numbers of Required WBSGs and WIFMs
This section describes how to calculate the numbers of required WBSGs and WIFMs.
2.3.7 Calculating the Number of Required MHRUs
This section describes how to calculate the number of required MHRUs.
2.3.8 Calculating the Number of TDM Signaling Links
This section describes how to calculate the number of TDM signaling links.
2.3.9 Calculating the Number of E1 Circuits Required for the TDM Bearer Mode
This section describes how to calculate the number of E1 circuits required for the TDM bearer
mode.
2.3.1 Calculating the IP Traffic of BICC Signaling and Number of

BICC or M3UA Links over the Nc Interface
This section describes how to calculate the IP traffic of BICC signaling and number of BICC or
M3UA links over the Nc interface.
Input Parameters
To calculate the IP traffic of BICC signaling and number of BICC or M3UA links over the Nc
interface, the following parameters are required:
l Number of incoming Busy Hour Call Attempt (BHCA) and traffic in a certain office
direction
l Number of outgoing BHCA and traffic in a certain office direction

l Number of subscribers on the original network (original network traffic)

l Number of subscribers on the target network (target network traffic)
Calculation Formulas for the Original Network

To calculate the IP traffic of BICC signaling and number of BICC or M3UA links for the original
network, use the following formulas based on the preceding input parameters:
l Inter-MSC BHCA = Incoming BHCA + Outgoing BHCA
l Number of inter-MSC message packets per call in the BICC office direction = Number of
incoming message packets per call in the BICC office direction + Number of outgoing
BICC message packets per call in the BICC office direction
l CAPS = Inter-MSC BHCA 3600
l BICC signaling traffic (Mpbs) = CAPS Maximum number of bytes per call in the
unidirectional BICC office direction 8 1,000,000 IP signaling bandwidth redundancy
factor
l BICC signaling PPS = CAPS Number of inter-MSC message packets per call in the BICC
office direction
l Number of BICC or M3UA signaling links:
N = Roundup (BICC signaling PPS PPS per BICC or M3UA link, 0)
Number of BICC or M3UA signaling links = 2
NOTE
Roundup (value, 0) indicates that the value is rounded up to an integral number.
Calculation Formulas for the Target Network

To calculate the IP traffic of BICC signaling and number of BICC or M3UA links for the target
network, use the following formulas based on the preceding results:
l Growth rate = Number of subscribers on the target network (traffic) Number of
subscribers on the original network (traffic)
l Target network parameter = Original network parameter Growth rate
l Number of BICC or M3UA signaling links = Max (Number of BICC or M3UA signaling
links from MSC A to MSC B, Number of BICC or M3UA signaling links from MSC B to
MSC A)
2.3.2 Calculating the IP Traffic of H.248 Signaling and Number of

H.248 Links over the Mc Interface
This section describes how to calculate the IP traffic of H.248 signaling and number of H.248
links over the Mc interface.
Input Parameters
To calculate the IP traffic of H.248 signaling and number of H.248 links over the Mc interface,
the following input parameters are required:
l BHCA
l Number of handover times


Formulas for Calculating the Parameters Required for the Original Network
To calculate the parameters required for the original network, use the following formulas based
on the preceding input parameters:
l Signaling traffic (Mbps) on the H.248 links = Call signaling traffic on the H.248 links +
Handover signaling traffic on the H.248 links = (BHCA Number of bytes per call on the
H.248 links + Number of handover times in busy hours Number of bytes per handover
on the H.248 links) 8 3,600 1,000,000 IP signaling bandwidth redundancy factor
l Signaling PPS on the H.248 links = Call signaling PPS on the H.248 links + Handover
signaling PPS on the H.248 links = (BHCA Number of message packets per call on the
H.248 links + Number of handover times in busy hours Number of message packets per
handover on the H.248 links) 3,600
l Number of H.248 signaling links = Roundup (Signaling PPS on the H.248 links / PPS per
H.248 link, 0)
NOTE
Formulas for Calculating the Parameters Required for the Target Network
To calculate the IP traffic of H.248 signaling and number of H.248 links on the target network,
you must obtain the number of subscribers or growth rate. To calculate the parameters required
for the target network, use the following formulas based on the preceding parameters of the
original network:
l H.248 signaling traffic on the target network = H.248 signaling traffic on the original
network Growth rate
l Number of H.248 signaling links on the target network = Roundup (H.248 signaling PPS
on the original network Growth rate PPS per H.248 link, 0)
2.3.3 Calculating the IP Traffic of BSSAP Signaling and Number of

M3UA Links over the A Interface
This section describes how to calculate the IP traffic of BSSAP signaling and number of M3UA
links over the A interface.
Input Parameters
To calculate the IP traffic of BSSAP signaling and number of M3UA links over the A interface,
the following input parameters are required:
l Number of location updates
l Number of IMSI detached times
l Number of mobile originated calls
l Number of mobile terminated calls

l Number of handover times on the source MSC side

l Number of handover times on the destination MSC side
l Number of mobile originated short messages
l Number of mobile terminated short messages
l BSSAP signaling traffic (Mbps) = (Number of signaling flows in busy hours Number
of message bytes of the corresponding flow) 8 3,600 1,000,000 IP signaling
bandwidth redundancy factor
l BSSAP signaling PPS = (Number of signaling flows x Number of message packets of
the corresponding flow) 3,600
l Number of BSSAP or M3UA links:
N = Roundup (BICC signaling PPS Number of message packets per M3UA link, 0)
Number of BSSAP or M3UA links = 2
NOTE
l (value) indicates that the value is obtained by adding all parameters.

l Roundup (value, 0) indicates that the value is rounded up to an integral number.
To calculate the parameters required for the target network, use the following formulas based
on the preceding parameters of the original network:

l BSSAP signaling traffic on the target network = BSSAP signaling traffic on the original
network Growth rate
l Number of BSSAP or M3UA links on the target network = BSSAP signaling PPS Growth
rate PPS per M3UA link
2.3.4 Calculating the Traffic over the Nb Interface

This section describes how to calculate the traffic over the Nb interface.
Input Parameters
To calculate the traffic over the Nb interface, the following input parameters are required:
l Incoming traffic in a certain office direction

l Outgoing traffic in a certain office direction

Inter-MSC IP traffic (Mbps) = (Incoming traffic in a certain office direction in busy hours +
Outgoing traffic in a certain office direction in busy hours) Rate of AMR 12.2K (20 MS) based
on IP over Ethernet (Kbps) 1,000 x VAD compression rate IP traffic bandwidth redundancy
factor
NOTE
l ARM 12.2K (20MS) = 30 Kbps

l If the VAD function is enabled on the MGW, the VAD compression rate is equal to 0.6. If the VAD
function is disabled on the MGW, the VAD compression rate is equal to 1.

l IP traffic on the target network = IP traffic on the original network Growth rate
l IP traffic between two MSCs = Max (IP traffic from MSC A to MSC B, IP traffic from
MSC B to MSC A)
NOTE
Max (value 1, value 2) indicates that the greater value is obtained by comparing value 1 with value 2.
2.3.5 Calculating BICC CICs

This section describes how to calculate BICC CICs.
Input Parameters
To calculate BICC CICs, the following input parameters are required:

Formulas for Calculating BICC CICs

To calculate BICC CICs, use the following formulas based on the preceding input parameters:


l BICC CIC = Roundup ((Incoming traffic in a certain office direction in busy hours +
Outgoing traffic in a certain office direction in busy hours) Growth rate 128 Traffic
per line, 0) 128
l BICC CIC between two MSCs = Max (BICC CIC from MSC A to MSC B, BICC CIC from
MSC B to MSC A)
NOTE

l Max (value 1, value 2) indicates that the greater value is obtained by comparing value 1 with value 2.
2.3.6 Calculating the Numbers of Required WBSGs and WIFMs

This section describes how to calculate the numbers of required WBSGs and WIFMs.
Input Parameters
To calculate the numbers of required WBSGs and WIFMs, the following input parameters are
required:
l Number of message packets on the M2UA links of the target network (PPS)
l Number of message packets on the M3UA links of the target network (PPS)
l Number of message packets on the H.248 links of the target network (PPS)
l Number of message packets on the BICC or M3UA links of the target network (PPS)
Formulas for Calculating the Numbers of Required WBSGs and WIFMs

To calculate the numbers of required WBSGs and WIFMs, use the following formulas based on
the preceding input parameters:
l Equivalent number of message packets on the target network (PPS) = Number of message
packets on the M2UA links of the target network (PPS) Conversion coefficient of number
of message packets on the M2UA links + Number of message packets on the M3UA links
of the target network (PPS) Conversion coefficient of number of message packets on the
M3UA links + Number of message packets on the H.248 links of the target network (PPS)
Conversion coefficient of number of message packets on the H.248 links + Number of
message packets on the BICC or M3UA links of the target network (PPS) Conversion
coefficient of number of message packets on the BICC or M3UA links
l Number of required WBSGs = Roundup (Equivalent number of message packets on the
target network (PPS) Equivalent PPS that can be processed by each WBSG, 0) + 1
l Number of required WIFMs = Roundup (Equivalent number of message packets on the
target network (PPS) Equivalent PPS that can be processed by each WIFM, 0)
NOTE
2.3.7 Calculating the Number of Required MHRUs

This section describes how to calculate the number of required MHRUs.
Input Parameters
To calculate the number of required MHRUs, the following input parameters are required:

l Number of incoming call attempts in a certain office direction based on the IP bearer mode
l Number of outgoing call attempts in a certain office direction based on the IP bearer mode
l Number of message packets on the H.248 links of the target network (PPS)
l Number of message packets on the BICC or M3UA links of the target network (PPS)
Formulas for Calculating the Number of Required MHRUs

To calculate the numbers of required MHRUs, use the following formulas based on the preceding
input parameters:

l CAPS based on the IP bearer mode = (Number of incoming BHCA in a certain office
direction based on the IP bearer mode + Number of outgoing BHCA in a certain office
direction based on the IP bearer mode) Growth rate 3,600
l Number of required MHRUs = Roundup (CAPS based on the IP bearer mode CAPS that
can be processed by each pair of the MHRUs, 0)
NOTE
2.3.8 Calculating the Number of TDM Signaling Links

This section describes how to calculate the number of TDM signaling links.
Input Parameters
To calculate the number of TDM signaling links, the following input parameters are required:
l Send seizure ratio

l Receive seizure ratio
l Total send load = (Send seizure ratio per link in a certain office direction)
l Total receive load = (Receive seizure ratio per link in a certain office direction)
l Total bidirectional load = Total send load + Total receive load
NOTE
(value) indicates that the value is obtained by adding all parameters.


l Number of TDM links on the target network = Total bidirectional load of the original
network Growth rate Maximum bidirectional load per link
NOTE
If the number of required TDM links (64 Kbit/s) exceeds 16, the TDM link (2 Mbit/s) is recommended.
2.3.9 Calculating the Number of E1 Circuits Required for the TDM

Bearer Mode
This section describes how to calculate the number of E1 circuits required for the TDM bearer
mode.
Input Parameters
To calculate the number of E1 circuits required for the TDM bearer mode, the following input
parameters are required:

Formula for Calculating the Parameters Required for the Original Network
Number of E1 circuits in a certain office direction on the original network = Roundup ((Incoming
traffic in a certain office direction + Outgoing traffic in a certain office direction) TDM traffic
redundancy factor 31, 0)
Formula for Calculating the Parameters Required for the Target Network

l Number of E1 circuits in a certain office direction for the target network = Roundup
((Incoming traffic in a certain office direction + Outgoing traffic in a certain office
direction) Growth rate TDM traffic redundancy factor 31, 0)
l Number of inter-MSC E1 links = Max (Number of E1 circuits from MSC A to MSC B,
Number of E1 circuits from MSC B to MSC A)
NOTE
l Number of E1 circuits is calculated on the basis of 31 timeslots in each E1 circuit.

l Max (value 1, value 2) indicates that the greater value is obtained by comparing value 1 with value 2.

VoIP User Manual 3 Newly Established VoIP Networking
3 Newly Established VoIP Networking
About This Chapter
This section describes the methods of the data configuration on the newly established VoIP
networking.
Figure 3-1 shows the ordinary networking structure of the VoIP networking.
Figure 3-1 VoIP networking structure
HLR SCP
MSC Server
MSC Server
MGW MGW
TDM
IP
BSC BSC

3 Newly Established VoIP Networking VoIP User Manual
From the preceding figure, you can know that:

l The messages between MSC servers, between the MSC server and MGW, and between
MGWs are transferred over the IP bearer.
The interworking between the MSC server and the MGW is based on the H.248 protocol.
The structure of the protocol stack on the MSC server side is H.248/SCTP/IP, which is
the same as that on the MGW side.
The interworking between MSC servers is based on the BICC protocol. The structures
of the protocol stacks on the MSC server side are the same, which are BICC/M3UA/
SCTP/IP.
l The MSC server can interwork with core network NEs, such as the HLR and the SCP, in
direct MTP connection mode, non-direct M2UA connection mode, or non-direct M3UA
connection mode.
When the MSC server interworks with the peer NE in direct MTP connection mode,
the structures of the bottom layer bearer protocols on the MSC server side and the peer
NE side are the same, which are MTP3/MTP2/MTP1.
When the MSC server interworks with the peer NE in non-direct M2UA connection
mode, the structure of the bottom layer bearer protocol on the MSC server side is MTP3/
M2UA/SCTP/IP, and the structure of the bottom layer bearer protocol on the peer NE
side is MTP3/MTP2/MTP1.
When the MSC server interworks with the peer NE in non-direct M3UA connection
mode, the structure of the bottom layer bearer protocol on the MSC server side is M3UA/
SCTP/IP, and the structure of the bottom layer bearer protocol on the peer NE side is
MTP3/MTP2/MTP1.
l The MSC server interworks with the BSC in non-direct M3UA connection mode. The
structure of the protocol stack on the MSC server side is BSSAP/SCCP/M3UA/SCTP/IP,
and the structure of the protocol stack on the BSC side is BSSAP/SCCP/MTP3/MTP2/
MTP1.
The difference between the newly established VoIP networking and traditional networking lies
in different IP-related data configurations, as shown in Figure 3-2.

Figure 3-2 Procedure of the VoIP data configuration
Start
Load a license that supports the

related speech codec
Set the codec type in the local

information
Set the IP address of the router
Add MGW control data between

the MSC server and the MGW
Add IP bearer data between MGWs
Configure M3UA signaling data for

interworking with peer NE
Configure BICC signaling data for

interworking with peer NE
End
The IP bearer data configuration is closely related to the interworking data of interfaces between
NEs, but merely related to the entire network structure. Therefore, this manual describes only
the data configuration on the typical networking by taking the single-MGW networking, multi-
MGW networking, and CMN networking as examples.
3.1 Single-MGW Networking

This section describes the VoIP-related data that must be configured when the VoIP service is
used in the single-MGW networking by taking a typical single-MGW networking mode as an
example.
3.2 Multi-MGW Networking

used in the multi-MGW networking by taking a typical multi-MGW networking mode as an
example.
3.3 CMN Networking
used in the CMN networking by taking a typical CMN networking mode as an example.


used in the single-MGW networking by taking a typical single-MGW networking mode as an
example.
Networking Description
The single-MGW networking is the simplest networking in the core network and easy to be
implemented. It, however, can provide limited channel resources. Figure 3-3 shows the typical
single-MGW VoIP networking mode.
Figure 3-3 Single-MGW VoIP networking
MSC Server
HLR
STP
CE1 CE2
MSC1 MGW MSC2
TDM
BSC IP
l When the MSC server interworks with the NE of the core network, the SPC for the national
network is used. When the MSC server interworks with the NE of the access network, the
SPC for the national reserved network is used.
l The interworking between the MSC server and the MGW is based on the H.248 protocol
and IP bearer. The structures of the protocol stack on the MSC server side and the MGW
side are the same, which are H.248/SCTP/IP.
l The interworking between the MSC server and MSC1 is based on the BICC protocol. The
MSC server interworks with MSC1 in M3UA direct connection mode. The structure of the
protocol stack on the MSC server side is BICC/M3UA/SCTP/IP, which is the same as that
on the MSC1 side.

l The interworking between the MSC server and MSC2 is based on the ISUP protocol. The
MSC server interworks with MSC2 in non-direct M3UA connection mode. The MGW (SG)
is used as the signaling transfer point (STP). The structure of the protocol stack on the MSC
server side is ISUP/SCCP/M3UA/SCTP/IP. The structure of the protocol stack on the
MSC2 side is ISUP/SCCP/MTP3/MTP2/MTP1.
l The MSC server is directly connected with the STP and the HLR through the TDM bearer.
Interworking Data
For the single-MGW networking, you must plan and collect the following data:
l Signaling point name
l Signaling point code
l Office direction name
l IP address
l SCTP port number
3.1.1 Configuration Flow

This section describes the procedure of the data configuration on the single-MGW VoIP
networking.
3.1.2 Interworking Data
This section describes the interworking data to be negotiated in the single-MGW VoIP
networking.
3.1.3 Data Configuration on the MSOFTX3000
This section describes the VoIP-related data configuration on the MSOFTX3000 in the single-
MGW VoIP networking.
3.1.4 Data Configuration on the UMG8900
This section describes the VoIP-related data configuration on the UMG8900 in the single-MGW
VoIP networking.

This section describes the procedure of the data configuration on the single-MGW VoIP
networking.
Figure 3-4 shows the procedure of the data configuration on the single-MGW VoIP networking.

Figure 3-4 Procedure of the data configuration on the single-MGW VoIP networking
Start
Load a license file
Set local office information
Configure data for

interworking between the MSC
server and the MGW
Configure data for

interworking with MSC1
Configure data for

Configure data for

interworking with the BSC
End

This section describes the interworking data to be negotiated in the single-MGW VoIP
networking.
For the networking shown in Figure 3-3, the interworking data to be negotiated is as follows.
Table 3-1 Office information data (single-MGW)
Office Information Value
Local MSC number 8613900001
Local VLR number 8613900001

Mobile country code 460
Mobile network code 00
Country code 86
Local area code 755
Call source name CALL_SRC1
Table 3-2 Interworking data between the MSC server and other NEs (single-MGW)
Nam MSC Server MGW BSC MSC1 MSC2

e
Signal MSX MGW1 BSC1 MSX01 MSX02

ing
point
name
Signal National National - National National

ing network: A0001 network: network: network:
point (in 24-bit code A1001 C0001 C0002
code format)
National National National - -

reserved reserved reserved
network: 001 (in network: 101 network:
14-bit code A01
format)
SCTP H.248 link: 5000 H.248 link: - Module 141: -

port Module 141: 5000 and 5001 7000
numb 6000, 6010, and SPF board 0: Module 142:
er 7000 6000 and 6010 7001
Module 142: SPF board 1:
6001, 6011, and 6001 and 6011
7001
IP Signaling plane: Signaling - Signaling -

addre 192.168.0.10/25 plane: plane:
ss 5.255.0.0 192.168.10.10/ 192.168.20.1
255.255.0.0 0/255.255.0.
Bearer plane: 0
10.168.10.10/2 Bearer plane:
55.255.0.0 10.168.10.10
/255.255.0.0
Speec The TrFO function, the 2198 redundancy function of level 1 (PT value is 99), the
h UMTS AMR2 12.2K codec, and the G.711a codec are supported. The IP bearer is
Codec adopted.


This section describes the VoIP-related data configuration on the MSOFTX3000 in the single-
Procedure
Step 1 Load a license file.
Descr Load a license file where License file name is LICENSE.dat.

iption
Script LOD LICENSE: FN="LICENSE.dat";
Rema For BICC inter-office calls, the license file must support the TrFO function and the
rks voice coding/decoding function. After a license file is loaded, you must run DSP
LICENSE to display the information on the license file, and then confirm whether
the information is correct. To meet the requirements of the VoIP networking, you
must confirm that the license file supports the TrFo and AMR2 functions.
Step 2 Configure local office information.

1. Set a local office information data record.
Desc Set a local office information data record.

riptio
n
Scrip SET OFI: OFN="MSX", LOT=LOCMSC, NN=YES, NN2=YES, SN1=NAT, SN2=NATB,

t NPC="A0001", NP2C="001", NNS=SP24, NN2S=SP14, SPF=YES, LAC="755",
LNC=K'86, SPFAXDT=YES, SPRED=YES, REDLEVEL=L1, REDPT=99,
REDCODEC=G711a;
Rem The local office is configured to support G.711a redundancy of level 1 and the
arks value of the redundancy payload type is set to 99.
2. Set a mobile local office information data record.
Desc Set a mobile local office information data record.

riptio
n
Scrip SET INOFFMSC: MSCN=K'8613900001, VLRN=K'8613900001, MCC=K'460,

t MNC=K'00, INNATIONPFX=K'00, NATIONPFX=K'0;
Rem This command defines the local MSC number and local VLR number in the
arks PLMN when the local MSOFTX3000 works as the VMSC or the GMSC. In
this example, the VLR is embedded in the MSOFTX3000 (MSC). Therefore,
the MSC number and the VLR number can be set to the same number.
3. Add a VLR configuration data record.

Desc Add a VLR configuration data record by setting the MSRN/HON number
ripti allocation mode.
on
Scrip ADD VLRCFG: MAXUSR=12000, MCC=K'460, CC=K'86, MSRNAM=LAI, HONAM=RAND;

t
4. Add a call source name data record.
Desc Add a call source name data record where Call source name is
riptio CALL_SRC1, Route selection source name is RSSN1, and Failure source
n name is FSN1.
Scrip ADD CALLSRC: CSCNAME="CALL_SRC1", RSSN="RSSN1", FSN="FSN1";

t
5. Add an FE port configuration data record.
Desc Add an FE port configuration data record. The default IP address of the gateway
riptio is the IP address (192.168.0.1) of the router.
n
Scrip ADD FECFG: MN=132, IP="192.168.0.10", MSK="255.255.0.0",

t DGW="192.168.0.1";
Rem The MSOFTX3000 is connected to the default router (gateway) through the FE
arks port, and then communicates with other IP devices through the default router
(gateway). You must correctly set the IP address of the default router (gateway)
connected to the FE port; otherwise, the MSOFTX3000 cannot properly
communicate with IP devices.
Step 3 Configure MGW data.

1. Add an MGW data record.
Descr Add an MGW data record where MGW name is MGW1, Transport
iptio protocol is SCTP protocol, and Server/Client is Server.
n
Scrip ADD MGW: MGWNAME="MGW1", TRNST=SCTP, CTRLMN=136, MGWDESC="MGW1",

t BCUID=1, BNCC=IP-1, DEVNAME="MGW1", SVRCLIENT=SVR, ENCT=NSUP,
CPB=TONE-1&PA-1&SENDDTMF-1&DETECTDTMF-1&MPTY-1&IWF-1, ECRATE=300,
IWFRATE=300, TONERATE=300, MPTYRATE=300, DETDTMFRATE=300,
SNDDTMFRATE=300,
UMTSAMR2R=RATE475-0&RATE515-0&RATE590-0&RATE670-0&RATE740-0&RATE795
-0&RATE102-0&RATE122-1, TC=UMTSAMR2-1&PCMA-1;
Rema l When the MSOFTX3000 interworks with an MGW through an IP interface,

rks the SCTP protocol is used for transport.
l The working mode of the MSOFTX3000 in associated SCTP mode must be
consistent with that of the MGW. When Server/Client in the command is
set to Server on the MSOFTX3000 side, Server/Client on the MGW side
must be set to Client.

2. Add two H.248 link data records.
Desc Add two H.248 links. For one link, set MGW name to MGW1, Transport
riptio protocol to SCTP protocol, Link name to MGW1H-LNK1, Local IP
n address to 192.168.0.10, Local port number to 5000, Remote IP address 1
to 192.168.10.10, and Remote port number to 5000. For the other link, set
MGW name to MGW1, Transport protocol to SCTP protocol, Link
name to MGW1H-LNK2, Local IP address to 192.168.0.10, Local port
number to 5000, Remote IP address 1 to 192.168.10.10, and Remote port
number to 5001.
Scrip ADD H248LNK: MGWNAME="MGW1 ", TRNST=SCTP, LNKNAME=" MGW1H-LNK1",

t MN=141, SLOCIP1="192.168.0.10", SLOCPORT=5000,
SRMTIP1="192.168.10.10", SRMTPORT=5000;
ADD H248LNK: MGWNAME=" MGW1 ", TRNST=SCTP, LNKNAME=" MGW1H-LNK2",
MN=141, SLOCIP1="192.168.0.10", SLOCPORT=5000,
SRMTIP1="192.168.10.10", SRMTPORT=5001;
Rem l When H.248 messages are transmitted between the MSOFTX3000 and an
arks MGW through the SCTP protocol, you can configure up to 30 H.248 links
between the MSOFTX3000 and the MGW.
l To ensure reliability and availability, you must configure multiple H.248
links between the MSOFTX3000 and the MGW.
l The MSOFTX3000 must act as the server. If the MSOFTX3000 acts as the
client, the virtual MGW cannot register and cannot be activated.
l If multiple H.248 links are configured between the MSOFTX3000 and an
MGW, it is recommended that those H.248 links adopt the same local port
number.
Step 4 Configure M3UA data for interworking with the MGW.
The M3UA data configured by using the following operations is used for the national network.
1. Add an M3UA local entity data record.
Desc Add an M3UA local entity data record where Local entity name is MSX-A,
riptio Signaling point code of local entity is A0001, and Local entity type is
n Application server.
Scrip ADD M3LE: LENM="MSX-A", NI=NAT, OPC="A0001", LET=AS;

t
Rem The value of Route context can be a decimal number or null (not configured).
arks Ensure that the value of Route context on the MSOFTX3000 is consistent with
the value of Route context on the MGW. Here, it is null.
2. Add an M3UA destination entity data record.
Desc Add an M3UA destination entity data record where Destination entity name
riptio is MGW1-A, indicating the UMG8900 (SG) directly connected to the
n MSOFTX3000.

Scrip ADD M3DE: DENM="MGW1-A", LENM=" MSX-A", NI=NAT, DPC="A1001",

t STPF=TRUE, DET=SG;
Rem l The connection between the MSOFTX3000 and the UMG8900 adopts the
arks non-peer-to-peer network mode, and the entity type of the MSOFTX3000 is
application sever. Therefore, the entity type of the UMG8900 must be the
signaling gateway.
l The ISUP signaling is exchanged between the MSOFTX3000 and the PSTN
switch through the UMG8900. Therefore, you must set Destination entity
type of the PSTN switch to Signaling point.
l The UMG8900 in the signaling network provides the signaling transfer
function. Therefore, you must set STP flag to Yes; otherwise, an error occurs
when you add an M3UA route data record by running ADD M3RT.
3. Add an M3UA link set data record.
Desc Add an M3UA link set data record where Linkset name is MGW1-A, and
riptio Adjacent entity name is MGW1-A.
n
Scrip ADD M3LKS: LSNM="MGW1-A", ADNM="MGW1-A", TM=LOADSHARE, WM=ASP;

t
Rem l There are straight-through M3UA signaling links between the

arks MSOFTX3000 and the UMG8900. Therefore, Adjacent entity name must
be set to the entity name of the UMG8900. Here, Adjacent entity name is
set to MGW1-A.
l The connection between the MSOFTX3000 and the UMG8900 adopts the
non-peer-to-peer network mode, and the MSOFTX3000 acts as an
application server. Therefore, Work mode must be set to ASP.
The traffic mode of the link set must be consistent with the traffic mode of the
link set on the SG side; otherwise, none of the links of the link set can work
properly. Usually, set Traffic mode to Loadshare.
4. Add an M3UA route data record.
Desc Add an M3UA route data record. The name of the destination entity of the
riptio UMG8900 (SG) route is MGW1-A.
n
Scrip ADD M3RT: DENM=" MGW1-A ", LSNM=" MGW1-A";

t
5. Add an M3UA link data record.
Desc Add two M3UA link data records. For one link, set Client/Server to Client,
riptio WBSG module number to 141, Link name to MGW1-A1, Local port
n number to 6000, and Peer port number to 6000; for the other link, set Client/
Server to Client, WBSG module number to 142, Link name to MGW1-
A2, Local port number to 6001, and Peer port number to 6001.

Scrip ADD M3LNK: MN=141, LNKNM="MGW1-A1", LOCIP1="192.168.0.10",

t LOCPORT=6000, PEERIP1="192.168.10.10", PEERPORT=6000, CS=C, LSNM="
MGW1-A", ASF=YES;
ADD M3LNK: MN=142, LNKNM="MGW1-A2", LOCIP1="192.168.0.10",
LOCPORT=6001, PEERIP1="192.168.10.10", PEERPORT=6001, CS=C, LSNM="
MGW1-A", ASF=YES;
Rem To ensure the reliability of M3UA links, those M3UA links to the same
arks destination signaling point must be assigned to different WBSGs.
The M3UA data configured by the following operations are used for the national reserved
network.
Desc Add an M3UA local entity data record where Local entity name is MSX-B,
riptio Signaling point code of local entity is 001, and Local entity type is
Scrip ADD M3LE: LENM="MSX-B", NI=NATB, OPC="001", LET=AS;

t

riptio is MGW1-B, indicating the UMG8900 (SG) directly connected to the
n MSOFTX3000.
Scrip ADD M3DE: DENM="MGW1-B", LENM="MSX-B", NI=NATB, DPC="101",

t STPF=TRUE, DET=SG, NM=INCLUSIVE;
signaling gateway.
switch through the UMG8900. Therefore, you must set the Destination
entity type of the PSTN switch to Signaling point.

Desc Add an M3UA link set data record where Linkset name is MGW1-B, and
riptio Adjacent entity name is MGW1-B.
n

Scrip ADD M3LKS: LSNM="MGW1-B", ADNM="MGW1-B", TM=LOADSHARE, WM=ASP;

t

set to MGW1-B.

riptio UMG8900 (SG) route is MGW1-B.
n
Scrip ADD M3RT: DENM=" MGW1-B ", LSNM=" MGW1-B";

t

riptio WBSG module number to 141, Link name to MGW1-B1, Local port
B2, Local port number to 6011, and Peer port number to 6011.
Scrip ADD M3LNK: MN=141, LNKNM="MGW1-B1", LOCIP1="192.168.0.10",

MGW1-B", ASF=YES;
ADD M3LNK: MN=142, LNKNM="MGW1-B2", LOCIP1="192.168.0.10",
MGW1-B", ASF=YES;
Step 5 Configure data for interworking with MSC1.

riptio is MSX01, indicating the IP MSC directly connected to the MSOFTX3000.
n
Scrip ADD M3DE: DENM="MSX01", LENM="MSX01-A", NI=NAT, DPC="C0001", STPF=

t FALSE, DET=AS;

Rem l The connection between the MSOFTX3000 and the IP MSC adopts the peer-
arks to-peer network mode, and the entity type of the MSOFTX3000 is
application sever. Therefore, the entity type of the IP MSC must be the
application sever.
l Usually, the IP MSC in the signaling network is not required to provide the
signaling transfer function. Therefore, you must set STP flag to No.
Desc Add an M3UA link set data record where Linkset name is MSX01, and
riptio Adjacent entity name is MSX01.
n
Scrip ADD M3LKS: LSNM="MSX01", ADNM="MSX01", TM=LOADSHARE, WM=IPSP;

t
Rem The connection between the MSOFTX3000 and the UMG8900 adopts the peer-
arks to-peer network mode, and the MSOFTX3000 acts as an application server.
Therefore, Work mode must be set to IPSP.
The traffic mode of the link set must be consistent with that of the peer network
element; otherwise, all M3UA links of the link set cannot work properly.
Usually, set Traffic mode to Loadshare.
Desc Add an M3UA route data record. The name of the destination entity of the IP
riptio MSC route is MSX01.
n
Scrip ADD M3RT: DENM="MSX01", LSNM=" MSX01";

t
4. Add two M3UA link data records.
riptio WBSG module number to 141, Link name to MSX01-1, Local port
Server to Client, WBSG module number to 142, Link name to MSX01-2,
Local port number to 7001, and Peer port number to 7001.
Scrip ADD M3LNK: MN=141, LNKNM="MSX01-1", LOCIP1="192.168.0.10",

t LOCPORT=7000, PEERIP1="192.168.20.10", PEERPORT=7000, CS=C,
LSNM="MSX01", ASF=YES;
ADD M3LNK: MN=142, LNKNM="MSX01-2", LOCIP1="192.168.0.10",
LOCPORT=7001, PEERIP1="192.168.20.10", PEERPORT=7001, CS=C,
5. Add an office direction data record.

Desc Add an office direction data record where Office direction name is MSX01,
riptio and Signaling type is M3UA.
n
Scrip ADD OFC: ON="MSX01", OOFFICT=MSC, DOL=SAME, BOFCNO=100, OFCTYPE=COM,

t SIG=M3UA, DPC1="C0001", OPC="A0001", MST=NO, CICSEL=CTRL,
CSCNAME="CALL_SRC1", SUPTTRFO=YES, IFSUPPORTRED=YES, AN="0";
Rem The local office adopts the master/slave circuit selection mode in the routing
arks process and preferentially controls the circuits whose CICs are odd numbers.
The office direction from the local office to the peer office supports the TrFo
function, and the peer office supports the redundancy function.
6. Add a subroute data record.
Desc Add a subroute data record where Sub-route name is MSX01.

riptio
n
Scrip ADD SRT: SRN="MSX01", ON="MSX01";

t
7. Add a route data record.
Desc Add a route data record where Route name is MSX01.

riptio
n
Scrip ADD RT: RN="MSX01", SR1N="MSX01";

t
8. Add a route analysis data record.
Desc Add a route analysis data record where Route selection name is RSSN1.
riptio
n
Scrip ADD RTANA: RSN="MSX01", RSSN="RSSN1", RN="MSX01", ISUP=BICCIP_F;

t
Rem Usually, set Caller category to All, and Address information indicator and
arks Transmission capability of the command to All categories.
9. Add a BICC trunk group data record.
Desc Add a BICC trunk group data record where Trunk group name is MSX01-
riptio MGW1, and MGW name is MGW1.
n
Scrip ADD BICCTG: TGN="MSX01-MGW1", MGWNAME="MGW1", SRN="MSX01", BTG=1,

t CT=BICC_IP, RELRED=NO;

10. Add a BICC CIC module data record.

Desc Add a BICC CIC module where Office direction name is MSX01.
riptio
n
Scrip ADD BICCCICMDU: ON="MSX01", MN=23, SCIC=0, ECIC=63, CCT=NOCHANGE;

t
Rem The number of BICC CICs is determined by the number of the inter-office calls,
arks the type of the voice codec, and the IP bandwidth of the bearer plane. The BICC
CICs do not directly correspond to the IP bearer interface boards of the MGW.
The CICs of the TDM circuits map the TDM timeslots (E1/T1) corresponding
to the MGW.

riptio is MSX02, indicating the MSC2.
n

t FALSE, DET=SP;
Rem The ISUP signaling is exchanged between the MSOFTX3000 and the TDM
arks MSC through the UMG8900. Therefore, you must set Destination entity
type of the TDM MSC to Signaling point.

riptio MSX02 route is MSC2.
n
Scrip ADD M3RT: DENM="MSX02", LSNM="MGW1-A";

t
3. Add an SCCP destination signaling point.

Desc Add an SCCP destination signaling point where DSP name is MSX02 and
riptio DPC is C0002.
n
Scrip ADD SCCPDSP: DPNM="MSX02", NI=NAT, DPC="C0002", OPC="A0001";

t
4. Add SCCP sub-system numbers.

Desc Add SCCP sub-system numbers where SCCP sub-system numbers are
riptio SCMGMSCVLR.
n
Scrip ADD SCCPSSN: SSNNM="MSX02-SCMG", NI=NAT, SSN=SCMG, SPC="C0002",

t OPC="A0001";
ADD SCCPSSN: SSNNM="MSX02-MSC", NI=NAT, SSN=MSC, SPC="C0002",
OPC="A0001";
ADD SCCPSSN: SSNNM="VLR", NI=NAT, SSN=VLR, SPC="C0002", OPC="A0001";
riptio and DPC1 is C0002.
n
Scrip ADD OFC: ON="MSX02", OOFFICT=NATT, DOL=SAME, DOA=PSTN, BOFCNO=11,

t OFCTYPE=COM, SIG=NONBICC/NONSIP, NI=NAT, DPC1="C0002";
Rem This office direction has SS7 trunk circuits. Therefore, you must specify
arks DPC in the command. Otherwise, errors occur when you add an SS7 trunk
group data record by running ADD N7TG.

riptio
n

t

riptio
n

t
riptio
n
Scrip ADD RTANA: RSN="MSX02", RSSN="RSSN1", RN="MSX02", ISUP= ISUP_F;

t

9. Add an SS7 trunk group data record.

Desc Add an SS7 trunk group data record where Trunk group name is MSX02, and
riptio MGW name is MGW1.
n
Scrip ADD N7TG: TGN="MSX02", MGWNAME="MGW1", CT=ISUP, SRN="MSX02",

t G=INOUT, BTG=20, SOPC=" A0001", SDPC="C0002", CSCNAME="CALL_SRC1",
ICR=LCO-1&LC-1&LCT-1&NTT-1&ITT-1, CC=NO, RELRED=NO;
Step 7 Configure data for interworking with the BSC.

riptio is BSC1, indicating the BSC.
n
Scrip ADD M3DE: DENM="BSC1", LENM="MSX-B", NI=NATB, DPC="A01", STPF=

t FALSE, DET=SP;
Rem The signaling is exchanged between the MSOFTX3000 and the BSC through
arks the UMG8900. Therefore, you must set Destination entity type of the BSC to
Signaling point.

Desc Add an M3UA route data record. The name of the destination entity of BSC1
riptio route is BSC1.
n
Scrip ADD M3RT: DENM="BSC1", LSNM="MGW1-B";

t
Desc Add an SCCP destination signaling point where DSP name is BSC1DPC is
riptio A01.
n
Scrip ADD SCCPDSP: DPNM="BSC1", NI=NATB, DPC="A01", OPC="001";

t

riptio SCMGBSSAP.
n

Scrip ADD SCCPSSN: SSNNM="BSC1-SCMG", NI=NATB, SSN=SCMG, SPC="A01",

t OPC="001";
ADD SCCPSSN: SSNNM="BSC1-BSSAP", NI=NATB, SSN=BSSAP, SPC="A01",
OPC="001";
5. Add an office direction.

Desc Add an office direction where Office direction name is BSC1.
riptio
n
Scrip ADD OFC: ON="BSC1", OOFFICT=BSC, DOL=LOW, DOA=BSC, BOFCNO=1,

t OFCTYPE=COM, SIG=NONBICC/NONSIP, NI=NATB, DPC1="A01", SVQE=NO;
6. Add a BSC.
Desc Add a BSC where Route selection source name is RSSN1 and LAI
riptio number is 460000003.
n
Scrip ADD BSC: DPC="A01", OPC="001", BSCNM="BSC1", RSSCNAME="RSSN1",

t MLAIF=NO, LAI="460000003", BSCBEARERTYPE=TDM;
7. Add a radio access network media gateway.

Desc Add a radio access network media gateway where Office direction name is
riptio BSC1 and Media gateway name is MGW1.
n
Scrip ADD RANMGW: OFFICENAME="BSC1", MGWNAME="MGW1";

t
8. Add 2G LAI or GCI informations.

Desc Add 2G LAI or GCI informations where Global cell ID are 460000003 and
riptio 4600000030003, Call source name is CALL_SRC1.
n
Scrip ADD LAIGCI: GCI="460000003", LAIGCINAME="BSC1-LAI",

t MSCN="8613900001", VLRN="8613900001", NONBCLAI=NO, LAICAT=LAI,
LAIT=HVLR, LOCNONAME="INVALID", BSCDPC1="A01", CSNAME="CALL_SRC1",
OFFICENAME="BSC1", TONENAME="INVALID", CELLGROUPNAME="INVALID",
TZDSTNAME="INVALID", LOCATIONIDNAME="INVALID";
ADD LAIGCI: GCI="4600000030003", LAIGCINAME="BSC1-GCI",
MSCN="8613900001", VLRN="8613900001", NONBCLAI=NO, LAICAT=GCI,
9. Add an A-interface circuit pool.

Desc Add an A-interface circuit pool where Circuit pool name is BSC1 .
riptio
n
Scrip ADD ACPOOL: CPNAME="BSC1",

t BCR=SFR1-1&SFR2-1&SFR3-1&SHR1-1&SHR2-1&SHR3-1&DFR1-1&DHR1-1;
10. Add an A-interface trunk group.
Desc Add an A-interface trunk group where Trunk group name is BSC1 and MGW
riptio name is MGW1.
n
Scrip ADD AIETG: TGNAME="BSC1", MGWNAME="MGW1", SOPC="001", SDPC="A01",

t CPNAME="BSC1", BTG=2;
11. Add A-interface trunk circuits.
Desc Add A-interface trunk circuits where Start CIC is 640 and End CIC is 703.
riptio
n
Scrip ADD AIETKC: MN=23, TGNAME="BSC1", SCIC=640, ECIC=703, TID=640;

t
----End

This section describes the VoIP-related data configuration on the UMG8900 in the single-MGW
VoIP networking.
Procedure
1. Set local office information.
Desc Set a local office information data record. The signaling point index of the
riptio national network is 0, and the signaling point index of the national reserved
n network is 1.
Scrip SET OFI: NAME="MGW1", NATRESVLD=YES, SERACH0=NATB, NATRESOPC=H'101,

t NATRESLEN=LABEL14, SPFLAG=NO, STPFLAG=YES;
ADD OFI: INDEX=1, NATRESOPC=H'A1001;
Rem l SET OFI is used to set the local information data record where the signaling
arks point index of the national network is 0.
l ADD OFI is used to add multiple signaling point data records.
Usually, the signaling point function is disabled, and the signaling transfer
function is enabled.

2. Add an IP address data record.
Desc Add an IP address data record where Interface IP address is 192.168.10.10.

riptio
n
Scrip ADD IPADDR: BT=MPU, BN=1, IFT=ETH, IFN=0, IPADDR="192.168.10.10",

t MASK="255.255.0.0", FLAG=MASTER, INVLAN=NO;
3. Add a static route data record.
Desc Add a static route data record.

riptio
n
Scrip ADD ROUTE: BT=MPU, BN=1, DSTIP="0.0.0.0", DSTMASK="0.0.0.0",

t RTTYPE=NEXTHOP, NEXTHOP="192.168.0.1";
Rem l If the MPU board is configured with the centralized transfer function, the
arks default route must be configured.
l The next hop address is the IP address (192.168.0.1) of the router that is
directly connected to the UMG8900.
Step 2 Configure data for IP bearer network.


riptio
n
Scrip ADD IPADDR: BT=HRB, BN=0, IFT=GE, IFN=0, IPADDR="10.168.10.10",

t MASK="255.255.0.0", FLAG=MASTER, INVLAN=NO, IFMPLS=NO;
2. Add a gateway address data record.
Desc Add a gateway address data record where Local IP is 10.168.10.10, and
riptio Gateway IP is 10.168.10.1.
n
Scrip ADD GWADDR: BT=HRB, BN=0, IPADDR="10.168.10.10", GWIP="10.168.10.1",

t TIMEOUT=Aging;
Step 3 Configure data for interworking with the MSOFTX3000.

1. Set a virtual MGW data record.
Desc Set a virtual MGW data record where Virtual media gateway id is 0, and
riptio Virtual media gateway MID type is IP.
n

Scrip SET VMGW: VMGWID=0, MIDTYPE=IP, MID="192.168.10.10:5000";

t
2. Add an MGW controller data record.
Desc Add an MGW controller data record where Virtual media gateway id is 0,
riptio Media gateway controller No. is 0, and Media gateway controller MID
n type is IP.
Scrip ADD MGC: VMGWID=0, MGCIDX=0, MIDTYPE=IP, MID="192.268.0.10:5000",

t MSS=MASTER;
3. Add two H.248 signaling link data records.
Desc Add two H.248 signaling link data records. For one H.248 link, set Transfer
riptio protocol type to SCTP, Local Master IP address to 192.168.100.1, Local
n port No. to 5000, Peer Master address to 192.168.0.1, and Peer port No. to
5000. For the other H.248 link, set Transfer protocol type to SCTP, Local
Master IP address to 192.168.100.1, Local port No. to 5001, Peer Master
address to 192.168.0.1, and Peer port No. to 5000.
Scrip ADD H248LNK: LINKID=48, VMGWID=0, MGCIDX=0, TT=SCTP,

t LOCALIP="192.168.10.10", LOCALPORT=5000, PEERIP="192.168.0.10",
PEERPORT=5000, LINKNAME="MGW1-LNK1", FN=1, SN=4, BP=FRONT;
ADD H248LNK: LINKID=49, VMGWID=0, MGCIDX=0, TT=SCTP,
LOCALIP="192.168.10.10", LOCALPORT=5001, PEERIP="192.168.0.10",
Rem l The numbers of the H.248 links of the SCTP type configured on the
arks UMG8900 must be greater than 47.
4. Activate a virtual MGW.
Desc Activate a virtual MGW.

riptio
n
Scrip ACT VMGW: VMGWID=0;

t
Step 4 Configure M3UA data for interworking with the MSOFTX3000.
Configure M3UA data that is used for the national network.
Desc Add an M3UA local entity data record where Local Entity Index is 0, Local
riptio Entity Name is MGW1-A, Source signaling Point Code is A1001, and Local
n Entity Type is signaling_Gateway.

Scrip ADD M3LE: LEX=0, LEN="MGW1-A", LET=SG, NI=NAT, OPC=H'A1001;

t
Rem The local entity type of the MSOFTX3000 is Application_Server. Therefore,

arks the local entity type of the UMG8900 must be signaling_Gateway.

Desc Add an M3UA destination entity data record. Destination Entity Index is set
riptio to 0, and Destination Entity Name is set to MSX-A, indicating the
n MSOFTX3000 (with AS embedded) directly connected to the UMG8900. The
local entity index corresponding to the destination entity index is 0.
Scrip ADD M3DE: DEX=0, DEN="MSX-A", DET=AS, NI=NAT, DPC=H'A0001, ADF=YES,

t LEX=0;
Rem The value of Route Context can be a decimal number or null (not configured).

Desc Add an M3UA link set data record where Link Set Index is 0, Link Set
riptio Name is MSX-A, Adjacent Destination Entity Index is 0, and Traffic
n Mode is Loadshare_Mode.
Scrip ADD M3LKS: LSX=0, LSN="MSX-A", ADX=0, TM=LOADSHARE, WM=SGP;

t

arks MSOFTX3000 and the UMG8900. Therefore, Adjacent Destination Entity
Index must be set to the index of the MSOFTX3000. Here, Adjacent
Destination Entity Index is set to 0.
l The local entity type of the MSOFTX3000 is signaling_Gateway. Therefore,
the working mode in the command must be SGP.
The traffic mode of the link set must be consistent with that of the
MSOFTX3000 side; otherwise, all M3UA links of the link set cannot work
properly. Usually, set Traffic Mode to Loadshare_Mode.

Desc Add an M3UA route data record. The index of the destination entity of the
riptio MSOFTX3000 (AS) is 0.
n
Scrip ADD M3RT: RN=" MSX-A", DEX=0, LSX=0;

t

Desc Add two M3UA link data records. For one M3UA link, set CS Mode to
riptio SERVER, Board No. to 0, Board Type to SPF, Link Name to MSX01-A1,
n Local Port to 6000, and Remote Port to 6000. For the other M3UA link, set
CS Mode to SERVER, Board No. to 1, Board Type to SPF, Link Name to
MSX01-A2, Local Port to 6001, and Remote Port to 6001.
Scrip ADD M3LNK: LNK=0, BT=SPF, BN=0, LKN="MSX-A1", LIP1="192.168.10.10",

t LP=6000, RIP1="192.168.0.10", RP=6000, CS=SERVER, LSX=0, ASF=ACTIVE;
ADD M3LNK: LNK=1, BT=SPF, BN=1, LKN="MSX-A2", LIP1="192.168.10.10",
LP=6001, RIP1="192.168.0.10", RP=6001, CS=SERVER, LSX=0, ASF=ACTIVE;
Rem The traffic mode is Loadshare_Mode. Therefore, all the M3UA links in a link
arks set must be in the ACTIVE state. To ensure the reliability of M3UA links, those
M3UA links to the same destination signaling point must be assigned to
different SPFs.
Configure M3UA data that is used for the national reserved network.
riptio Entity Name is MGW1-B, Source signaling Point Code is 101, and Local
Scrip ADD M3LE: LEX=1, LEN="MGW1-B", LET=SG, NI=NATB, OPC=H'101;

t

riptio to 1, and Destination Entity Name is set to MSX-B, indicating the
n MSOFTX3000 (AS) that is directly connected to the UMG8900. The local
entity index corresponding to the destination entity index is 1.
Scrip ADD M3DE: DEX=1, DEN="MSX-B", DET=AS, NI=NATB, DPC=H'001, ADF=YES,

t LEX=1;
Desc Add an M3UA link set data record where Link Set Name is MSX01-B,
riptio Adjacent Destination Entity Index is 1, and Traffic Mode is
n Loadshare_Mode.
Scrip ADD M3LKS: LSX=1, LSN="MSX01-B", ADX=1, TM=LOADSHARE, WM=SGP;

t



n
Scrip ADD M3RT: RN=" MSX-B", DEX=1, LSX=1;

t

Desc Add two M3UA link data records. For one link, set CS Mode to SERVER,
riptio Board No. to 0, Board Type to SPF, Link Name to MSX-B1, Local Port to
n 6010, and Remote Port to 6010. For the other link, set CS Mode to
SERVER, Board No. to 1, Board Type to SPF, Link Name to MSX-B2,
Local Port to 6011, and Remote Port to 6011.
Scrip ADD M3LNK: LNK=2, BT=SPF, BN=0, LKN="MSX-B1", LIP1="192.168.10.10",

ADD M3LNK: LNK=3, BT=SPF, BN=1, LKN="MSX-B2", LIP1="192.168.10.10",
different SPFs.
Step 5 Configure MTP3 data for interworking with the BSC.

1. Add an MTP3 destination signaling point data record.
Desc Add an MTP3 destination signaling point data record where DSP index is 0,
riptio DSP name is BSC1, and OPC index is 0.
n
Scrip ADD N7DSP: INDEX=0, NAME="BSC1", NI=NATB, DPC=H'A01, OSPINDEX=0 ,

t STP=YES, ADJACENT=YES;
Rem The value of STP function on the UMG8900 must be consistent with the value
arks of STP function on BSC1. Otherwise, the MTP3 link cannot be set up properly.
2. Add an MTP3 link set data record.

Desc Add an MTP3 link set data record where Linkset index is 0, Linkset name is
riptio BSC1, and Adjacent DSP index is 0.
n
Scrip ADD N7LKS: INDEX=0, NAME="BSC1", DSPIDX=0;

t
Rem There are MTP3 signaling links in the straight-through connection mode
arks between the MSOFTX3000 and the UMG8900. Therefore, Adjacent DSP
index must be set to the index of MSX. Here, Adjacent DSP index is set to
1.
3. Add an MTP3 route data record.

Desc Add an MTP3 route data record where Route index is 0, and Route name is
riptio 0.
n
Scrip ADD N7RT: INDEX=0, NAME="BSC1", LKSIDX=0, DSPIDX=0;

t
Rem The destination signaling point of the MTP3 route is BSC1. Therefore, the route
arks index is 0.
4. Add two TDM resource of VMGW data records.

Desc Add two TDM resource of VMGW data records. For one TDM termination,
riptio set Board type to E32, Board No. to 0, Start TID to 0, End TID to 31, and
n Virtual media gateway ID to 0. For the other TDM termination, set Board
type to E32, Board No. to 1, Start TID to 1024, End TID to 1055, and Virtual
media gateway ID to 0.
Scrip ADD TDMIU: BT=E32, BN=0, TIDFV=0, TIDLV=31, VMGWID=0, HOSTID=30,

t RT=Extern, CASNO=0;
ADD TDMIU: BT=E32, BN=1, TIDFV=1024, TIDLV=1055, VMGWID=0,
HOSTID=30, RT=Extern, CASNO=0;
Rem The relay type of the signaling timeslot is External timeslot.

arks
5. Add two MTP2 signaling link data records.

Desc Add two MTP2 signaling link data records. For one MTP2 signaling link, set
riptio Interface board No. to 0, Interface board type to E32, E1T1 No. to 0, Start
n time slot to 16, and Link type to MTP3 64K LINK. For the other MTP2
signaling link, set Interface board No. to 1, Interface board type to E32,
E1T1 No. to 0, Start time slot to 16, and Link type to MTP3 64K LINK.
Scrip ADD MTP2LNK: LNKNO=0, LNKNAME="BSC1-1", IFBT=E32, IFBN=0, E1T1N=0,

t STRTTS=16, SPFBN=0, SUBBN=0, LNKTYPE=MTP364K;
ADD MTP2LNK: LNKNO=1, LNKNAME="BSC1-2", IFBT=E32, IFBN=1, E1T1N=0,
STRTTS=16, SPFBN=1, SUBBN=0, LNKTYPE=MTP364K;

Rem To ensure the reliability of MTP2 links, those MTP2 links must be assigned to
arks different SPFs.
6. Add two MTP3 link data records.
Desc Add two MTP3 link data records. For one MTP3 link, set Linkset index to 0,
riptio and MTP2 link index to 0. For the other MTP3 link, set Linkset index to 0,
n and MTP2 link index to 1.
Scrip ADD N7LNK: INDEX=0, NAME="BSC1-1", LKSIDX=0, SLC=0, MTP2NO=0;

t ADD N7LNK: INDEX=1, NAME="BSC1-2", LKSIDX=0, SLC=1, MTP2NO=1;
Rem The signaling link code must be negotiated with BSC1. Otherwise, the MPT3
arks link cannot be set up properly.
Step 6 Configure MTP3 data for interworking with the MSC2.

riptio DSP name is MSX02, and OPC index is 1.
n
Scrip ADD N7DSP: INDEX=1, NAME="MSX02", NI=NAT, DPC=H'C0002, OSPINDEX=1 ,

arks of STP function on MSX02. Otherwise, the MTP3 link cannot be set up
properly.
riptio MSX02, and Adjacent DSP index is 1.
n
Scrip ADD N7LKS: INDEX=1, NAME="MSX02", DSPIDX=1;

t
index must be set to the index of MSX02. Here, Adjacent DSP index is set to
1.
riptio MSX02.
n
Scrip ADD N7RT: INDEX=1, NAME="MSX02", LKSIDX=1, DSPIDX=1;

t

Rem The destination signaling point of the MTP3 route is MSX02. Therefore, the
arks route index is 1.
4. Add TDM resource of VMGW data records.

riptio set Board type to E32, Board No. to 2, Start TID to 2080, End TID to
n 2111, and Virtual media gateway ID to 1. For the other TDM termination, set
Board type to E32, Board No. to 3, Start TID to 3104, End TID to 3135, and
Virtual media gateway ID to 0.
Scrip ADD TDMIU: BT=E32, BN=2, TIDFV=2080, TIDLV=2111, VMGWID=0,

t HOSTID=31, RT=Extern, CASNO=0;

arks

Scrip ADD MTP2LNK: LNKNO=2, LNKNAME="MSX02-1", IFBT=E32, IFBN=2, E1T1N=1,

ADD MTP2LNK: LNKNO=3, LNKNAME="MSX02-2", IFBT=E32, IFBN=3, E1T1N=1,
6. Add an MTP3 link data record.

Scrip ADD N7LNK: INDEX=0, NAME="MSX02-1", LKSIDX=1, SLC=0, MTP2NO=2;

t ADD N7LNK: INDEX=1, NAME="MSX02-2", LKSIDX=1, SLC=1, MTP2NO=3;
Rem The signaling link code must be negotiated with TDMMSC. Otherwise, the
arks MPT3 link cannot be set up properly.
----End


used in the multi-MGW networking by taking a typical multi-MGW networking mode as an
example.
Multiple MGWs coordinate to provide services in multi-MGW networking mode, thus
improving the traffic processing capability. The multi-MGW networking is applicable to the
scenarios in which the traffic is heavy or the geographic locations of the NEs are decentralized.
Figure 3-5 shows the typical multi-MGW VoIP networking mode.
Figure 3-5 Multi-MGW VoIP networking
MSC Server
HLR
STP
CE1 CE2
MSC1
MGW1 MGW2 MGW3
TDM
IP
BSC MSC2

l The interworking between the MSC server and MSC1 is based on the BICC protocol. The
MSC server interworks with MSC1 in direct M3UA connection mode. The structure of the
protocol stack on the MSC server side is BICC/M3UA/SCTP/IP, which is the same as that
on the MSC1 side.
l The interworking between the MSC server and MSC2 is based on the ISUP protocol. The
MSC server interworks with MSC2 in non-direct M3UA connection mode. The MGW (SG)

is used as the STP. The structure of the protocol stack on the MSC server side is ISUP/
SCCP/M3UA/SCTP/IP. The structure of the protocol stack on the MSC2 side is ISUP/
SCCP/MTP3/MTP2/MTP1.
l The interworking between the MSC server and the BSC is based on the BSSAP protocol.
The MSC server interworks with the BSC in M3UA connection mode. The MGW (SG) is
used as the STP. The structure of the protocol stack on the MSC server side is BSSAP/
SCCP/M3UA/SCTP/IP. The structure of the protocol stack on the BSC side is BSSAP/
l The interworking between the MSC server and the MGW is based on the H.248 protocol
and IP bearer. The structure of the protocol stack on the MSC server side is H.248/SCTP/
IP, which is the same as that on the MGW side.
l The MGWs interwork with each other through the IP bearer.
Interworking Data
For the multi-MGW networking, you must plan and collect the following data:
l IP address
l SCTP port number

This section describes the procedure of the data configuration on the multi-MGW VoIP
networking.
This section describes the interworking data to be negotiated in the multi-MGW VoIP
networking.
This section describes the VoIP-related data configuration on the MSOFTX3000 in the multi-
This section describes the VoIP-related data configuration on the UMG8900 in the multi-MGW
VoIP networking.

This section describes the procedure of the data configuration on the multi-MGW VoIP
networking.
Figure 3-6 shows the flow chart of the data configuration on the multi-MGW VoIP networking.

Figure 3-6 Procedure of the data configuration on the multi-MGW VoIP networking
Start
Load a license file
Set local office information
Configure data for

interworking between the MSC
server and the MGW
Configure data for

Configure data for

Configure data for

interworking with the BSC
End

This section describes the interworking data to be negotiated in the multi-MGW VoIP
networking.
Table 3-3 Office information data (multi-MGW)
Local MSC number 8613900001
Local VLR number 8613900001

Mobile country code 460
Mobile network code 00
Country code 86
Local area code 755
Call source name CALL_SRC1
Table 3-4 Interworking data between the MSC server and other NEs (multi-MGW)
Name MSC MGW1 MGW2 MGW3 BSC MSC1 MSC2
Server
Node MSX MGW1 MGW2 MGW3 BSC1 MSX01 MXS0

name 2
Signaling National National National National - Nationa Nation

point network: network: network network l al
name A0001 (in A1001 : A1002 : A1003 network networ
24-bit code : C0001 k:
format) C0002
National National - - Nationa - -

reserved reserved l
network: 001 network: reserve
(in 14-bit 101 d
code format) network
: A01
SCTP H.248 link: H.248 H.248 H.248 - Module -

port 5000, 5100, link: link: link: 141:
number and 5200 5000 and 5100 5200 7000
Module 141: 5001 and and Module
6000, 6010, SPF 5101 5201 142:
and 7000 board 0: 7001
Module 142: 6000 and
6001, 6011, 6010
and 7001 SPF
board 1:
6001 and
6011

Name MSC MGW1 MGW2 MGW3 BSC MSC1 MSC2

Server
IP Signaling Signalin Signalin Signalin - Signali -

address plane: g plane: g plane: g plane: ng
192.168.0.10 192.168. 192.168. 192.168 plane:
/255.255.0.0 10.10/25 10.20/2 . 192.168
5.255.0. 55.255.0 10.30/2 .
0 .0 55.255. 20.10/2
Bearer Bearer 0.0 55.255.
plane: plane: Bearer 0.0
10.168.1 10.168.1 plane: Bearer
0.10/255 0.20/25 10.168. plane:
.255.0.0 5.255.0. 10.30/2 10.168.
0 55.255. 20.10/2
0.0 55.255.
0.0
Speech The TrFo function, the 2198 redundancy function of level 1 (PT value is 99), the
Codec UMTS AMR2 12.2K codec, and the G.711a codec are supported. The IP bearer
is adopted.

This section describes the VoIP-related data configuration on the MSOFTX3000 in the multi-
Procedure

iption


riptio
n


REDCODEC=G711a;
2. Set a mobile local office information data record.

Desc Set a mobile local office information data record.
riptio
n
Scrip SET INOFFMSC: MSCN=K'8613900001, VLRN=K'8613900001, MCC=K'460,

t MNC=K'00, INNATIONPFX=K'00, NATIONPFX=K'0;
Rem This command defines the local MSC number and local VLR number in the
arks PLMN when the local MSOFTX3000 works as the VMSC or the GMSC. In
this example, the VLR is embedded in the MSOFTX3000 (MSC). Therefore,
the MSC number and the VLR number can be set to the same number.
3. Add a VLR configuration data record.

Desc Add a VLR configuration data record by setting the MSRN/HON number
ripti allocation mode.
on
Scrip ADD VLRCFG: MAXUSR=12000, MCC=K'460, CC=K'86, MSRNAM=LAI, HONAM=RAND;

t
4. Add a call source name data record.

n name is FSN1.

t

n

t DGW="192.168.0.1";

Step 3 Configure MGW data.

1. Add MGW data records.
Descr Add MGW data records where , Transport protocol is SCTP protocol, and
iptio Server/Client is Server.
n,
Scrip ADD MGW: MGWNAME="MGW1", TRNST=SCTP, CTRLMN=136, MGWDESC="MGW1",

t BCUID=1, BNCC=IP-1, DEVNAME="MGW1", SVRCLIENT=SVR, ENCT=NSUP,
SNDDTMFRATE=300,
ADD MGW: MGWNAME="MGW2", TRNST=SCTP, CTRLMN=136, MGWDESC="MGW2",
BCUID=1, BNCC=IP-1, DEVNAME="MGW2", SVRCLIENT=SVR, ENCT=NSUP,
SNDDTMFRATE=300,
ADD MGW: MGWNAME="MGW3", TRNST=SCTP, CTRLMN=136, MGWDESC="MGW3",
BCUID=1, BNCC=IP-1, DEVNAME="MGW3", SVRCLIENT=SVR, ENCT=NSUP,
SNDDTMFRATE=300,
Rema l When the MSOFTX3000 interworks with an MGW through an IP interface,

rks the SCTP protocol is used for transport.
l The working mode of the MSOFTX3000 in associated SCTP mode must be
consistent with that of the MGW. When Server/Client in the command is
set to Server on the MSOFTX3000 side, Server/Client on the MGW side
must be set to Client.
2. Add H.248 link data records.

Desc Set MGW name to MGW1, Transport protocol to SCTP protocol, Local
riptio IP address to 192.168.0.10, Local port number to 5000, Remote IP address
n 1 to 192.168.10.10, and Remote port number to 5000 and 5001.
Set MGW name to MGW2, Transport protocol to SCTP protocol, Local
IP address to 192.168.0.10, Local port number to 5100, Remote IP address
1 to 192.168.10.20, and Remote port number to 5100 and 5101.
Set MGW name to MGW3, Transport protocol to SCTP protocol, Local
IP address to 192.168.0.10, Local port number to 5200, Remote IP address
1 to 192.168.10.30, and Remote port number to 5200 and 5201.

Scrip ADD H248LNK: MGWNAME="MGW1 ", TRNST=SCTP, LNKNAME=" MGW1H-LNK1",

t MN=141, SLOCIP1="192.168.0.10", SLOCPORT=5000,
SRMTIP1="192.168.10.10", SRMTPORT=5000;ADD H248LNK: MGWNAME=" MGW1
", TRNST=SCTP, LNKNAME=" MGW1H-LNK2", MN=141,
SLOCIP1="192.168.0.10", SLOCPORT=5000, SRMTIP1="192.168.10.10",
SRMTPORT=5001;
ADD H248LNK: MGWNAME="MGW2 ", TRNST=SCTP, LNKNAME=" MGW2H-LNK1",
SRMTPORT=5101;
ADD H248LNK: MGWNAME="MGW3 ", TRNST=SCTP, LNKNAME=" MGW3H-LNK1",
SRMTPORT=5201;
Rem l When H.248 messages are transmitted between the MSOFTX3000 and an
arks MGW through the SCTP protocol, you can configure up to 30 H.248 links
between the MSOFTX3000 and the MGW.
l The MSOFTX3000 must act as the server. If the MSOFTX3000 acts as the
client, the virtual MGW cannot register and cannot be activated.
l If multiple H.248 links are configured between the MSOFTX3000 and an
MGW, it is recommended that those H.248 links adopt the same local port
number.
3. Modify a dual-homing server node data record.

Desc Modify a dual-homing server node data record where Internal MGW media
riptio type is IP, Internal MGW connection type is All MGW connection, and
n Internal MGW path select mode is Automatic.
Scrip MOD SRVNODE: SN=LOCAL, IMMT=IP, IMCT=ALLMGW, IMSM=AUTO;

t
Rem The preceding is the default configuration, which can be modified according
arks to the actual networking.
Step 4 Configure M3UA data for interworking with the MGW1.

The M3UA data configured by using the following operations is used for the national network.
Desc Add an M3UA local entity data record where Local entity name is MSX-A,
Scrip ADD M3LE: LENM="MSX-A", NI=NAT, OPC="A0001", LET=AS;

t


riptio is MGW1-A, indicating the UMG8900 (SG) directly connected to the
n MSOFTX3000.
Scrip ADD M3DE: DENM="MGW1-A", LENM=" MSX-A", NI=NAT, DPC="A1001",

t STPF=TRUE, DET=SG;
signaling gateway.
switch through the UMG8900. Therefore, you must set Destination entity
type of the PSTN switch to Signaling point.

Desc Add an M3UA link set data record where Linkset name is MGW1-A, and
riptio Adjacent entity name is MGW1-A.
n
Scrip ADD M3LKS: LSNM="MGW1-A", ADNM="MGW1-A", TM=LOADSHARE, WM=ASP;

t

set to MGW1-A.

riptio UMG8900 (SG) route is MGW1-A.
n

Scrip ADD M3RT: DENM=" MGW1-A ", LSNM=" MGW1-A";

t

riptio WBSG module number to 141, Link name to MGW1-A1, Local port
A2, Local port number to 6001, and Peer port number to 6001.
Scrip ADD M3LNK: MN=141, LNKNM="MGW1-A1", LOCIP1="192.168.0.10",

MGW1-A", ASF=YES;
ADD M3LNK: MN=142, LNKNM="MGW1-A2", LOCIP1="192.168.0.10",
MGW1-A", ASF=YES;
The M3UA data configured by the following operations are used for the national reserved
network.
Desc Add an M3UA local entity data record where Local entity name is MSX-B,
Scrip ADD M3LE: LENM="MSX-B", NI=NATB, OPC="001", LET=AS;

t

riptio is MGW1-B, indicating the UMG8900 (SG) directly connected to the
n MSOFTX3000.
Scrip ADD M3DE: DENM="MGW1-B", LENM="MSX-B", NI=NATB, DPC="101",

t STPF=TRUE, DET=SG, NM=INCLUSIVE;

signaling gateway.
switch through the UMG8900. Therefore, you must set the Destination
entity type of the PSTN switch to Signaling point.

Desc Add an M3UA link set data record where Linkset name is MGW1-B, and
riptio Adjacent entity name is MGW1-B.
n
Scrip ADD M3LKS: LSNM="MGW1-B", ADNM="MGW1-B", TM=LOADSHARE, WM=ASP;

t

set to MGW1-B.

riptio UMG8900 (SG) route is MGW1-B.
n
Scrip ADD M3RT: DENM=" MGW1-B ", LSNM=" MGW1-B";

t

riptio WBSG module number to 141, Link name to MGW1-B1, Local port
B2, Local port number to 6011, and Peer port number to 6011.

Scrip ADD M3LNK: MN=141, LNKNM="MGW1-B1", LOCIP1="192.168.0.10",

MGW1-B", ASF=YES;
ADD M3LNK: MN=142, LNKNM="MGW1-B2", LOCIP1="192.168.0.10",
MGW1-B", ASF=YES;

n

t FALSE, DET=AS;
application sever.
n

t
n

t




n


riptio
n

t

riptio
n

t

riptio
n

t

Desc Add a BICC trunk group data record where Trunk group name is MSX01-
riptio MGW1, and MGW name is MGW1.
n
Scrip ADD BICCTG: TGN="MSX01-MGW1", MGWNAME="MGW1", SRN="MSX01", BTG=1,


riptio
n

t
to the MGW.

n
Scrip ADD M3DE: DENM="MSX02", LENM="MSX-A", NI=NAT, DPC="C0002", STPF=

t FALSE, DET=SP;
Rem The ISUP signaling is exchanged between the MSOFTX3000 and the MSC2
arks through the UMG8900. Therefore, you must set Destination entity type of the
MSC2 to Signaling point.

n
Scrip ADD M3RT: DENM="MSX02", LSNM="MGW1-A";

t
Desc Add an SCCP destination signaling point where DSP name is MSX02 and
riptio DPC is C0002.
n
Scrip ADD SCCPDSP: DPNM="MSX02", NI=NAT, DPC="C0002", OPC="A0001";

t
riptio SCMGMSCVLR.
n
Scrip ADD SCCPSSN: SSNNM="MSX02-SCMG", NI=NAT, SSN=SCMG, SPC="C0002",

t OPC="A0001";
ADD SCCPSSN: SSNNM="MSX02-MSC", NI=NAT, SSN=MSC, SPC="C0002",
OPC="A0001";
ADD SCCPSSN: SSNNM="VLR", NI=NAT, SSN=VLR, SPC="C0002", OPC="A0001";
riptio and DPC1 is C0002.
n
Scrip ADD OFC: ON="MSX02", OOFFICT=NATT, DOL=SAME, DOA=PSTN, BOFCNO=11,

t OFCTYPE=COM, SIG=NONBICC/NONSIP, NI=NAT, DPC1="C0002";
Rem This office direction has SS7 trunk circuits. Therefore, you must specify
arks DPC in the command. Otherwise, errors occur when you add an SS7 trunk
group data record by running ADD N7TG.

riptio
n

t


riptio
n

t
riptio
n

t
9. Add an SS7 trunk group data record.
Desc Add an SS7 trunk group data record where Trunk group name is MSX02, and
riptio MGW name is MGW1.
n
Scrip ADD N7TG: TGN="MSX02", MGWNAME="MGW1", CT=ISUP, SRN="MSX02",

t G=INOUT, BTG=20, SOPC=" A0001", SDPC="C0002", CSCNAME="CALL_SRC1",
ICR=LCO-1&LC-1&LCT-1&NTT-1&ITT-1, CC=NO, RELRED=NO;
Step 7 Configure data for interworking with the BSC.

n
Scrip ADD M3DE: DENM="BSC1", LENM="MSX-B", NI=NATB, DPC="A01", STPF=

t FALSE, DET=SP;
Rem The signaling is exchanged between the MSOFTX3000 and the BSC through
arks the UMG8900. Therefore, you must set Destination entity type of the BSC to
Signaling point.
n
Scrip ADD M3RT: DENM="BSC1", LSNM="MGW1-B";

t

Desc Add an SCCP destination signaling point where DSP name is BSC1DPC is
riptio A01.
n
Scrip ADD SCCPDSP: DPNM="BSC1", NI=NATB, DPC="A01", OPC="001";

t

riptio SCMGBSSAP.
n
Scrip ADD SCCPSSN: SSNNM="BSC1-SCMG", NI=NATB, SSN=SCMG, SPC="A01",

t OPC="001";
ADD SCCPSSN: SSNNM="BSC1-BSSAP", NI=NATB, SSN=BSSAP, SPC="A01",
OPC="001";
5. Add an office direction.

Desc Add an office direction where Office direction name is BSC1.
riptio
n
Scrip ADD OFC: ON="BSC1", OOFFICT=BSC, DOL=LOW, DOA=BSC, BOFCNO=1,

t OFCTYPE=COM, SIG=NONBICC/NONSIP, NI=NATB, DPC1="A01", SVQE=NO;
6. Add a BSC.
Desc Add a BSC where Route selection source name is RSSN1 and LAI
riptio number is 460000003.
n
Scrip ADD BSC: DPC="A01", OPC="001", BSCNM="BSC1", RSSCNAME="RSSN1",

t MLAIF=NO, LAI="460000003", BSCBEARERTYPE=TDM;
7. Add a radio access network media gateway.

Desc Add a radio access network media gateway where Office direction name is
riptio BSC1 and Media gateway name is MGW1.
n
Scrip ADD RANMGW: OFFICENAME="BSC1", MGWNAME="MGW1";

t
8. Add 2G LAI or GCI informations.

Desc Add 2G LAI or GCI informations where Global cell ID are 460000003 and
riptio 4600000030003, Call source name is CALL_SRC1.
n
Scrip ADD LAIGCI: GCI="460000003", LAIGCINAME="BSC1-LAI",

t MSCN="8613900001", VLRN="8613900001", NONBCLAI=NO, LAICAT=LAI,
ADD LAIGCI: GCI="4600000030003", LAIGCINAME="BSC1-GCI",
MSCN="8613900001", VLRN="8613900001", NONBCLAI=NO, LAICAT=GCI,
9. Add an A-interface circuit pool.
Desc Add an A-interface circuit pool where Circuit pool name is BSC1 .
riptio
n
Scrip ADD ACPOOL: CPNAME="BSC1",

t BCR=SFR1-1&SFR2-1&SFR3-1&SHR1-1&SHR2-1&SHR3-1&DFR1-1&DHR1-1;
10. Add an A-interface trunk group.
Desc Add an A-interface trunk group where Trunk group name is BSC1 and MGW
riptio name is MGW1.
n
Scrip ADD AIETG: TGNAME="BSC1", MGWNAME="MGW1", SOPC="001", SDPC="A01",

t CPNAME="BSC1", BTG=2;
11. Add A-interface trunk circuits.
Desc Add A-interface trunk circuits where Start CIC is 640 and End CIC is 703.
riptio
n
Scrip ADD AIETKC: MN=23, TGNAME="BSC1", SCIC=640, ECIC=703, TID=640;

t
----End

This section describes the VoIP-related data configuration on the UMG8900 in the multi-MGW
VoIP networking.
Procedure

1. Set local office information.
Desc Set a local office information data record. The signaling point index of the
riptio national network is 0, and the signaling point index of the national reserved
n network is 1.
Scrip SET OFI: NAME="MGW1", NATRESVLD=YES, SERACH0=NATB, NATRESOPC=H'101,

t NATRESLEN=LABEL14, SPFLAG=NO, STPFLAG=YES;
ADD OFI: INDEX=1, NATRESOPC=H'A1001;
Rem l SET OFI is used to set the local information data record where the signaling
arks point index of the national network is 0.
l ADD OFI is used to add multiple signaling point data records.
Usually, the signaling point function is disabled, and the signaling transfer
function is enabled.

riptio
n
Scrip ADD IPADDR: BT=MPU, BN=1, IFT=ETH, IFN=0, IPADDR="192.168.10.10",

t MASK="255.255.0.0", FLAG=MASTER, INVLAN=NO;
3. Add a static route data record.
Desc Add a static route data record.

riptio
n
Scrip ADD ROUTE: BT=MPU, BN=1, DSTIP="0.0.0.0", DSTMASK="0.0.0.0",

t RTTYPE=NEXTHOP, NEXTHOP="192.168.0.1";
Rem l If the MPU board is configured with the centralized transfer function, the
arks default route must be configured.
l The next hop address is the IP address (192.168.0.1) of the router that is
directly connected to the UMG8900.
Step 2 Configure data for IP bearer network.


riptio
n


n

t TIMEOUT=Aging;
Step 3 Configure data for interworking with the MSOFTX3000.

1. Set a virtual MGW data record.
Desc Set a virtual MGW data record where Virtual media gateway id is 0, and
riptio Virtual media gateway MID type is IP.
n
Scrip SET VMGW: VMGWID=0, MIDTYPE=IP, MID="192.168.10.10:5000";

t
2. Add an MGW controller data record.

Desc Add an MGW controller data record where Virtual media gateway id is 0,
riptio Media gateway controller No. is 0, and Media gateway controller MID
n type is IP.
Scrip ADD MGC: VMGWID=0, MGCIDX=0, MIDTYPE=IP, MID="192.268.0.10:5000",

t MSS=MASTER;
3. Add two H.248 signaling link data records.

Desc Add two H.248 signaling link data records. For one H.248 link, set Transfer
riptio protocol type to SCTP, Local Master IP address to 192.168.100.1, Local
n port No. to 5000, Peer Master address to 192.168.0.1, and Peer port No. to
5000. For the other H.248 link, set Transfer protocol type to SCTP, Local
Master IP address to 192.168.100.1, Local port No. to 5001, Peer Master
address to 192.168.0.1, and Peer port No. to 5000.
Scrip ADD H248LNK: LINKID=48, VMGWID=0, MGCIDX=0, TT=SCTP,

t LOCALIP="192.168.10.10", LOCALPORT=5000, PEERIP="192.168.0.10",
ADD H248LNK: LINKID=49, VMGWID=0, MGCIDX=0, TT=SCTP,
LOCALIP="192.168.10.10", LOCALPORT=5001, PEERIP="192.168.0.10",
Rem l The numbers of the H.248 links of the SCTP type configured on the
arks UMG8900 must be greater than 47.
4. Activate a virtual MGW.

Desc Activate a virtual MGW.

riptio
n
Scrip ACT VMGW: VMGWID=0;

t

Configure M3UA data that is used for the national network.
riptio Entity Name is MGW1-A, Source signaling Point Code is A1001, and Local
Scrip ADD M3LE: LEX=0, LEN="MGW1-A", LET=SG, NI=NAT, OPC=H'A1001;

t


riptio to 0, and Destination Entity Name is set to MSX-A, indicating the
n MSOFTX3000 (with AS embedded) directly connected to the UMG8900. The
local entity index corresponding to the destination entity index is 0.
Scrip ADD M3DE: DEX=0, DEN="MSX-A", DET=AS, NI=NAT, DPC=H'A0001, ADF=YES,

t LEX=0;

Desc Add an M3UA link set data record where Link Set Index is 0, Link Set
riptio Name is MSX-A, Adjacent Destination Entity Index is 0, and Traffic
n Mode is Loadshare_Mode.
Scrip ADD M3LKS: LSX=0, LSN="MSX-A", ADX=0, TM=LOADSHARE, WM=SGP;

t


n
Scrip ADD M3RT: RN=" MSX-A", DEX=0, LSX=0;

t
Desc Add two M3UA link data records. For one M3UA link, set CS Mode to
riptio SERVER, Board No. to 0, Board Type to SPF, Link Name to MSX01-A1,
n Local Port to 6000, and Remote Port to 6000. For the other M3UA link, set
CS Mode to SERVER, Board No. to 1, Board Type to SPF, Link Name to
MSX01-A2, Local Port to 6001, and Remote Port to 6001.
Scrip ADD M3LNK: LNK=0, BT=SPF, BN=0, LKN="MSX-A1", LIP1="192.168.10.10",

ADD M3LNK: LNK=1, BT=SPF, BN=1, LKN="MSX-A2", LIP1="192.168.10.10",
different SPFs.

t



riptio to 1, and Destination Entity Name is set to MSX-B, indicating the
Scrip ADD M3DE: DEX=1, DEN="MSX-B", DET=AS, NI=NATB, DPC=H'001, ADF=YES,

t LEX=1;

Desc Add an M3UA link set data record where Link Set Name is MSX01-B,
riptio Adjacent Destination Entity Index is 1, and Traffic Mode is
n Loadshare_Mode.
Scrip ADD M3LKS: LSX=1, LSN="MSX01-B", ADX=1, TM=LOADSHARE, WM=SGP;

t


n
Scrip ADD M3RT: RN=" MSX-B", DEX=1, LSX=1;

t


Scrip ADD M3LNK: LNK=2, BT=SPF, BN=0, LKN="MSX-B1", LIP1="192.168.10.10",

ADD M3LNK: LNK=3, BT=SPF, BN=1, LKN="MSX-B2", LIP1="192.168.10.10",
different SPFs.
Step 5 Configure MTP3 data for interworking with the BSC.

riptio DSP name is BSC1, and OPC index is 0.
n
Scrip ADD N7DSP: INDEX=0, NAME="BSC1", NI=NATB, DPC=H'A01, OSPINDEX=0 ,

arks of STP function on BSC1. Otherwise, the MTP3 link cannot be set up properly.

riptio BSC1, and Adjacent DSP index is 0.
n
Scrip ADD N7LKS: INDEX=0, NAME="BSC1", DSPIDX=0;

t
index must be set to the index of MSX. Here, Adjacent DSP index is set to
1.

riptio 0.
n
Scrip ADD N7RT: INDEX=0, NAME="BSC1", LKSIDX=0, DSPIDX=0;

t
Rem The destination signaling point of the MTP3 route is BSC1. Therefore, the route
arks index is 0.
4. Add two TDM resource of VMGW data records.

riptio set Board type to E32, Board No. to 0, Start TID to 0, End TID to 31, and
n Virtual media gateway ID to 0. For the other TDM termination, set Board
type to E32, Board No. to 1, Start TID to 1024, End TID to 1055, and Virtual
media gateway ID to 0.
Scrip ADD TDMIU: BT=E32, BN=0, TIDFV=0, TIDLV=31, VMGWID=0, HOSTID=30,

t RT=Extern, CASNO=0;

arks

Scrip ADD MTP2LNK: LNKNO=0, LNKNAME="BSC1-1", IFBT=E32, IFBN=0, E1T1N=0,

ADD MTP2LNK: LNKNO=1, LNKNAME="BSC1-2", IFBT=E32, IFBN=1, E1T1N=0,
6. Add two MTP3 link data records.

Scrip ADD N7LNK: INDEX=0, NAME="BSC1-1", LKSIDX=0, SLC=0, MTP2NO=0;

t ADD N7LNK: INDEX=1, NAME="BSC1-2", LKSIDX=0, SLC=1, MTP2NO=1;
Rem The signaling link code must be negotiated with BSC1. Otherwise, the MPT3
arks link cannot be set up properly.
Step 6 Configure MTP3 data for interworking with the MSC2.

riptio DSP name is MSX02, and OPC index is 1.
n
Scrip ADD N7DSP: INDEX=1, NAME="MSX02", NI=NAT, DPC=H'C0002, OSPINDEX=1 ,


arks of STP function on MSX02. Otherwise, the MTP3 link cannot be set up
properly.

riptio MSX02, and Adjacent DSP index is 1.
n
Scrip ADD N7LKS: INDEX=1, NAME="MSX02", DSPIDX=1;

t
index must be set to the index of MSX02. Here, Adjacent DSP index is set to
1.

riptio MSX02.
n
Scrip ADD N7RT: INDEX=1, NAME="MSX02", LKSIDX=1, DSPIDX=1;

t
Rem The destination signaling point of the MTP3 route is MSX02. Therefore, the
arks route index is 1.
4. Add TDM resource of VMGW data records.

riptio set Board type to E32, Board No. to 2, Start TID to 2080, End TID to
n 2111, and Virtual media gateway ID to 1. For the other TDM termination, set
Board type to E32, Board No. to 3, Start TID to 3104, End TID to 3135, and
Virtual media gateway ID to 0.
Scrip ADD TDMIU: BT=E32, BN=2, TIDFV=2080, TIDLV=2111, VMGWID=0,

t HOSTID=31, RT=Extern, CASNO=0;

arks

Scrip ADD MTP2LNK: LNKNO=2, LNKNAME="MSX02-1", IFBT=E32, IFBN=2, E1T1N=1,

ADD MTP2LNK: LNKNO=3, LNKNAME="MSX02-2", IFBT=E32, IFBN=3, E1T1N=1,
6. Add an MTP3 link data record.

Scrip ADD N7LNK: INDEX=0, NAME="MSX02-1", LKSIDX=1, SLC=0, MTP2NO=2;

t ADD N7LNK: INDEX=1, NAME="MSX02-2", LKSIDX=1, SLC=1, MTP2NO=3;
Rem The signaling link code must be negotiated with TDMMSC. Otherwise, the
arks MPT3 link cannot be set up properly.
Operator just need to configur data for interworking with the MSC Server on MGW2 and
MGW3. The configuration is similar with the data on MGW1. Therefor, only the configuration
on MGW1 is described here.
----End
3.3 CMN Networking

used in the CMN networking by taking a typical CMN networking mode as an example.
In the CMN networking, the MSC server independently provides the CMN function, transfers
calls, and routes calls. The MGWs controlled by the active and standby MSC servers that are
connected to the CMN are directly connected through the IP bearer network.
Figure 3-7 shows the typical CMN VoIP networking.

Figure 3-7 CMN VoIP networking
CMN
MSC MSC
Server1 Server2
MGW1 MGW2 MGW3 MGW4
TDM
BSC IP MSC

l The CMN call flow is a BICC incoming and outgoing call flow, which does not control the
MGW.
l The CMN is not involved in the codec negotiation.
l The interworking between the CMN and MSC server1 is based on the BICC protocol. The
CMN interworks with MSC server1 in direct M3UA connection mode. The structure of the
protocol stack on the CMN side is BICC/M3UA/SCTP/IP, which is the same as that on the
MSC server1 side.
l The interworking between the CMN and MSC server2 is based on the BICC protocol. The
CMN interworks with MSC server2 in direct M3UA connection mode. The structure of the
protocol stack on the CMN side is BICC/M3UA/SCTP/IP, which is the same as that on the
MSC server2 side.
l MGWs are directly connected through the IP bearer network.
Interworking Data
For the CMN networking, you must plan and collect the following data:
l IP address

l SCTP port number

This section describes the procedure of the data configuration on the CMN VoIP networking.
This section describes the interworking data to be negotiated in the CMN VoIP networking.
This section describes the VoIP-related data configuration on the MSOFTX3000 in the CMN
VoIP networking.

This section describes the procedure of the data configuration on the CMN VoIP networking.
Figure 3-8 shows the procedure of the data configuration on the CMN VoIP networking.
Figure 3-8 Procedure of the data configuration on the CMN VoIP networking
Start
Load a license file
Set local office

information
Configure data for

Configure data for

End

This section describes the interworking data to be negotiated in the CMN VoIP networking.

Table 3-5 Interworking data between the MSC server and other NEs (CMN)
Name CMN MSC Server1 MSC Server2
Node MSX MSX01 MSX02

name
Signalin National network: National network: C0001 National network: C0002

g point A0001 (in 24-bit code
name format)
SCTP Module 141: 7000 and Module 141: 7000 Module 141: 7010
port 7010 Module 142: 7001 Module 142: 7011
number Module 142: 7001 and
7011
IP Signaling plane: Signaling plane: Signaling plane:

address 192.168.0.10/255.255.0. 192.168.20.10/255.255. 192.168.20.20/255.255.
0 0.0 0.0

This section describes the VoIP-related data configuration on the MSOFTX3000 in the CMN
VoIP networking.
Procedure

iption


riptio
n

REDCODEC=G711a;

2. Add call source name data records.
n name is FSN1.
Add a call source name data record where Call source name is
CALL_SRC2, Route selection source name is RSSN2, and Failure source
name is FSN2.

t ADD CALLSRC: CSCNAME="CALL_SRC2", RSSN="RSSN2", FSN="FSN2";
n

t DGW="192.168.0.1";

Desc Add an M3UA local entity data record where Local entity name is MSX,
Scrip ADD M3LE: LENM="MSX", NI=NAT, OPC="A0001", LET=AS;

t
n

Scrip ADD M3DE: DENM="MSX01", LENM="MSX", NI=NAT, DPC="C0001", STPF=

t FALSE, DET=AS;
application sever.
n

t
n

t


n


riptio
n

t

riptio
n

t
riptio
n

t
Desc Add a BICC trunk group data record where Trunk group name is MSX01,
riptio and MGW name is CMN.
n

Scrip ADD BICCTG: TGN="MSX01", MGWNAME="CMN", SRN="MSX01", BTG=1,

riptio
n

t

n
Scrip ADD M3DE: DENM="MSX02", LENM="MSX", NI=NAT, DPC="C0002", STPF=

t FALSE, DET=SP;
Rem The ISUP signaling is exchanged between the MSOFTX3000 and the TDM
arks MSC through the UMG8900. Therefore, you must set Destination entity
type of the TDM MSC to Signaling point.
n

t
n
Scrip ADD M3RT: DENM="MSX02", LSNM="MSX02";

t

riptio WBSG module number to 141, Link name to SX02-1, Local port number
n to 7010, and Peer port number to 7010; for the other link, set Client/Server
to Client, WBSG module number to 142, Link name to SX02-2, Local port
number to 7011, and Peer port number to 7011.

n

The office direction supports the TrFo function, and the peer office supports
the redundancy function.

riptio
n

t

riptio
n

t

riptio
n

t

n
Scrip ADD BICCTG: TGN="MSX02", MGWNAME="CMN", SRN="MSX02", BTG=1,


Desc Add a BICC CIC module data record where Office direction name is
riptio MSX02.
n

t
----End

VoIP User Manual 4 VoIP Networking Evolving from Network in Use
4 VoIP Networking Evolving from Network in

Use
About This Chapter
This section describes the data configuration of the VoIP bearer network.
The following aspects are involved in the VoIP networking evolving from network in use.
l The data configuration of the Nc interface between the MSC servers is modified.
In the original network, if the ISUP is used between the MSC servers, the MSC servers
interwork with each other by using the M3UA direct connection mode or the M3UA
indirect connection mode. The original ISUP data configuration is not changed. The
BICC signaling data is configured between the MSC servers using the M3UA direct
connection mode.
During the evolution to the VoIP networking, the signaling point for direct connection
is added between the MSC servers. The M3UA data and BICC signaling data is
configured.
In the original network, if the ISUP is used between the MSC servers, the MSC servers
interwork with each other by using the MTP direct connection mode or the M2UA
connection mode. When the two MSC servers must be connected through the VoIP
bearer network, the the BICC signaling is used between the two MSC servers instead
of the ISUP signaling. The M3UA direct connection mode is used. If one of the MSC
servers does not need to use the VoIP bearer networking, the data configuration of the
interface between the MSC servers is not changed.
During the evolution to the VoIP networking, the signaling point for direct connection
is added between the MSC servers and M3UA data and BICC signaling data is
configured. The data must be configured according to actual situations.
l The data configuration of the Nb interface between the MSC servers is modified.
In the original network, if the MSC server and the MGW are interconnected through
the IP networking mode, the data of the default gateways must be configured for the
MSC server and the MGW.
In the original network, if the MSC server and the MGW are interconnected through
the TDM networking mode, the data of IP interfaces and default gateways must be

4 VoIP Networking Evolving from Network in Use VoIP User Manual
configured for the MSC server and the MGW. The IP bearer networking must be used
between the MGWs.
l The data configuration of the A interface between the MSC server and the BSC is modified.
In the original network, if the MSC server and the BSC are interconnected through the
M3UA direct connection mode, the data configuration of the interface between the MSC
server and the BSC is not changed.
In the original network, if the MSC server and the BSC are interconnected through the
MTP or M2UA networking mode, the M3UA networking mode must be used between
the MSC server and the BSC. The MGW (SG) is used as the signalling transfer point.
The impact of evolution on services indicates the impact of inter-office Nc and Nb interface
changes on services. The BICC protocol evolves from the ISUP protocol, which support all the
ISUP services besides IP bearer services. In theory, the Nc interface adopts the BICC signaling
instead of the ISUP signaling, thus ensuring the forward compatibility of the services. Due to
the restrictions of the IP bearer, for example, because the packet loss, time delay, jitter, and
bandwidth are difficult to control, the IP bearer is less reliable than the TDM bearer. During the
evolution to the IP networking, the quality of the bearer network is very important. The MSC
server can use proper speech codec in cooperation with the bearer network. (The voice quality
can be assured while low bandwidth usage can be achieved.) The service quality can be assured
by using 2198 redundancy and IP CAC functions.
The IP bearer data configuration is closely related to the interworking data of interfaces between
NEs, but merely related to the entire network architecture. Therefore, this manual takes the
single-MGW networking, multi-MGW networking, and CMN networking as examples to
describe the data configuration on the typical networking.
This section describes the data configuration in the single-MGW VoIP networking mode.
This section describes the data configuration in the multi-MGW VoIP networking.
4.3 CMN Networking
This section describes the data configuration in the CMN VoIP networking mode.


This section describes the data configuration in the single-MGW VoIP networking mode.
Original Network
Figure 4-1 shows the typical single-MGW networking.
Figure 4-1 Single-MGW networking
MSC Server HLR
STP
MSC1 MGW MSC2
TDM
BSC IP
Seeing from the preceding figure, you can know that:
l The ISUP is used between the MSC server and MSC1. The MSC server interworks with
MSC1 by using the M3UA direct connection mode. The structure of the protocol stack at
the MSC server side is ISUP/SCCP/M3UA/SCTP/IP, which is the same as that at MSC1
side.
l The ISUP is used between the MSC server and MSC2. The MSC server interworks with
MSC1 by using the M3UA non-direct connection mode. The MGW (SG) is used as the
signaling transfer point. The structure of the protocol stack at the MSC server side is ISUP/
SCCP/M3UA/SCTP/IP. The structure of the protocol stack at MSC2 side is ISUP/SCCP/
MTP3/MTP2/MTP1.
l The BSSAP is used between the MSC server and the BSC. The MSC server interworks
with the BSC by using the M2UA connection mode. The MGW (SG) is used as the signaling
transfer point. The structure of the protocol stack at the MSC server side is BSSAP/SCCP/

MTP3/M2UA/SCTP/IP. The structure of the protocol stack at the BSC side is BSSAP/
Destination Network
Figure 4-2 shows the typical single-MGW VoIP networking.
Figure 4-2 Single-MGW VoIP networking
MSC Server HLR
STP
CE1 CE2
MSC1 MGW MSC2
TDM
BSC IP

l When the MSC server interworks with the NE of the access network, the SPC for the
national reserved network is used.
l The BICC is used between the MSC server and MSC1. The MSC server interworks with
MSC1 by using the M3UA direct connection mode. The structure of the protocol stack at
the MSC server side is BICC/M3UA/SCTP/IP, which is the same as that at MSC1 side.
l The BICC and ISUP are used between the MSC server and MSC2.
The ISUP is used between the MSC server and MSC2. The MSC server interworks with
MSC1 by using the M3UA non-direct connection mode. The MGW (SG) is used as the
signaling transfer point. The structure of the protocol stack at the MSC server side is
ISUP/SCCP/M3UA/SCTP/IP. The structure of the protocol stack at the MSC2 side is
ISUP/SCCP/MTP3/MTP2/MTP1.
The BICC is used between the MSC server and MSC2. The MSC server interworks
with MSC2 by using the M3UA direct connection mode. The structure of the protocol
stack at the MSC server side is BICC/M3UA/SCTP/IP, which is the same as that at
MSC2 side.

M3UA/SCTP/IP. The structure of the protocol stack at the BSC side is BSSAP/SCCP/
MTP3/MTP2/MTP1.
Evolution Method
The single-MGW networking shown in Figure 4-1 is evolved to the single-MGW VoIP
networking shown in Figure 4-2 in the following ways:
l Use the BICC signaling instead of the ISUP signaling between the MSC server and MSC1.
l Maintain the ISUP signaling between the MSC server and MSC2. Configure the data for
the new BICC signaling between the two NEs.
l Use the M3UA link instead of the M2UA link between the MSC server and the BSC.
Interworking Data
In the above evolution methods, you must plan and collect the following data:
l IP address
l SCTP port

This section introduces the data configuration flow for the single-MGW VoIP networking.
This section describes interworking data to be negotiated in the single-MGW VoIP networking
mode.
This section describes the data configuration on the MSOFTX3000 when the network is evolved
to the VoIP bearer network using the singe-MGW networking.
This section describes the data configuration on the UMG8900 in the singe-MGW VoIP
networking mode.

This section introduces the data configuration flow for the single-MGW VoIP networking.
Figure 4-3 shows the data configuration flow for the single-MGW VoIP networking.

Figure 4-3 Flow of the data configuration for the single-MGW VoIP networking
Start
Configure the IP bearer

data for the MGW
Configure the BICC

signaling between the
MSC server and MSC 1
Configure the BICC

MSC server and MSC 2.
Configure the M3UA link

between the MSC server
and the MGW

and the BSC
End

This section describes interworking data to be negotiated in the single-MGW VoIP networking
mode.
As shown in Table 4-1, the interworking data to be collected and planned during the flow shown
in Figure 4-2 is as follows:
NOTE
The interworking data must be configured according to actual situations or the data configuration of the
original network. If the interworking data is configured according to the data configuration of the original
network, the interworking data must be consistent with the data configuration of the original network. You
can check the interworking data by using the LST commands. The configuration of the interworking data
described in this section is only an example.

Table 4-1 Interworking data of single-MGW VoIP networking

Na MSC Server MGW MSC1 MSC2 BSC
me
Nod MSX MGW1 MSX01 MSX02 BSC1

e
Nam
e
SPC National National National National -

network: network: network: network:
A1001 and A1001 C0001 C0002
A0002
National National - - National

reserved reserved reserved
network: 001 network: 101 network: A01
IP Signaling Signaling Signaling Signaling -

Addr plane: plane: plane: plane:
ess 192.168.0.10/ 192.168.10.10 192.168.20.10 192.168.30.10
255.255.0.0 /255.255.0.0 /255.255.0.0 /255.255.0.0
Bearer plane: Bearer plane: Bearer plane:
10.168.10.10/ 10.168.20.10/ 10.168.30.10/
255.255.0.0 255.255.0.0 255.255.0.0
SCT Module 141: SPF board 0: Module 141: Module 141: -

P 7000, 7002, 7002 and 7006 7000 7004
Port 7004, and SPF board 1: Module 142: Module 142:
7006 7003 and 7007 7001 7005
Module 142:
7001, 7003,
7005, and
7007

This section describes the data configuration on the MSOFTX3000 when the network is evolved
to the VoIP bearer network using the singe-MGW networking.
Procedure
Step 1 Modify MGW data.
Descr Modify the MGW data record to support the IP bearer and codec.
iption
Script MOD MGW: MGWNAME="MGW1", TRNST=SCTP, BNCC=IP-1, SPCATTR=SUPCODECCFG-1,

UMTSAMR2R=RATE475-0&RATE515-0&RATE590-0&RATE670-0&RATE740-0&RATE795-0&R
ATE102-0&RATE122-1, TC=UMTSAMR2-1&PCMA-1;

Rema When the codec UMTS AMR/AMR2 is supported, set Support codec rate
rks configuration in Special attributes.
Step 2 Modify the ISUP signalling to BICC signalling between MSC Server and MSC1.
1. Modify the office direction.
Desc Modify the ISUP direction to ISUP and BICC mixed office direction.
riptio
n
Scrip MOD OFC: ON="MSX01", OOFFICT=MSC, OFCTYPE=COM, BICCSIG=M3UA,

t CCOT=YES, OPC="A0001", DPC1="C0001";
2. Modify the route analysis.
Desc Modify the route analysis to Prefer to select BICC IP.

riptio
n
Scrip MOD RTANA: RSN="MSX01", RSSN="RSSN1", CC=CAT254, ADI=ALL, TP=ALL,

t ORT=ALL, TSN="DEFAULT", ISUP=BICCIP_F;
riptio and MGW name is MGW1.
n
Scrip ADD BICCTG: TGN="MSX01", MGWNAME="MGW1", SRN="MSX01", BTG=1,

t CT=BICC_IP, ESTDIR=FORWARD, TUNNELMODE=DELAY, CNCTNOTI=NO,
RELRED=NO, HCF=FALSE;
riptio
n
Scrip ADD BICCCICMDU: ON="MSX01", SCIC=0, ECIC=63;

t
to the MGW.
Step 3 Configure BICC signalling data for interworking with MSC2.

1. Add a local office signaling point code.

Desc Add a local office signaling point code where National network code is
riptio A0002.
n
Scrip ADD OFI: IDX=1, NATC="A0002";

t
2. Add a local mobile office information.

Desc Add a local mobile office information.
riptio
n
Scrip ADD INOFFMSC: ID=1, MSCN=K'8613900001, VLRN=K'8613900001, MCC=K'460,

t MNC=K'00, INNATIONPFX=K'86;

Desc Add an M3UA local entity data record where Local entity name is MSXc,
Scrip ADD M3LE: LENM="MSXc", OPC="A0002", LET=AS;

t

n
Scrip ADD M3DE: DENM="MSX02", DPC="C0002", DET=AS;

t
Rem The M3UA signaling data is newly added, the Destination entity name and
arks Signaling point code of destination entity must be negotiated, and can not be
same to the original network.

n

t


n

t
riptio Signaling type is M3UA, BICC call source name is CALL_SRC1, and
n Multi-area name is 1.
Scrip ADD OFC: ON="MSX02", OOFFICT=MSC, DOL=SAME, DOA=MSC, BOFCNO=1,

t OFCTYPE=MIX, BICCSIG=M3UA, DPC1="C0002", OPC="A0002",
CSCNAME="CALL_SRC1", AN="0", IFADJUSTCODEC=NO, IFADJUSTCODECRATE=NO;
Rem BICC call source name and Multi-area name should be same to original
arks network.
Desc Add a subroute data record where Sub-route name is MSX02_01.

riptio
n
Scrip ADD SRT: SRN="MSX02_01", ON="MSX02", ACC1="INVALID", ACC2="INVALID",

t BFSM=INVALID;

Desc Add a route data record where Route name is MSX02_01.

riptio
n
Scrip ADD RT: RN="MSX02_01", SRSM=SEQ, SR1N="MSX02_01";

t
Desc Add a route analysis data record where Route selection name is RSN2, and
riptio Signaling priority is Prefer to select BICC IP.
n
Scrip ADD RTANA: RSN="RSN2", RSSN="RSSN1", TSN="DEFAULT", RN="MSX02_01",

t ISUP=BICCIP_F;
n
Scrip ADD BICCTG: TGN="MSX02", MGWNAME="MGW1", SRN="MSX02_01", BTG=1,

riptio MSX02.
n

t
to the MGW.
Step 4 Modify the M2UA signalling to M3UA signalling of national reserved network between MSC
Server and MGW.

Desc Add an M3UA local entity data record where Local entity name is MSXd,
Scrip ADD M3LE: LENM="MSXd", OPC="001", LET=AS;

t

riptio is MGW1, indicating the UMG8900 (SG) directly connected to the
n MSOFTX3000.
Scrip ADD M3DE: DENM="MGW1", DPC="101", STPF=TRUE, DET=SG;

t

Desc Add an M3UA link set data record where Linkset name is BSC1, and Adjacent
riptio entity name is MGW1.
n
Scrip ADD M3LKS: LSNM="BSC1", ADNM="MGW1", TM=LOADSHARE, WM=ASP;

t

set to MGW1.

riptio WBSG module number to 141, Link name to BSC1_01, Local port
Server to Client, WBSG module number to 142, Link name to BSC1_02,

Scrip ADD M3LNK: MN=141, LNKNM="BSC1_01", LOCIP1="192.168.0.10",

LSNM="BSC1", QoS=TOS;
ADD M3LNK: MN=142, LNKNM="BSC1_02", LOCIP1="192.168.0.10",
riptio UMG8900 (SG) route is MGW1.
n
Scrip ADD M3RT: DENM="MGW1", LSNM="BSC1";

t
6. Delete the data in original network which is not needed now.
CAUTION
Confirm the data is not needed in the new network and the deletion should not affects the
services before delete the data.
Step 5 Modify the M2UA signalling to M3UA signalling between MSC Server and BSC.
n
Scrip ADD M3DE: DENM="BSC1", NI=NATB, DPC="A01", DET=SP;

t
n
Scrip ADD M3RT: DENM="BSC1", LSNM="BSC1";

t
----End

This section describes the data configuration on the UMG8900 in the singe-MGW VoIP
networking mode.

Procedure
Step 1 Configure data for IP bearer network on MGW.

riptio
n

n

t TIMEOUT=Aging;

t

riptio to 1, and Destination Entity Name is set to MSXd, indicating the
Scrip ADD M3DE: DEX=1, DEN="MSXd", DET=AS, NI=NATB, DPC=H'101, ADF=YES,

t LEX=1;

Desc Add an M3UA link set data record where Link Set Name is MSXd, Adjacent
riptio Destination Entity Index is 1, and Traffic Mode is Loadshare_Mode.
n
Scrip ADD M3LKS: LSX=1, LSN="MSXd", ADX=1, TM=LOADSHARE, WM=SGP;

t


n
Scrip ADD M3RT: RN=" MSXd", DEX=1, LSX=1;

t

Scrip ADD M3LNK: LNK=2, BT=SPF, BN=0, LKN="MSXd-B1", LIP1="192.168.10.10",

ADD M3LNK: LNK=3, BT=SPF, BN=1, LKN="MSXd-B2", LIP1="192.168.10.10",
different SPFs.
Step 3 Delete the data in original network which is not needed now.

CAUTION
Confirm the data is not needed in the new network and the deletion should not affects the services
before delete the data.
----End

This section describes the data configuration in the multi-MGW VoIP networking.
Original Network
Figure 4-4 shows the typical multi-MGW networking.
Figure 4-4 Multi-MGW networking
MSC Server
HLR
STP
MGW2
MSC MGW1 MGW3
TDM
BSC IP

l The ISUP is used between the MSC server and the MSC. The MSC server interworks MSC
by using the M3UA indirect connection mode. The MGW (SG) is used as the signaling
transfer point. The structure of the protocol stack at the MSC server side is ISUP/SCCP/
M3UA/SCTP/IP. The structure of the protocol stack at the MSC side is ISUP/SCCP/MTP3/
MTP2/MTP1.
l The BSSAP is used between the MSC server and the BSC. The MSC server interworks the
BSC by using the M2UA connection mode. The MGW (SG) is used as the signaling transfer
point. The structure of the protocol stack at the MSC server side is BSSAP/SCCP/MTP3/

MTP3/MTP2/MTP1.
l The H.248 protocol is used between the MSC server and the MGW. The MSC server
interworks the MGW by using the IP bearer networking. The structure of the protocol stack
at the MSC server side is H.248/SCTP/IP, which is the same as that at the MGW side.
l The MGWs interwork with each other by using the TDM networking.
Destination Network
Figure 4-5 shows the typical multi-MGW networking.
Figure 4-5 Multi-MGW VoIP networking
MSC Server
HLR
STP
CE1 CE2
MSC MGW1 MGW2 MGW3
TDM
IP
BSC
Seeing from the preceding figure, you can infer that:

l When the MSC server interworks with the NE of the access network, the SPC for the
national reserved network is used.
l The BICC and ISUP are used between the MSC server and the MSC.
The ISUP is used between the MSC server and the MSC. The MSC server interworks
with the MSC by using the M3UA indirect connection mode. The MGW (SG) is used
as the signaling transfer point. The structure of the protocol stack at the MSC server
side is ISUP/SCCP/M3UA/SCTP/IP. The structure of the protocol stack at the MSC
side is ISUP/SCCP/MTP3/MTP2/MTP1.
The BICC is used between the MSC server and the MSC. The MSC server interworks
with the MSC by using the M3UA direct connection mode. The structure of the protocol
stack at the MSC server side is BICC/M3UA/SCTP/IP, which is the same as that at the
MSC side.

MTP3/MTP2/MTP1.
l The H.248 protocol is used between the MSC server and the MGW. The MSC server
interworks with the MGW by using the IP bearer networking. The structure of the protocol
stack at the MSC server side is H.248/SCTP/IP, which is the same as that at the MGW side.
l The MGWs interwork with each other by using the IP bearer networking.
Evolution Method
The multi-MGW networking shown in Figure 4-4 is evolved to the multi-MGW VoIP
networking shown in Figure 4-5 in the following ways:
l Maintain the ISUP signaling between the MSC server and MSC2. Configure the data for
l Use the M3UA link instead of the M2UA link between the MSC server and the BSC.
l Use the IP bearer networking between the MGWs.
Interworking Data
In the proceeding evolution methods, you must plan and collect the following data:
l IP address
l SCTP port

This section introduces the flow of the data configuration for the multi-MGW VoIP networking.
This section describes interworking data to be negotiated for the single-MGW VoIP networking.
This section describes the data configuration on the MSOFTX3000 in the VoIP bearer network
using the multi-MGW networking.
This section describes the data configuration of the MSOFTX3000 in the multi-MGW VoIP
networking mode.

This section introduces the flow of the data configuration for the multi-MGW VoIP networking.
Figure 4-6 shows the data configuration flow for the multi-MGW VoIP networking.

Figure 4-6 Flow of the data configuration for the multi-MGW VoIP networking
Start
Configure the IP bearer

data for the MGW
Configure the BICC

MSC server and MSC 1
Configure the BICC

MSC server and MSC 2.

and the MGW

and the BSC
End

This section describes interworking data to be negotiated for the single-MGW VoIP networking.
As shown in Table 4-2, the interworking data to be planned and collected during the flow as
shown in Figure 4-2 is as follows:
NOTE

Table 4-2 Interworking data of single-MGW VoIP networking
Name MSC Server MGW MSC BSC
Node MSX MGW1 MSX02 BSC1

Name MGW2
MGW3
SPC National network: National network: National network: -

A1001 and A0002 MGW1:A1001 C0002
MGW2:-
MGW3:-
National reserved National reserved - National reserved

network: A01 network: network: A01
MGW1:101
MGW2:-
MGW3:-
IP Signaling plane: Signaling plane: Signaling plane: 192.168.40.10/25

Addre 192.168.0.10/255.2 MGW1:192.168.1 192.168.30.10/25 5.255.0.0
ss 55.0.0 0.10/255.255.0.0 5.255.0.0
MGW2:192.100.1 Bearer plane:
0.10/255.255.0.0 10.168.30.10/255.
255.0.0
MGW3:192.200.1
0.10/255.255.0.0
Bearer plane:
MGW1:10.168.10
.10/255.255.0.0
MGW2:10.100.10
.10/255.255.0.0
MGW3:10.200.10
.10/255.255.0.0
SCTP Module 141: 7002, SPF board 0: Module 141: 7004 -

Port 7004, and 7006 7002, 7006, - , - Module 142: 7005
Module 142: 7003, SPF board 1:
7005, and 7007 7003, 7007, - , -

This section describes the data configuration on the MSOFTX3000 in the VoIP bearer network
using the multi-MGW networking.
Procedure
Step 1 Modify MGW data.

1. Modify the MGW data record to support the IP bearer and codec.
Descr Modify the MGW data record to support the IP bearer and codec.
iptio
n
Scrip MOD MGW: MGWNAME="MGW1", TRNST=SCTP, BNCC=IP-1,

t SPCATTR=SUPCODECCFG-1,
MOD MGW: MGWNAME="MGW2", TRNST=SCTP, BNCC=IP-1,
SPCATTR=SUPCODECCFG-1,
MOD MGW: MGWNAME="MGW3", TRNST=SCTP, BNCC=IP-1,
SPCATTR=SUPCODECCFG-1,
Rema When the codec UMTS AMR/AMR2 is supported, set Support codec rate
rks configuration in Special attributes.
2. Modify a dual-homing server node data record.
Desc Modify a dual-homing server node data record where Internal MGW media
riptio type is IP, Internal MGW connection type is All MGW connection, and
n Internal MGW path select mode is Automatic.
Scrip MOD SRVNODE: SN=LOCAL, IMMT=IP, IMCT=ALLMGW, IMSM=AUTO;

t
Rem The preceding is the default configuration, which can be modified according
arks to the actual networking.
Step 2 Configure BICC signalling data for interworking with MSC.

riptio A0002.
n

t

riptio
n



t
n

t
n

t


n

t

arks network.

riptio
n

t BFSM=INVALID;

riptio
n

t
n

t ISUP=BICCIP_F;

n
Scrip ADD BICCTG: TGN="MSX02", MGWNAME="MGW1", SRN="MSX02_01", BTG=1,


riptio MSX02.
n

t
to the MGW.
Step 3 Modify the M2UA signalling to M3UA signalling of national reserved network between MSC
Server and MGW1.
Desc Add an M3UA local entity data record where Local entity name is MSXd,
Scrip ADD M3LE: LENM="MSXd", OPC="001", LET=AS;

t
the value of Route context on the MGW1. Here, it is null.

riptio is MGW1, indicating the UMG8900 (SG) directly connected to the
n MSOFTX3000.
Scrip ADD M3DE: DENM="MGW1", DPC="101", STPF=TRUE, DET=SG;

t

Desc Add an M3UA link set data record where Linkset name is BSC1, and Adjacent
riptio entity name is MGW1.
n
Scrip ADD M3LKS: LSNM="BSC1", ADNM="MGW1", TM=LOADSHARE, WM=ASP;

t

set to MGW1.
riptio WBSG module number to 141, Link name to BSC1_01, Local port
Server to Client, WBSG module number to 142, Link name to BSC1_02,
Scrip ADD M3LNK: MN=141, LNKNM="BSC1_01", LOCIP1="192.168.0.10",

ADD M3LNK: MN=142, LNKNM="BSC1_02", LOCIP1="192.168.0.10",
riptio UMG8900 (SG) route is MGW1.
n
Scrip ADD M3RT: DENM="MGW1", LSNM="BSC1";

t
6. Delete the data in original network which is not needed now.
CAUTION
Confirm the data is not needed in the new network and the deletion should not affects the
services before delete the data.
Step 4 Modify the M2UA signalling to M3UA signalling between MSC Server and BSC.

n
Scrip ADD M3DE: DENM="BSC1", NI=NATB, DPC="A01", DET=SP;

t
n
Scrip ADD M3RT: DENM="BSC1", LSNM="BSC1";

t
----End

This section describes the data configuration of the MSOFTX3000 in the multi-MGW VoIP
networking mode.
Procedure
Step 1 Configure data for IP bearer network on MGW.
Desc MGW1: Add an IP address data record where Interface IP address is

riptio 10.168.10.10.
n MGW2: Add an IP address data record where Interface IP address is
10.100.10.10.
MGW3: Add an IP address data record where Interface IP address is
10.200.10.10.

ADD IPADDR: BT=HRB, BN=0, IFT=GE, IFN=0, IPADDR="10.100.10.10",
MASK="255.255.0.0", FLAG=MASTER, INVLAN=NO, IFMPLS=NO;
ADD IPADDR: BT=HRB, BN=0, IFT=GE, IFN=0, IPADDR="10.200.10.10",
MASK="255.255.0.0", FLAG=MASTER, INVLAN=NO, IFMPLS=NO;
n

t TIMEOUT=Aging;

Step 2 Configure M3UA data for interworking between the MSC Server and MGW1.MSOFTX3000.

t

riptio to 1, and Destination Entity Name is set to MSXd, indicating the
Scrip ADD M3DE: DEX=1, DEN="MSXd", DET=AS, NI=NATB, DPC=H'101, ADF=YES,

t LEX=1;
Desc Add an M3UA link set data record where Link Set Name is MSXd, Adjacent
riptio Destination Entity Index is 1, and Traffic Mode is Loadshare_Mode.
n
Scrip ADD M3LKS: LSX=1, LSN="MSXd", ADX=1, TM=LOADSHARE, WM=SGP;

t


n
Scrip ADD M3RT: RN=" MSXd", DEX=1, LSX=1;

t

Scrip ADD M3LNK: LNK=2, BT=SPF, BN=0, LKN="MSXd-B1", LIP1="192.168.10.10",

ADD M3LNK: LNK=3, BT=SPF, BN=1, LKN="MSXd-B2", LIP1="192.168.10.10",
different SPFs.
Step 3 Delete the data in original network which is not needed now.
CAUTION
Confirm the data is not needed in the new network and the deletion should not affects the services
before delete the data.
----End
4.3 CMN Networking

This section describes the data configuration in the CMN VoIP networking mode.
Original Network
Figure 4-7 shows the typical TDM networking.

Figure 4-7 TDM networking
TMSC
MSC MSC
Server1 Server2
MGW1 MGW2 MGW3 MGW4
TDM
BSC IP MSC

l When the TMSC interworks with the NE of the core network, the SPC for the national
network is used.
l The ISUP is used between the TMSC server and MSC server 1. The TMSC interworks with
MSC server 1 by using the M3UA direct connection mode. The structure of the protocol
stack at the TMSC side is ISUP/SCCP/M3UA/SCTP/IP, which is the same as that at MSC
server 1 side.
l The ISUP is used between the TMSC server and MSC server 2. The TMSC interworks with
MSC server 2 by using the M3UA direct connection mode. The structure of the protocol
stack at the TMSC side is ISUP/SCCP/M3UA/SCTP/IP, which is the same as that at MSC
server 2 side.
Destination Network
Figure 4-8 shows the typical CMN networking.

Figure 4-8 CMN networking
CMN
MSC MSC
Server1 Server2
MGW1 MGW2 MGW3 MGW4
TDM
BSC IP MSC
l When the TMSC interworks with the NE of the core network, the SPC for the national
network is used.
l The BICC and ISUP are used between the MSC server 1 and the TMSC.
The ISUP is used between the TMSC server and MSC server 1. The TMSC interworks
with MSC server 1 by using the M3UA direct connection mode. The structure of the
protocol stack at the TMSC side is ISUP/SCCP/M3UA/SCTP/IP, which is the same as
that at MSC server 1 side.
The BICC is used between the TMSC server and MSC server 1. The TMSC interworks
protocol stack on the TMSC side is BICC/M3UA/SCTP/IP, which is the same as that
on the MSC server1 side.
l The BICC and ISUP are used between the MSC server 2 and the TMSC.
The ISUP is used between the TMSC server and MSC server 2. The TMSC interworks
protocol stack at the TMSC side is ISUP/SCCP/M3UA/SCTP/IP, which is the same as
that at MSC server 2 side.
The BICC is used between the TMSC server and MSC server 2. The TMSC interworks
protocol stack on the TMSC side is BICC/M3UA/SCTP/IP, which is the same as that
on the MSC server 2 side.

Evolution Method
When the TDM networking shown in Figure 4-7 is evolved to the CNM networking shown in
Figure 4-8, the data configuration for the CMN networking must coexist with the ISUP data
configuration. The data configuration is as follows:
l Maintain the ISUP signaling between the TMSC and MSC server 1. Configure the data for
l Maintain the ISUP signaling between the TMSC and MSC server 2. Configure the data for
Interworking Data
In the above evolution methods, you must plan and collect the following data:
l IP address
l SCTP port number
This section introduces the flow of the data configuration in the CMN VoIP networking mode.
This section describes interworking data to be negotiated in the CMN networking mode.
4.3.3 Data Configuration MSOFTX3000
This section describes the data configuration of the MSOFTX3000 in the CMN networking.

This section introduces the flow of the data configuration in the CMN VoIP networking mode.
Figure 4-9 shows the data configuration flow for the CMN networking mode.
Figure 4-9 Flow of the data configuration for the CMN networking mode
Start
Configure the BICC

TMSC and MSC server 1
Configure the BICC

TMSC and MSC server 2
End


This section describes interworking data to be negotiated in the CMN networking mode.
As shown in Table 4-3, the interworking data to be planned and collected during the flow as
shown in Figure 4-8 is as follows:
NOTE
Table 4-3 Interworking data in the CMN networking mode
Name TMSC Server MSC Server1 MSC Server2
Node MSX MSX01 MSX02

Name
SPC National network: National network: National network:

A1001 and A0002 C0001 C0002
IP 192.168.0.10/255.255.0. 192.168.20.10/255.255. 192.168.30.10/255.255.

Address 0 0.0 0.0
SCTP Module 141: 7000 and SPF board 0: 7000 Module 141: 7004
Port 7004 SPF board 1: 7001 Module 142: 7005
Module 142: 7001 and
7005
4.3.3 Data Configuration MSOFTX3000

This section describes the data configuration of the MSOFTX3000 in the CMN networking.
Procedure
riptio A0002.
n

t


riptio
n



t

n

t

n

t

riptio WBSG module number to 141, Link name to MSX01_01, Local port
Server to Client, WBSG module number to 142, Link name to
MSX01_02, Local port number to 7001, and Peer port number to 7001.
Scrip ADD M3LNK: MN=141, LNKNM="MSX01_01", LOCIP1="192.168.0.10",

LSNM="MSX02", QoS=TOS;
ADD M3LNK: MN=142, LNKNM="MSX01_02", LOCIP1="192.168.0.10",
LSNM="MSX02", QoS=TOS;
n

t

arks network.

riptio
n

t BFSM=INVALID;

riptio
n

t

n

t ISUP=BICCIP_F;
Rem Route selection name and should be same to original network.

arks
n
Scrip ADD BICCTG: TGN="MSX01", MGWNAME="CMN", SRN="MSX01_01", BTG=1,

RELRED=NO;
riptio MSX01.
n

t
to the MGW.

n

t

n

t

n

t

arks network.


riptio
n

t BFSM=INVALID;

riptio
n

t
n

t ISUP=BICCIP_F;
Rem Route selection name and should be same to original network.

arks
n
Scrip ADD BICCTG: TGN="MSX02", MGWNAME="CMN", SRN="MSX02_01", BTG=1,

RELRED=NO;
riptio MSX02.
n

t
to the MGW.

----End

VoIP User Manual 5 Routine Maintenance on the VoIP Networking
5 Routine Maintenance on the VoIP

Networking
About This Chapter
The routine maintenance on the VoIP networking indicates measurement of the VoIP-related
performance entities which provides references for the routine maintenance and network
optimization. If the value differences between measurement entities of two subsequent days are
greater, it indicates that the network may be faulty. In this case, you must use related methods
to locate and clear the faults, to ensure the normal running of the network.
5.1 MSOFTX3000 Performance Entity Measurement

During the routine maintenance, the MSOFTX3000 must measure the following VoIP-related
measurement units: BICC Capability Measurement, BICC SCTP Link Measurement, BICC
Office Direction Data Service Traffic, TKGRP Incoming Office Traffic, TKGRP Outgoing
Office Traffic, BICC CMN Traffic, M3UA Signaling Link Traffic, IP Flow Measure, and IP
QOS Measure between two end stations.
5.2 UMG8900 Performance Entity Measurement
During the routine maintenance, the UMG8900 must measure the following VoIP-related
measurement units: Statistics of Flow, IP QoS Measurement, IP Bearer User Plane Statistics, IP
Bearer Statistics, and RTP Packet Statistics of IP Bearer.

5 Routine Maintenance on the VoIP Networking VoIP User Manual
5.1 MSOFTX3000 Performance Entity Measurement

During the routine maintenance, the MSOFTX3000 must measure the following VoIP-related
measurement units: BICC Capability Measurement, BICC SCTP Link Measurement, BICC
Office Direction Data Service Traffic, TKGRP Incoming Office Traffic, TKGRP Outgoing
Office Traffic, BICC CMN Traffic, M3UA Signaling Link Traffic, IP Flow Measure, and IP
QOS Measure between two end stations.
For details, refer to the HUAWEI MSOFTX3000 Performance Metrics Reference.
BICC Capability Measurement

The performance of the BICC node is measured by measuring the number of failed BICC calls
due to various reasons and the number of incoming call attempts.
The performance measurement entities are as follows:
Incoming call attempt, Device congestion, No free circuit, Address incomplete, Temporary
reason, Called number is null, Called subscriber busy, Called termination out of service, and
Other reason.
BICC SCTP Link Measurement

The traffic over the BICC SCTP link is measured by measuring entities such as the rate, buffer
seizure ratio, time delay, the number of unavailable BICC SCTP links, the number of times of
congestion, the number of messages, the number of times of retransmission, and the number of
buffer overflows.
Number of disconnected, Duration of disconnected, Number of congestion, Duration of

congestion, Number of message send, Bytes of message send, Number of Message Received,
Bytes of Message Received, Rate of Message send, Rate of Message Received, Bit Rate of
Message send, Bit Rate of Message Received, Transmission Load, Reception Load, Retransfer
Count, Retransfer Ratio, Sending Buffer Overflow Count, Receiving Buffer Overflow Count,
Sending Buffer Seizure Ratio, Sending Buffer Seizure Ratio Aux, Receiving Buffer Seizure
Ratio, Receiving Buffer Seizure Ratio Aux, Average Acknowledgement Delay, and Average
Deviation.
The involved formulas are as follows:
l Packets sent per second = number of messages sent / measurement period (second)
l Packets received per second = Number of messages received / Measurement period
(second)
l Bits sent per second = Bytes of messages sent * 8 / Measurement period (second)
l Bits received per second = Bytes of messages received * 8 / Measurement period (second)
l Load of data packets sent = Number of data packets sent / (Basic load of the links *
Measurement period in seconds)
l Load of data packets received = Number of data packets received / (Basic load of the links
* Measurement period in seconds)
l Retransfer ratio = Retransfer count / Number of message send

BICC Office Direction Data Service Traffic

The traffic of the BICC office direction data service is measured by measuring entities such as
the number of the peer codec modification requests and the number of the peer successful codec
modifications.
CODEC MODIFY TIMES and CODEC MODIFY SUCCESSTIMES.
TKGRP Incoming Office Traffic

The trunk group incoming bearer traffic is measured by measuring entities such as the entities
such as different types of traffic, and the number of seizure times, successful calls, links in
different states, circuit contention times, and continuity check times.
Seizure Times, Call Connected Times, Answer Times, Local Office Fail Times, CONGESTION
DURATION, Called Busy Times, Abandon After Ring Times, Ringed No Answer Times,
Installed Circuit Num, Avail Circuit Num, Blocked Circuit Num, Installed Bothway Circuit
Num, Avail Bothway Circuit Num, Blocked Bi-directional Circuit Num, InTK Avail Ratio,
Connected Ratio, Answer Ratio, Blocked Circuit Ratio, Blocked Bi-directional Circuit Ratio,
Bidirection TK Seizure Traffic, Seizure Traffic, Connected Traffic, Answer Traffic, Average
Seizure Traffic per Line, Average Connected Traffic Per Line, Average Answer Traffic Per Line,
Invalid Address, Unallocated Number Times, Congestion Times,
MS_H_TKGRP_COMBIN_IN_EMLPP_RAP, CONTINUITY CHECK TIMES, and
CONTINUITY CHECK FAILURE TIMES.
TKGRP Outgoing Office Traffic

The trunk group outgoing bearer traffic is measured by measuring the entities such as different
types of traffic, and the number of seizure times, successful calls, links in different states, circuit
contention times, and continuity check times.
Bid Times, Seizure Times, Call Connected Times, Answer Times, Overflow Times, Trunk
Circuit Mismatched Times, Bearer Capability Mismatched Times, Dual Seizure Times, TK
Retry Times, Peer End Congestion Times, CONGESTION DURATION, Called Busy Times,
Called Toll Busy Times, Called Local Busy Times, Abandon After Ring Times, Ringed No
Answer Times, Installed Circuit Num, Avail Circuit Num, Blocked Circuit Num, Installed
Bothway Circuit Num, Avail Bothway Circuit Num, Blocked Bi-directional Circuit Num,
OutTK Ratio, Seizure Ratio, Connected Ratio, Answer Ratio, Blocked Circuit Ratio, Blocked
Bi-directional Circuit Ratio, Bothway Circuit Seizure Traffic, Seizure Traffic, Connected
Traffic, Answer Traffic, Trunk Outgoing Call Subscriber Line Faulty, Trunk Outgoing Call Peer
Office No Response, Outgoing Call Abandon Before Ring Times, Average Seizure Traffic per
Line, Average Connected Traffic Per Line, Average Answer Traffic Per Line, Invalid Address,
Unallocated Number Times, Duration without Free Circuits,
MS_H_TKGRP_COMBIN_OUT_EMLPP_RAP, CONTINUITY CHECK TIMES,
CONTINUITY CHECK FAILURE TIMES, MS_H_TKGRP_AIP_OUT_BIDS,
MS_H_TKGRP_AIP_OUT_SEIZ, MS_H_TKGRP_AIP_OUT_CALLED_CONNECTED,
MS_H_TKGRP_AIP_OUT_ANS, MS_H_TKGRP_AIP_OUT_OVERFLOW,
MS_H_TKGRP_AIP_OUT_SEIZ_USAGE,
MS_H_TKGRP_AIP_OUT_CONNECTED_USAGE,

MS_H_TKGRP_AIP_OUT_ANS_USAGE, and
MS_H_TKGRP_AIP_OUT_AVER_SEIZ_PER_LINE.
BICC CMN Traffic

The BICC CMN traffic is measured by measuring different types of traffic and the number of
BICC CMN incoming call seizure times and BICC CMN outgoing call seizure times.
BICC CMN INCOMING SEIZURE TIMES, BICC CMN INCOMING CALL CONNECTED
TIMES, BICC CMN INCOMING ANSWER TIMES, BICC CMN INCOMING CONNECTED
RATIO, BICC CMN INCOMING ANSWER RATIO, BICC CMN INCOMING SEIZURE
TRAFFIC, BICC CMN INCOMING CONNECTED TRAFFIC, BICC CMN INCOMING
ANSWER TRAFFIC, BICC CMN INCOMING ABANDON BEFORE RING TIMES, BICC
CMN INCOMING ABANDON AFTER RING TIMES, BICC CMN INCOMING RINGED
NO ANSWER TIMES, BICC CMN INCOMING CALLED DETERMINDED BUSY TIMES,
BICC CMN INCOMING CALLED BUSY TIMES, BICC CMN INCOMING INVALID
ADDRESS, BICC CMN INCOMING ABSENT SUB TIMES, BICC CMN INCOMING
SERVICES LIMITED TIMES, BICC CMN INCOMING SEND SPECIAL TONE TIMES,
BICC CMN INCOMING CONGESTION TIMES, BICC CMN INCOMING PAGE NO
RESPONSE, BICC CMN INCOMING INTERWORK FAIL TIMES, BICC CMN INCOMING
CALLED PARTY NO RESPONSE, BICC CMN INCOMING UMG UNAVAILABLE TIMES,
BICC CMN INCOMING ADD CIRCUIT BUSY TIMES, BICC CMN INCOMING
UNALLOC NUMBER TIMES, BICC CMN INCOMING OTHER FAIL TIMES, BICC CMN
OUTGOING BID TIMES, BICC CMN OUTGOING SEIZURE TIMES, BICC CMN
OUTGOING CALL CONNECTED TIMES, BICC CMN OUTGOING ANSWER TIMES,
BICC CMN OUTGOING CONNECTED RATIO, BICC CMN OUTGOING ANSWER
RATIO, BICC CMN OUTGOING DUAL SEIZURE TIMES MASTER, BICC CMN
OUTGOING DUAL SEIZURE TIMES SLAVE, BICC CMN OUTGOING TK RETRY
TIMES, BICC CMN OUTGOING SEIZURE TRAFFIC, BICC CMN OUTGOING
CONNECTED TRAFFIC, BICC CMN OUTGOING ANSWER TRAFFIC, BICC CMN
OUTGOING ABANDON BEFORE RING TIMES, BICC CMN OUTGOING ABANDON
AFTER RING TIMES, BICC CMN OUTGOING RINGED NO ANSWER TIMES, BICC CMN
OUTGOING CALLED DETERMINDED BUSY TIMES, BICC CMN OUTGOING CALLED
BUSY TIMES, BICC CMN OUTGOING INVALID ADDRESS, BICC CMN OUTGOING
ABSENT SUB TIMES, BICC CMN OUTGOING SERVICES LIMITED TIMES, BICC CMN
OUTGOING SEND SPECIAL TONE TIMES, BICC CMN OUTGOING CONGESTION
TIMES, BICC CMN OUTGOING PAGE NO RESPONSE, BICC CMN OUTGOING
INTERWORK FAIL TIMES, BICC CMN OUTGOING CALLED PARTY NO RESPONSE,
BICC CMN OUTGOING UMG UNAVAILABLE TIMES, BICC CMN OUTGOING ADD
CIRCUIT BUSY TIMES, BICC CMN OUTGOING OTHER FAIL TIMES, BICC CMN
REROOT TIMES, and BICC CMN REROOT FAIL TIMES.
M3UA Signaling Link Traffic

The traffic over the M3UA signaling link is measured by measuring entities such as the duration
of M3UA signaling link unavailability, the rate, time delay, and the number of M3UA signaling
link unavailability times, congestion times, messages, buffer overflows, and association
unavailability times.
Unavailable Times, Unavailable Duration, Congestion Times, Congestion Duration, Number of
Control Message Transmited, Number of Control Message Received, Bytes of Control Message

Transmited, Bytes of Control Message Received, Number of Data Message Transmited, Number
of Data Message Received, Bytes of Data Message Transmited, Bytes of Data Message
Received, Number of SCCP Message Transmited, Number of SCCP Message Received, Bytes
of SCCP Message Transmited OF ISUP MESSAGE, Bytes of SCCP Message Received,
Number of TUP Message Transmited, Number of TUP Message Received, Bytes of TUP
Message Transmited, Bytes of TUP Message Received, Number of ISUP Message Transmited,
Number of ISUP Message Received, Bytes of ISUP Message Transmited OF ISUP MESSAGE,
Bytes of ISUP Message Received, Number of BICC Message Transmited, Number of BICC
Message Received, Bytes of BICC Message Transmited, Bytes of BICC Message Received,
Number of H248 Message Transmited, Number of H248 Message Received, Bytes of H248
Message Transmited, Bytes of H248 Message Received, Number of Message Transmited,
Number of Message Received, Bytes of Message Transmited, Bytes of Message Received,
Packests Send per Second, Packests Receive per Second, Bits Send per Second, Bits Receive
per Second, Maxium Transmit Bits Per Second, Maxium Receive Bits Per Second, Send Load,
Receive Load, Retransfer Count, Retransfer Ratio, Sending Buffer Overflow Count, Receiving
Buffer Overflow Count, Average Acknowledgement Delay, Average Deviation, Times of
Association Unavailable, and Duration of Association Unavailable.
IP Flow Measure
The traffic through the IP port is measured by measuring the number of the packets sent and
received through the IP port, successfully sent and received packets, limited ICMP packets.
Packets of the sent, Packets of the received, Bytes of the successfully sent, Bytes of the
successfully received, Speed of the sent, Speed of the received, Bits of the sent, Bits of the
received, Rates of band with sending, Rates of band with receiving, Packets of dropped, Packets
of invalid IP address, Packets of access denied, Packets of limited ICMP, Packets of unknow
IP, Packets of failure in forwarding, Packets of failure in sending, Maxinum Bits Speed of sent,
Maxinum Bits Speed of received, M3UA Packets of sent, M3UA Packets of received, M3UA
Bytes of sent, M3UA Bytes of received, Rate of bandwidth with sending M3UA, Rate of
bandwidth with receiving M3UA, M2UA Packets of sent, M2UA Packets of received, M2UA
Bytes of sent, M2UA Bytes of received, Rate of bandwidth with sending M2UA, Rate of
bandwidth with receiving M2UA, H248 SCTP Packets of sent, H248 SCTP Packets of received,
H248 SCTP Bytes of sent, H248 SCTP Bytes of received, Rate of bandwidth with sending H248
SCTP, Rate of bandwidth with receiving H248 SCTP, BICC SCTP Packets of sent, BICC SCTP
Packets of received, BICC SCTP Bytes of sent, BICC SCTP Bytes of received, Rate of
bandwidth with sending BICC SCTP, and Rate of bandwidth with receiving BICC SCTP.
IP QOS Measure Between Two End Station

The QoS of IP links is measured by measuring entities such as the band occupation, packet loss
rate, time delay, rate, and the number of messages.
Rate of band with sending, Rate of band with receiving, Rate of lost packet, Average Round
Robin Trip Time, Packets of sent, Packets of received, Speed of the sent, Speed of the received,
Bit Speed of the sent, Bit Speed of the received, Maximum Bit Speed of the sent, Maximum Bit
Speed of the received, M3UA Packets of sent, M3UA Packets of received, M3UA Bytes of sent,
M3UA Bytes of received, Rate of bandwidth with sending M3UA, Rate of bandwidth with
receiving M3UA, M2UA Packets of sent, M2UA Packets of received, M2UA Bytes of sent,
M2UA Bytes of received, Rate of bandwidth with sending M2UA, Rate of bandwidth with
receiving M2UA, H248 SCTP Packets of sent, H248 SCTP Packets of received, H248 SCTP

Bytes of sent, H248 SCTP Bytes of received, Rate of bandwidth with sending H248 SCTP, Rate
of bandwidth with receiving H248 SCTP, BICC SCTP Packets of sent, BICC SCTP Packets of
received, BICC SCTP Bytes of sent, BICC SCTP Bytes of received, Rate of bandwidth with
sending BICC SCTP, and Rate of bandwidth with receiving BICC SCTP.
5.2 UMG8900 Performance Entity Measurement

During the routine maintenance, the UMG8900 must measure the following VoIP-related
measurement units: Statistics of Flow, IP QoS Measurement, IP Bearer User Plane Statistics, IP
Bearer Statistics, and RTP Packet Statistics of IP Bearer.
For details, refer to the HUAWEI UMG8900 Universal Media Gateway Performance Metrics
Reference.
Statistics of Flow
This task is used to measure bandwidth of the HRB. Through this task, you can learn about the
current bandwidth utilization of interfaces on the HRB board. Through long-term observation
on measurement results of the flow statistics, you can learn about the trend of the service flow
change, and predict whether the capacity of the current network equipment needs to be expanded
for services.
Interface Received Flow, Interface Sent Flow, Use Rate of Interface Sent Bandwidth, and Use
Rate of Interface Received Bandwidth
l Use Rate of Interface Sent Bandwidth = Bandwidth in the sending direction of a physical
interface at the end of a measurement period/Line speed of the interface
l Use Rate of Interface Received Bandwidth = Bandwidth in the receiving direction of a
physical interface at the end of a measurement period / Line speed of the interface
IP QoS Measurement
The measurement task indicates the reliability of the IP bearer transmission and the quality of
data transmission. The task helps to learn about the running status of the network and provides
important reference data for maintenance, management, and planning of the network.
Number of Received RTP Packets, Number of Sent RTP Packets, Number of Received RTP
Bytes, Number of Sent RTP Bytes, Number of Lost Receive RTP Packets, Local Max Lost Rate
of Packet, Local IP of Max Lost Rate of Packet, Remote IP of Max Lost Rate of Packet, Local
Min Lost Rate of Packet, Local IP of Min Lost Rate of Packet, Remote IP of Min Lost Rate of
Packet, Local Max Delay Jitter of Packet, Local IP of Max Delay Jitter of Packet, Remote IP of
Max Delay Jitter of Packet, Local Min Delay Jitter of Packet, Local IP of Min Delay Jitter of
Packet, Remote IP of Min Delay Jitter of Packet, Local Max Circle Delay of Packet, Local IP
of Max Circle Delay of Packet, Remote IP of Max Circle Delay of Packet, Local Min Circle
Delay of Packet, Local IP of Min Circle Delay of Packet, Remote IP of Min Circle Delay of
Packet, Received RTP Traffic Bandwidth, Sent RTP Traffic Bandwidth, Duration of Packet Lost
Rate Exceeded Thrd, Duration of Packet Delay Jitter Exceeded Thrd, Duration of Packet Circle
Delay Exceeded Thrd, Local Average Lost Rate of Packet, Local Average Delay Jitter of Packet,
and Local Average Circle Delay of Packet


l Received RTP Traffic Bandwidth = Total number of received RTP bytes / Measure period
l Received RTP Traffic Bandwidth = Total number of sent RTP bytes / Measure period
IP Bearer User Plane Statistics

The task reflects the reliability of the Nb bearer transmission and the quality of data transmission.
This task helps to learn about the running status of the network and provides important reference
data for maintenance, management, and planning of the network.
IP Bearer User Plane Send Initialize Request Times, IP Bearer User Plane Receive Initialize
Confirm Times, IP Bearer User Plane Receive Initialize Request Times, and IP Bearer User
Plane Send Initialize Confirm Times
IP Bearer Statistics
The task is to measure the success ratio of bearer establishment. This task helps to learn about
the running status of the network, and provides important reference data for maintenance,
management, and planning of the network.
IP Send Establish Request Times, IP Receive Establish Confirm Times, IP Receive Establish
Request Times, and IP Send Establish Confirm Times.
RTP Packet Statistics of IP Bearer

The task is used to measure the average traffic of the bearer transmission on the Nb interface. It
reflects the load of bearer transmission on the Nb interface. This task helps to learn about the
running status of the network and provides important reference data for maintenance,
management, and planning of the network.
Number of Received RTP Packets, Number of Sent RTP Packets, Number of Received Messages
on Physical Layer, Number of Sent Messages on Physical Layer, Number of Lost Packets,
Number of Sent Packets, Average Delay, Max Delay, Average Delay Jitter, Max Delay Jitter,
and Number of Lost Bytes
l Average Delay = Total time length of round trip delays / Number of sampling times
l Average Delay Jitter = Total jitters / Number of sampling times

VoIP User Manual 6 VoIP Troubleshooting
6 VoIP Troubleshooting
About This Chapter
The procedure of troubleshooting VoIP faults includes collecting fault information, checking
the fault scope and type, locating the fault cause, removing the fault, checking equipment and
service operation, and contacting Huawei's technical support.
Figure 6-1 shows the troubleshooting procedure.

6 VoIP Troubleshooting VoIP User Manual
Figure 6-1 Troubleshooting procedure
Start
Collect fault information
Check the fault scope

and type
Locate the fault cause
Remove the fault
Check equipment No Contact Huawei's

and service operation technical support
Yes
End
6.1 Collecting VoIP Fault Information

Fault information collecting helps maintenance engineers to obtain fault information and locate
fault causes.
6.2 Analyzing VoIP Fault
When you obtain the fault information, you must determine the scope and category of the fault
by analyzing the fault symptom.
6.3 Troubleshooting VoIP Faults
During troubleshooting, you can exclude the impossible factors by analyzing and comparing the
possible causes by using certain methods. Then, you can find out the cause of the fault.
6.4 Troubleshooting VoIP Faults Based on Alarms
When certain fault occurs, the related alarm is generated at the alarm console. You can handle
the fault based on the prompt information provided by the alarm console and the help information
about the fault.
6.5 Troubleshooting VoIP Faults Based on Performance Measurement
If abrupt change is found in the performance measurement results, the network operation may
not be normal. In this case, you must locate and remove faults based on the performance
measurement.

6.6 Troubleshooting VoIP Faults Related to Services

VoIP faults related to services include call setup failure, one-way audio, no audio, and speech
noise.

6.1 Collecting VoIP Fault Information

Fault information collecting helps maintenance engineers to obtain fault information and locate
fault causes.
Collect the following fault information before handling faults:
l Symptom
l When and where the fault occurs and the frequency of the fault
l Range and impact of the fault
l The operating status of the equipment before the fault occurs
l Operation on the equipment and operation results before the fault occurs
l Measures taken after the fault occurs and the result
l Alarms or related alarms reported when the fault occurs
l Board LED status when the fault occurs
l Messages on which the interface/signaling trace is performed when the fault occurs
You can collect the required fault information by following methods:
l Ask the staff of the customer service center receiving the fault report for the detailed
symptom, time, location and frequency of the fault.
l Ask the operation and maintenance staff for the daily operation status of the equipment,
symptom, operations before the fault occurs, and measures taken after the fault occurs and
the result.
l Check the board LEDs and obtain the software and hardware operation status of the
equipment in the alarm management system of the LMT.
l Obtain the range and impact of the fault through service tests, performance measurement,
interfaces and signaling trace.
Note the following when collecting the fault information:
l Keep in mind to collect related fault information. When a fault occurs, especially a critical
fault, you must obtain the comprehensive information about the fault before performing
another operation.
l Maintain good relationship with the maintenance staff of other offices or related
departments. This can help information communication and technical support.
6.2 Analyzing VoIP Fault

When you obtain the fault information, you must determine the scope and category of the fault
by analyzing the fault symptom.
According to the equipment features, the faults can be classified into the following categories:
The faults are classified according to services when the IP network bears the services.
l Basic telecommunication service fault

l IN service fault

l Supplementary service fault
The faults are classified according to function subsystems:
l Hardware system fault

l BICC trunk system fault
l Voice fault
6.3 Troubleshooting VoIP Faults

During troubleshooting, you can exclude the impossible factors by analyzing and comparing the
possible causes by using certain methods. Then, you can find out the cause of the fault.
Context
Comparing with the traditional TDM network, the VoIP network uses the IP bearer network to
transmit the voice service data. The transmission quality of the IP bearer network is closely
related to the quality of the voice services. Therefore, you must first determine whether the VoIP
fault is caused by the fault of the IP bearer network or the fault of the softswitch of the core
network. To locate the VoIP fault, proceed as follows:
Procedure
Step 1 Check whether the boards and interfaces are faulty, whether the port negotiation is normal, and
whether the ARP detection is normal by using the board querying command on the
MSOFTX3000 and UMG8900. In this way, you can find out whether the core network
equipment works normally.
l If it works normally, go to Step 2
l If it does not work normally, go to Step 6
Step 2 Locate the fault in cooperation with the maintenance staff of the IP bearer network.
Run the ping command on the core network equipment to check whether the fault occurs at the
access side or in the backbone network.
l Trace IP messages on the client of the MSOFTX3000 and ping the MGW and the Mc interface
of the remote MGW.
l Trace the IP messages on the client of the UMG8900.
If a fault is found in the signaling plane, ping the gateway of the signaling plane and the
Mc interface of the remote MSC.
If a fault is found in the bearer plane, ping the gateway of the bearer plane and the Nb
interface of the remote MGW.
When the following operation results are displayed:
l If the local gateway cannot be pinged or the packet loss occurs, the fault is caused by the
problem at the accessing side of the local IP bearer network. Go to Step 4
l If the local gateway can be pinged and the packet loss is not found, but the remote core
network equipment cannot be pinged or the packet loss is found, go to Step 3
Step 3 Log into the remote core network equipment and ping the remote gateway.

l If the remote gateway cannot be pinged or the packet loss is found, the fault is caused by the
problem at the accessing side of the remote IP bearer network. Then go to Step 4
l If the remote gateway can be pinged and the packet loss is not found, the fault is caused by
the problem of the backbone IP bearer network. Then go to Step 5
Step 4 On the MSOFTX3000, ping the virtual IP address of the router (IP1), the VLAN IP address of
the active router (IP2) and the VLAN IP address of the standby router (IP3).
l If IP1 and IP2 cannot be pinged but IP3 can be pinged, the fault is caused by the level 3
logical failure of the router.
l If only IP2 can be pinged, the fault is caused by the level 2 logical failure of the router.
Step 5 Contact the maintenance staff of the IP bearer network to handle the fault.
Step 6 Locate the fault of the core network equipment.
l You can locate service faults through signaling trace, dialing tests, service comparison, and
performance measurement.
l You can locate function faults through tests, loopback tests, service comparison, hardware
replacement, and performance measurement.
For details, see the HUAWEI MSOFTX3000 Troubleshooting Manual.
----End
6.4 Troubleshooting VoIP Faults Based on Alarms

When certain fault occurs, the related alarm is generated at the alarm console. You can handle
the fault based on the prompt information provided by the alarm console and the help information
about the fault.
The alarms related to VoIP faults are as follows:
l ALM-1453 H.248 SCTP Link Failure
l ALM-1455 H.248 SCTP Link Congestion
l ALM-1457 H.248 SCTP Transmission Quality Alarm
l ALM-1522 MGW Out of Service
l ALM-1524 H.248 UDP Link Failure
l ALM-1542 MGW Notifies Itself Overload
l ALM-1546 H.248 SCTP Link Overload
l ALM-1809 M3UA Link Congestion
l ALM-1811 M3UA Link Out of Service
l ALM-1815 M3UA Route Unavailable
l ALM-1817 M3UA Destination Entity Is Inaccessible
l ALM-2001 Board Fault
l ALM-2039 FE Port Fault
l ALM-2273 ARP Detects Abnormality
l ALM-2607 BICC CIC Group Local End Block
l ALM-2611 BICC Group Blocks for Maintain on Other Side

6.5 Troubleshooting VoIP Faults Based on Performance

Measurement
If abrupt change is found in the performance measurement results, the network operation may
not be normal. In this case, you must locate and remove faults based on the performance
measurement.
Performance of the VoIP network is measured in terms of two types of indexes: stream indexes
and traffic indexes.
The following performance indexes of the traffic often change abruptly: IP port traffic, Mc
interface H.248 traffic, and M3UA signaling traffic.
The following performance index of performance measurement often changes abruptly: BICC
trunk performance measurement.
Abrupt Change in Traffic Statistics

The traffic statistics consists of sent and received packets on the IP port, sending and receiving
rate statistics, the statistics of sent and received packets from the MSC server to the MGW, and
the traffic statistics of M3UA signaling.
The abnormal cases in statistics include: abrupt traffic increase, abrupt traffic decrease and zero
traffic change.
l Abrupt traffic increase often occurs when the traffic is increased abnormally. This fault is
caused by repeated call attempts due to subscriber communication failure. If the reserved
bandwidth is small in this case, the continuous call loss occurs, which can cause repeated
call attempts. As a result, the traffic at the IP port maintains at a high level.
l Abrupt traffic decrease often occurs when the IP bearer network is damaged but the
communication is not interrupted completely. When this fault occurs, the bearer network
discards some IP packets being forwarded. The MSC cannot receive the responses of some
messages from the peer NE, so subsequent IP packets cannot be sent through the bearer
network. As a result, the link congestion occurs. The intermittent alarm of the M3UA link
is often reported in this case.
l Zero traffic change occurs when the IP bearer network is interrupted completely or the IP
interface board is faulty. This fault can result in service interruption.
When abrupt change in traffic statistics result is found, you must compare the current traffic
statistics result with that in the same time segment of the previous day. You can categorize the
fault based on the normality of the traffic statistics.
l For abrupt traffic increase, you must locate the cause of repeated call attempts and remove
this fault. You must bring down the call traffic through flow control or other methods to
avoid the adverse impact of system overload on communication bandwidths.
l For abrupt traffic decrease, you must locate the bearer network damage through the ping
command. After you locate the bearer network damage, you must switch over the IP bearer
plane to restore the services immediately. Then you can locate the faulty node of the bearer
network and remove the fault.

l For zero traffic change, you must remove the fault in the IP interface board of the MSC
server. Then you must switch the IP bearer plane and restore the services immediately. You
can locate the faulty node of the bearer network and remove the fault.
Abrupt Change in Performance Measurement

When the M3UA protocol is used to bear the BICC signaling message, you must measure the
traffic in the BICC trunk office direction in terms of the following indexes: number of call
attempts, call completion rate, and idle and faulty CICs.
The abnormal cases in traffic measurement include: call attempt decrease, call completion rate
decrease and faulty CIC ratio increase.
When abrupt change in traffic measurement result is found, you must compare the current traffic
measurement result with that in the same time segment of the previous day. You can categorize
the fault based on the normality of the traffic measurement.
l For call attempt decrease, you must check whether the increase of blocked CICs is caused
by the fault in the office direction.
l For call completion rate decrease, you must check whether the failure processing function
and CIC contention for BICC signaling are enabled. If not, enable the two functions first.
If call completion rate decrease persists after the two functions are enabled, you must check
the packet loss ratio of the bearer network between the two sits by using the ping command.
Then you can determine whether the bandwidth damage of the bearer network occurs or
the transmission quality of the bearer network is decreased. If the bearer network works
properly, you must check whether the fault is caused by the data modification on the peer
office.
l For faulty CIC ratio increase, you must check whether the peer M3UA entity is reachable.
Then you can determine whether the CIC fault is caused is caused by the problem that the
M3UA route is unreachable. If the M3UA route works properly, you must check whether
the fault is caused by the data modification on the peer office.
When the traffic on this office direction is normal in a short period, redirect call data over the
abnormal route to the standby route by modifying the data on the local office.
6.6 Troubleshooting VoIP Faults Related to Services

VoIP faults related to services include call setup failure, one-way audio, no audio, and speech
noise.
6.6.1 Call Setup Failure

The call setup failure in the VoIP service is related to the following factors: signaling links,
number analysis, trunk routes, voice codec, and bandwidth at the bearer plane. This section
describes the basic method to handle call setup failure.
6.6.2 One-Way Audio and No Audio Faults
When one-way audio occurs, only the caller or the callee hears the peer's voice. When one-way
audio occurs, neither the caller nor the callee can hear the peer's voice. This section describes
the basic method used to remove one-way audio and no audio faults.
6.6.3 Speech Noise
Speech noise refers to the sound other than the voice during the conversation. This section
describes the basic method to remove the speech noise fault.

6.6.1 Call Setup Failure

The call setup failure in the VoIP service is related to the following factors: signaling links,
number analysis, trunk routes, voice codec, and bandwidth at the bearer plane. This section
describes the basic method to handle call setup failure.
Procedure
Step 1 Check whether the BICC signaling link is normal.
1. Check whether the bearer type of the BICC signaling in the office direction table is the
same as the bearer type in use.
l If the BICC signaling adopts the MTP, MTP3B or M3UA bearer, check whether the
destination signaling point code in the office direction table is correct.
If it is correct, go to Step 1.7.
If it is not correct, go to Step 1.2.
l If the BICC signaling adopts the MTP, MTP3B, M3UA, or SCTP bearer, go to Step
1.3.
2. Modify the destination signaling point code. Check whether the call setup is successful.
l If the call setup is successful, go to the end.
l If the call setup fails, go to Step 1.7.
3. Trace the SCTP link messages. Check whether the SCTP link setup is successful.
l If it is successful, go to Step 1.6.
l If it fails, go to Step 1.4.
4. Check the data configuration on both sides of the SCTP link. Check whether the IP address,
port number, and client/server property are correct.
l If the data configuration is correct, go to Step 1.6.
l If the data configuration is incorrect, go to Step 1.5.
5. Modify the data configuration on both sides of the SCTP link. Check whether the call setup
is successful.
6. Deactivate and activate the SCTP link by using the DEA BSCTP and ACT BSCTP
command. Check whether the call setup is successful.
7. Check whether the physical connection is faulty.
l If the connection is normal, go to Step 2.
l If the connection is faulty, go to Step 1.8.
8. Handle the physical link fault. Check whether the call setup is successful.
l If the call setup fails, go to Step 2.
Step 2 Trace messages A interface. Check whether the IAM is sent.

l If the IAM is sent, go to Step 3.

l If no IAM is sent, go to Step 20.
Step 3 Check messages at the A interface. Check whether A interface assignment is successful.
l If it is successful, go to Step 4.
l If it fails, contact the maintenance personnel to locate the fault.
Step 4 Check the number attribute of the call prefix of the called number by using LST CNACLD.
l If the called number is a PSTN number, go to Step 5.
l If the called number is an MSISDN, go to Step 6.
Step 5 Check whether the configuration of the route analysis index of the DN set is correct.
l If it is correct, go to Step 8.
l If it is incorrect, go to Step 7.
Step 6 Obtain the MSRN of the callee by tracing the MAP messages. Check whether the prefix of the
MSRN is correct by using LST CNACLD.
Step 7 Modify the configuration data. Check whether the call setup is successful.
Step 8 Check whether the data in the route analysis table is correct by using LST RTANA.
Step 10 Check the configuration of the route, sub-route and gateway of the BICC trunk group in the
trunk group table.
Step 12 Trace the messages at the Mc interface. Check whether the MSOFTX3000 sends the outgoing
ADD.req message to the UMG8900.
l If the ADD.req message is not sent, go to Step 13.
l If the ADD.req message is sent, go to Step 15.
Step 13 Check the functions supported by the UMG8900 by using LST MGW. Check whether the codec
type of the call is correct.


Step 15 Check the messages at the Mc interface. Check whether the ADD.rsp message fails to be returned
after the ADD.req message is sent.
l If the ADD.rsp message fails to be returned, go to Step 16.
l If the ADD.rsp message is returned, go to Step 27.
Step 16 Check whether the value of the codec recorded in the system log is one of the following values
by using LST SYSLOG: G771A(1), G711U(2), AMR(3), AMR2(4), AMR_WB(15), G7231
(5), G726(7), G726_40(35), G726_32(36), G726_24(37), G726_16(38), G729(6), GSM_HR
(12), GSM_FR(13), GSM_EFR(14), T38(24), EVRC_HW(30), EVRC(20), EVRC0(51),
Q13K_HW(32), Q13K(22), Q8K(21), Q8K_HW(31), ISDN(17), MPEG4_VIDEO(8), H263
(10), H324(19), or the five common codecs (55-59).
l If not, go to Step 17.
l If yes, go to Step 18.
Step 17 Delete the codec type other than the ones listed previously. Check whether the call setup is
successful.
Step 18 Check whether the configuration of the trunk resources of the UMG8900 is correct.
Step 20 Check the messages at the A interface and locate the cause value for the REL message.
l If the cause value is "Network not work normally", go to Step 21.
l If the cause value is "No route available", go to Step 23.
l If the cause value is "Network not work normally", go to the process of no paging response
from the callee.
l If the cause value is "Network not work normally", go to Step 13.
Step 21 Check whether the value of the codec recorded in the system log is one of the following values
by using LST SYSLOG: G771A(1), G711U(2), AMR(3), AMR2(4), AMR_WB(15), G7231
(5), G726(7), G726_40(35), G726_32(36), G726_24(37), G726_16(38), G729(6), GSM_HR
(12), GSM_FR(13), GSM_EFR(14), T38(24), EVRC_HW(30), EVRC(20), EVRC0(51),
Q13K_HW(32), Q13K(22), Q8K(21), Q8K_HW(31), ISDN(17), MPEG4_VIDEO(8), H263
(10), H324(19), or the five common codecs (55-59).

Step 22 Delete the codec type other than the ones listed previously. Check whether the call setup is
successful.
Step 23 Check the number attribute of the call prefix of the called number by using LST CNACLD.
l If the called number is a PSTN number, go to Step 24.
l If the called number is an MSISDN, go to Step 25.
Step 24 Check whether the configuration of the route analysis index of the DN set is correct.
Step 25 Obtain the MSRN of the callee by tracing the MAP messages. Check whether the prefix of the
MSRN is correct by using LST CNACLD.
Step 27 Check whether the back board of the HRU board of the UMG8900 is an E8T, an E1G or a PAL.
Step 28 Check whether the back board of the replaced HRU board is an E8T, an E1G or a PAL. Check
whether the call setup is successful.
Step 29 Check whether the bearer bandwidth of the interface is zero by using LST IFBW.
Step 30 Modify the bearer bandwidth of the interface by using MOD IPIF.
l The bandwidth of the GE interface is less than 1,000 Mbit/s.
l The bandwidth of the FE interface is less than 100 Mbit/s.
l The bandwidth of the POS interface is less than 155 Mbit/s.
Check whether the call setup is successful.

Step 31 Check whether the bearer bandwidth of the interface or the VLAN is configured.

Step 32 Configure the bearer bandwidth of the interface or the VLAN. Check whether the call setup is
successful.
Step 33 Check whether the port for none-bearer purpose is reserved board of the specified IP address by
using LST RSVPORT.
l If not, contact Huawei's technical support.
Step 34 Remove the reserved port by using RMV RSVPORT. Check whether the call setup is
successful.
l If the call setup fails, contact Huawei's technical support.
----End
6.6.2 One-Way Audio and No Audio Faults

When one-way audio occurs, only the caller or the callee hears the peer's voice. When one-way
audio occurs, neither the caller nor the callee can hear the peer's voice. This section describes
the basic method used to remove one-way audio and no audio faults.
Procedure
Step 1 Check whether the network connection between the UMG8900s is normal by using ping.
You can perform the following operations between UMG8900 1 and UMG8900 2:
1. Ping the bearer IP address of UMG8900 2 on UMG8900 1.
2. Ping the bearer IP address of the gateway of UMG8900 2 on UMG8900 1.
The packet size is 400 to 500 bytes. Execute the ping command 10 times.
l The packet is sent successfully, but the delay and jitter are significant.
l The packet is sent successfully and the delay and jitter are normal. Go to Step 4.
l If the packet fails to be sent, go to Step 2.
Step 2 Trace the call flow. Check whether the call flow is normal.
l If it is normal, go to Step 6.
l If it is not normal, go to Step 3.
Step 3 Handle the faults in the call flow. Check whether the fault is removed.
l If the fault is removed, go to the end.
l If the fault is not removed, go to Step 6.
Step 4 Check the RTP traffic of each IP port. Check whether the traffic is normal.
l If it is normal, go to Step 6.

l If it is not normal, go to Step 5.
Step 5 If the RTP traffic of the IP port is not normal, handle the related fault. Check whether the fault
is removed.
Step 6 Capture the packets transmitted between the UMG8900s. Check whether the codec type and
codec packing interval are the same in the RTP packets.
l If they are the same, contact Huawei's technical support.
l If they are different, go to Step 7.
Step 7 Modify the configuration data of the MSOFTX3000 and UMG8900 so that the codec type and
codec packing interval are the same in the RTP packets. Check whether the fault is removed.
l If the fault is not removed, contact Huawei's technical support.
----End
6.6.3 Speech Noise

Speech noise refers to the sound other than the voice during the conversation. This section
describes the basic method to remove the speech noise fault.
Procedure
Step 1 Check whether the network connection between the UMG8900s is normal by using ping.
You can perform the following operations between UMG8900 1 and UMG8900 2:
1. Ping the bearer IP address of UMG8900 2 on UMG8900 1.
The packet size is 400 to 500 bytes. Run the ping command 10 times.
l If the delay and jitter are significant, or the packet loss ratio is over 5%, contact the
maintenance personnel of the bearer network.
l If packet loss is found but the packet loss ratio is less than 5%, go to Step 2.
l If the packet is sent successfully, go to Step 4.
Step 2 Check whether the packet loss compensation function is enabled by using LST TCPARA.
l If it is enabled, go to Step 4.
l If it is disabled, go to Step 3.
Step 3 Enable the packet loss compensation function. Check whether the fault is removed.
Step 4 Check whether voice codec uses G.711 by using LST MG.

Step 5 Disable the VAD function. Check whether the fault is removed.
Step 6 Set Special attributes to Not support echo cancellation by using MOD MGW. Check whether
the fault is removed.
Step 7 Check whether the peer is a PSTN subscriber.

l If the peer is a PSTN subscriber, go to Step 8.
l If the peer is not a PSTN subscriber, contact Huawei's technical support.
Step 8 Perform E1 loopback operation and check whether the fault is removed.
l If the fault is removed, contact the PSTN maintenance personnel.
l If the fault is not removed, contact Huawei's technical support.
----End

VoIP User Manual 7 Principles and Data Configuration of Bearer Network
7 Principles and Data Configuration of Bearer

Network
About This Chapter
This section describes the principles and data configuration of the VoIP bearer network.
7.1 Basic Principles

In VoIP applications, the basic principles involved in the bearer network include BFD, VRRP,
FRR, BGP/MPLS IP VPN, and resources isolation in VPN.
7.2 Reliable Networking Technologies for IP Network
This section describes the fault detection technologies to ensure the reliability of the IP bearer
network.
7.3 Example of VoIP Bearer Network Solution
This section describes the configuration of the VoIP bearer network by using an example of a
typical bearer network.

7 Principles and Data Configuration of Bearer Network VoIP User Manual
7.1 Basic Principles

In VoIP applications, the basic principles involved in the bearer network include BFD, VRRP,
FRR, BGP/MPLS IP VPN, and resources isolation in VPN.
BFD
Bidirectional Forwarding Detection (BFD) is a set of global detection mechanism for the entire
network. It is used to check and monitor the forwarding and connecting status of the links or IP
routes in the network. The BFD can be used to check connectivity of the data protocol transmitted
in the same path between two systems. This path can be a physical link or a logical link, or a
channel as well.
BFD provides the following functions:
l Providing light load and short duration detection to the channel fault between the
neighboring forwarding engines.
l Providing realtime detection to media of all types and the protocol layer by using single
mechanism.
VRRP
Virtual Router Redundancy Protocol (VRRP) is a protocol constituted by Internet Engineering
Task Force (IETF). This protocol is used to handle the reliability problem when a machine in
the LAN accesses external networks.
Normally, all machines in the internal network are configured with the same default route to a
breakout gateway. In this way, the machines can communicate with external networks. When
the breakout gateway is faulty, the machines cannot communicate with external networks.
The VRRP can be used to form a backup group by using a group of routers in the LAN. This
backup group can serve as a virtual router. The machines in the LAN can use the IP address of
the virtual router as the IP address of the default gateway. The machines then can communicate
with external networks through the virtual router.
The VRRP can be used to dynamically associate the virtual router with the physical router
transmitting service data. When the physical router is faulty, another router is selected to transmit
service data. The subscriber data is transparently transmitted in the process. In this way, the
communication between the internal network and the external network is not interrupted.
The advantage of the VRRP is: A default route with higher availability is provided without
changing networking modes or configuring dynamic routes or route detection protocol. In the
LAN capable of multicast or broadcast services (such as the Ethernet), the VRRP can be used
to provide a logical gateway to ensure the high availability of the transmission links. This can
solve the service interruption problem caused by certain router failure without modifying the
configuration of the route protocol.
FRR
Fast ReRoute (FRR) covers the technologies including IP FRR, LDP FRR, MPLS TE FRR, and
VPN FRR. The FRR can enable fast route switch through the backup data in the forwarding
plane.

l IP FRR
IP FRR can be used to record the status of each port and the load sharing status in the
network through the port status table in the forwarding module. Then the available route
can be set up for transmitting the data packet immediately. This technology can enable the
FRR function to take effect immediately. The technology can also enable data forwarding
with high reliability by using multiple data sharing entries in the forwarding table. In this
way, it can enable fast switch in case of failures.
l LDP FRR
When Label Distribution Protocol (LDP) works in the Downstream Unsolicited (DU) label
distribution mode, the ordered label control mode or the liberal retention mode, the Label
Switched Router (LSR) saves the received label mappings. Only the label mapping received
from the next hop according to the routing table is added to the label forwarding table. The
standby label switched path (LSP) is created. In this way, the system can perform fast
switchover upon failure without calculating routes after failure and re-establishing the LSP.
LDP FRR can provide the port-class protection scheme for the MPLS network.
l MPLS TE FRR
MPLS TE FRR is a common technology used for fast switchover upon failure. This
technology is used to establish a Traffic Engineering (TE) channel between two PE devices.
It is also used to establish a standby LSP for the desired active LSP. When the active LSP
is found unavailable (node failure or link failure), the system can switch the traffic to the
standby LSP. In this way, it enables fast switchover for services.
l VPN FRR
If the PE device is configured with VPN FRR, it records the route information of the selected
next hop node and the route information of the secondary next hop node. When the selected
next hop node is faulty, the PE device can use technologies such as BFD to detect that the
external channel to the selected next hop node is unavailable. The PE device can use the
secondary route in the forwarding record to forward messages.
BGP/MPLS IP VPN
BGP/MPLS IPx VPN is a PE-based L3VPN technology in the Provider Provisioned VPN
(PPVPN). This technology is used to advertise the VPN routes by using the BGP in the backbone
network of the service provider. This technology is also used to forward VPN packets by using
the MPLS in the backbone network of the service provider. It features in flexible networking
modes and good expandability. It can support MPLS QoS and MPLS TE easily, so it is applied
widely.
The BGP/MPLS IP VPN model consists of CE, PE and P.
l Customer Edge (CE): It is the network edge device of the customer. It is directly connected
to the Service Provider (SP). A CE can be a router, a switch or a workstation. Normally,
the CE is required to support the MPLS.
l Provider Edge (PE): It is the edge router of the service provider. It is the edge device of the
SP network. The PE is connected to CE directly. In the MPLS network, all the processing
of the VPN is handled PE.
l Provider (P): It is the backbone router of the SP network. The P is not connected to the CE
directly. The P is required to provide basic MPLS forwarding functions. It does not maintain
the VPN information.
Site refers to a group of systems that provide IP connectivity mutually. The IP connectivity of
the group of IP systems does not rely on network of the service provider. The Site is connected

to the network of the provider through the CE. A site can contain several CEs, but one CE belongs
to only one Site.
Resources Isolation VPN

The resources isolation VPN adopts the multiplexing technology to provide transmission
channels for various services. The resources isolation VPN can solve the problems of service
interference and resources allocation when the data of Expedited Forwarding (EF), Assured
Forwarding (AF) and Best-Effort (BE) is transmitted concurrently in the channel. It also provides
the peer to peer QoS assurance for the VPN users.
l To solve the interference problem between services, you can create a service channel
between two PE devices for each type of service. When the VPN service of the same type
uses the same channel between two PE devices, different services can be isolated. In this
way, resources contention can be avoided. If a new VPN user is added or a user requires
more bandwidths, you can adjust the bandwidth of a TE channel dynamically or create a
new TE channel to meet the user's need.
l When the VPN services of the same type use the same TE channel between two PE devices,
resources contention may occur. You can use committed access rate (CAR) to solve this
problem. After the service type of VPN user and the required bandwidth are acknowledged,
you can limit the bandwidth of the user service by setting the CAR at the entrance of the
TE channel.
7.2 Reliable Networking Technologies for IP Network

This section describes the fault detection technologies to ensure the reliability of the IP bearer
network.
The reliable networking principles for the IP bearer network are as follows:
l The IP network consists of core layer, convergence layer and edge layer. According to the
service status and service prediction, the accessing node at accessing edge layer adopts
redundant backup configuration. The accessing node is connected to convergence nodes.
The devices at the convergence layer are connected to the devices in dual-homing mode at
the upper layer. The devices at the convergence layer and core layer can be set according
to the actual situations. The devices at the core layer adopt full interconnection mode or
semi-interconnection mode to ensure reliable traffic forwarding.
l The devices in the same layer are interconnected. One device can be connected to multiple
devices.
l The devices in the lower layer are connected to dual-homing or multi-homing devices in
the upper layer.
l Parts of the network can be adjusted based on actual traffic.
Fault Detection of Intermediate Transmission Links

The BFD technology is often used to detect faults of intermediate transmission links. The system
can detect faults of the links in time through the bidirectional detection mechanism of the BFD.
The BFD is a bidirectional detection mechanism. The detection packets are sent on both sides
of the link. Normally, the detection packets can be received on both sides on time. If the detection
packets are not received at any side within specified period, the system determines that the link
is interrupted. Then the system reports this fault to other modules and starts link switchover.

Fault Detection of Downlink Port of Router at Access Point

The IP FRR technology is used to detect faults of the downlink port of the router at the access
point. The backup route information is set up at the forwarding layer. The system can use the
backup route information to enable fast route switchover in case of failure.
The link between the CE and the active PE is faulty when data is transmitted to the CE device
in the IP bearer network. In this case, the system can switch the route to the standby PE based
load sharing information in the forwarded packet. The data is forwarded from the active PE to
the standby PE, and then sent to the CE. In this way, the service is not affected. The neighbor
in the private network must be set up between the active PE and the standby PE. The backup
path to the CE can be set for the active PE.
Fault Detection of Intermediate Nodes and Links to Intermediate Nodes

The LDP/TE FRR technology is used to detect faults of intermediate nodes and links to
intermediate nodes. When the standby LSP is set up for the active LSP, the TE is also used to
assure the QoS. In this way, the QoS of the entire network is improved. If providers or the links
of providers are faulty when no transmission equipment is found between providers, the
neighboring PE starts the LSP switchover to maintain normal service operations.
If the LDP/TE FRR technology cannot detect faults when transmission equipment is found
between providers, only the BFD and OAM technology can be used to detect faults.
Fault Detection of Router at Remote Accessing Point

The VPN FRR technology is used to detect faults of routers at the remote accessing point. The
local router can use two remote routers in the VPN as the next hop. In this way, the active and
standby route configuration is set. When the active route is faulty, the system can switch data
transmission to the standby route without affecting services.
7.3 Example of VoIP Bearer Network Solution

This section describes the configuration of the VoIP bearer network by using an example of a
typical bearer network.
Figure 7-1 shows the typical networking of the VoIP bearer network.

Figure 7-1 Networking of the VoIP Bearer Network
POS1/0/0 11.4.1.2/30
PE1 PE2
POS1/0/0 11.4.1.1/30
OSPF VPN Multi-instance

Eth-trunk2 Eth-trunk2
Eth-trunk1
CE1 CE2
GE4/0/0 11.1.1.1/30
GE3/0/0 11.2.2.1/30 GE4/0/0 11.2.1.1/30
GE3/0/0 11.1.2.1/30
GE5/0/0,GE6/0/0
GE5/0/0,GE6/0/0
Media interface
Signal interface
MSC Server Backup Signal interface MGW
From the preceding figure, you can infer that:

l CE1 and CE2 are customer edge devices in the VPN. They are used to receive the media
streams and signaling streams from the MSC server and MGW.
l PE1 and PE2 are edge routers of the backbone network.
l By separating the VPN, media streams and signaling streams of CE and PE can be
transmitted separately. The VPN-instance signal is used to transmit signaling streams and
the VPN-instance media is used to transmit media streams.
l The VPN-instance signal and the VPN-instance media use router distinguisher (RD)s of
different attributes.
l OSPF VPN Multi-instance is used among CE1, CE2, PE1 and PE2.

This section describes the configuration flow of the VoIP bearer network.
7.3.2 Configuration Preparations
This section describes configuration preparations of the VoIP bearer network.
7.3.3 Data Configuration
7.3.4 Configuration File
This section describes configuration files of CEs and PEs.


This section describes the configuration flow of the VoIP bearer network.
To configure the VoIP bearer network, proceed as follows:
1. Assign VLANs and add ports to VLANs.
2. Configure VPN instances and bind the VLANIF interfaces to the VPN instances.
3. Configure OSPF VPN Multi-instance to exchange VPN route information between the CE
and the PE.
7.3.2 Configuration Preparations

This section describes configuration preparations of the VoIP bearer network.
To complete the data configuration in Figure 7-1, prepare the following data:
l VLAN ID and the IP address of VLANIF interface
l RD and real time (RT) attributes of VPN-instance signal and VPN-instance media
7.3.3 Data Configuration

Prerequisite
The physical port of the Eth-trunk port is bound.
Procedure
Step 1 Assign VLANs in CE1 and add the ports to the VLANs.
1. Assign VLAN2 for the signaling streams to connect the UMG on CE1.
<CE1> system-view
[CE1] interface gigabitethernet 5/0/0
[CE1-GigabitEthernet5/0/0] undo shutdown
[CE1-GigabitEthernet5/0/0] portswitch
[CE1-GigabitEthernet5/0/0] quit
# Create VLAN2.
[CE1] vlan 2
# Add GE5/0/0 and GE6/0/0 to VLAN2.
[CE1-vlan2] port gigabitethernet 5/0/0 6/0/0
[CE1-vlan2] quit
# Configure VLANIF interface.
[CE1] interface vlanif 2
[CE1-Vlanif2] undo shutdown
[CE1-Vlanif2] ip address 10.2.3.1 30
[CE1-Vlanif2] quit
2. Assign VLAN3 and VLAN4 for connecting signal stream interface and media stream
interface of CE2 on CE1.
# Create VLAN3.
[CE1] vlan 3
[CE1-Vlan3] quit


[CE1-Vlanif3] quit
# Create VLAN4.
[CE1] vlan 4
[CE1-Vlan4] quit
[CE1-Vlanif4] quit
interface of PE1 on CE1.
# Create VLAN5.
[CE1] vlan 5
[CE1-Vlan5] quit
[CE1-Vlanif5] quit
# Create VLAN6.
[CE1] vlan 6
[CE1-Vlan6] quit
[CE1-Vlanif6] quit
Step 2 Assign VLANs in CE22 and add the ports to the VLANs.
1. Assign VLAN2 for the signaling streams to connect the MGW on CE2.
<CE2> system-view
# Create VLAN2.
[CE2] vlan 2
[CE2-Vlan2] quit
# Add GE5/0/0 and GE6/0/0 to VLAN2.
[CE2-vlan2] port gigabitethernet 5/0/0 6/0/0
[CE2-vlan2] quit
[CE2-Vlanif2] quit
interface of CE1 on CE2.
# Create VLAN3.
[CE2] vlan 3
[CE1-Vlan3] quit
[CE2-Vlanif3] quit
# Create VLAN4.
[CE2] vlan 4
[CE2-Vlan4] quit


[CE2-Vlanif4] quit
interface of PE2 on CE2.
# Create VLAN5.
[CE2] vlan 5
[CE2-Vlan5] quit
[CE2-Vlanif5] quit
# Create VLAN6.
[CE2] vlan 6
[CE2-Vlan6] quit
[CE2-Vlanif6] quit
Step 3 Assign VLANs on PE1.
Assign VLAN5 and VLAN6 for connecting signal stream interface and media stream interface
of CE1 on PE1.
# Create VLAN5.
[PE1] vlan 5
[PE1-Vlan5] quit
[PE1] interface vlanif 5
[PE1-Vlanif5] undo shutdown
[PE1-Vlanif5] ip address 10.2.6.2 30
[PE1-Vlanif5] quit
# Create VLAN6.
[PE1] vlan 6
[PE1-Vlan6] quit
[PE1-Vlanif6] quit
Step 4 Assign VLANs on PE2.
Assign VLAN5 and VLAN6 for connecting signal stream interface and media stream interface
of CE2 on PE2.
# Create VLAN5.
[PE2] vlan 5
[PE2-Vlan5] quit
[PE2-Vlanif5] quit
# Create VLAN6.
[PE2] vlan 6
[PE2-Vlan6] quit
[PE2-Vlanif6] quit

Step 5 Configure allow-pass VLANs Eth-Trunk interface.

1. Configure the attribute of Eth-Trunk interface on CE1.
# Configure allow-pass VLANs Eth-Trunk 1.
[CE1] interface eth-trunk 1
[CE1-Eth-Trunk1] undo shutdown
[CE1-Eth-Trunk1] portswitch
[CE1-Eth-Trunk1] port link-type trunk
[CE1-Eth-Trunk1] port trunk allow-pass vlan 3 to 4
[CE1-Eth-Trunk1] quit
2. Configure the attribute of Eth-Trunk interface on CE2.

3. Configure the attribute of Eth-Trunk interface on PE1.

[PE1] interface eth-trunk 1
[PE1-Eth-Trunk1] undo shutdown
[PE1-Eth-Trunk1] portswitch
[PE1-Eth-Trunk1] port link-type trunk
[PE1-Eth-Trunk1] port trunk allow-pass vlan 5 to 6
[PE1-Eth-Trunk1] quit
4. Configure the attribute of Eth-Trunk interface on PE2.

[PE1] interface eth-trunk 1
[PE2-Eth-Trunk2] undo shutdown
[PE2-Eth-Trunk2] portswitch
[PE2-Eth-Trunk2] port link-type trunk
[PE2-Eth-Trunk2] port trunk allow-pass vlan 5 to 6
[PE2-Eth-Trunk2] quit
Step 6 Configure VPN instances and bind the VLANIF interfaces to the VPN instances.
# Configure CE1.
[CE1] ip vpn-instance signal
[CE1-vpn-instance-signal] route-distinguisher 100:1
[CE1-vpn-instance-signal] vpn-target 100:1 both
[CE1-vpn-instance-signal] quit
[CE1] ip vpn-instance media
[CE1-vpn-instance-media] route-distinguisher 100:2
[CE1-vpn-instance-media] vpn-target 100:2 both
[CE1-vpn-instance-media] quit
[CE1-GigabitEthernet3/0/0] ip binding vpn-instance signal
[CE1-GigabitEthernet3/0/0] ip address 11.1.2.1 30
[CE1-GigabitEthernet4/0/0] ip binding vpn-instance media

[CE1-Vlanif1] ip binding vpn-instance signal
[CE1-Vlanif1] quit
[CE1-Vlanif2] quit
[CE1-Vlanif3] ip binding vpn-instance media
[CE1-Vlanif3] quit
[CE1-Vlanif4] quit
[CE1-Vlanif5] quit
# Configure CE2.
[CE2] ip vpn-instance signal
[CE2-vpn-instance-signal] route-distinguisher 100:1
[CE2-vpn-instance-signal] vpn-target 100:1 both
[CE2-vpn-instance-signal] quit
[CE2] ip vpn-instance media
[CE2-vpn-instance-media] route-distinguisher 100:2
[CE2-vpn-instance-media] vpn-target 100:2 both
[CE2-vpn-instance-media] quit
[CE2-GigabitEthernet3/0/0] ip binding vpn-instance signal
[CE2-GigabitEthernet4/0/0] ip binding vpn-instance media
[CE2-Vlanif1] quit
[CE2-Vlanif2] quit
[CE2-Vlanif3] quit
[CE2-Vlanif4] quit
[CE2-Vlanif5] quit
# Configure PE1.
[PE1] ip vpn-instance signal
[PE1-vpn-instance-signal] route-distinguisher 100:1
[PE1-vpn-instance-signal] vpn-target 100:1 both
[PE1-vpn-instance-signal] quit
[PE1] ip vpn-instance media
[PE1-vpn-instance-media] route-distinguisher 100:2
[PE1-vpn-instance-media] vpn-target 100:2 both
[PE1-vpn-instance-media] quit
[PE1-Vlanif4] ip binding vpn-instance signal


[PE1-Vlanif4] quit
[PE1-Vlanif5] ip binding vpn-instance media
[PE1-Vlanif5] quit
# Configure PE2.
[PE2] ip vpn-instance signal
[PE2-vpn-instance-signal] route-distinguisher 100:1
[PE2-vpn-instance-signal] vpn-target 100:1 both
[PE2-vpn-instance-signal] quit
[PE2] ip vpn-instance media
[PE2-vpn-instance-media] route-distinguisher 100:2
[PE2-vpn-instance-media] vpn-target 100:2 both
[PE2-vpn-instance-media] quit
[PE2-Vlanif4] ip binding vpn-instance signal
[PE2-Vlanif4] quit
[PE2-Vlanif5] ip binding vpn-instance media
[PE2-Vlanif5] quit
Step 7 Configure OSPF VPN Multi-instance for exchanging VPN route information.
# Configure CE1.
[CE1]ospf 1 router-id 1.1.1.1 vpn-instance signal
[CE1-ospf-1]area 0
[CE1-ospf-1] silent-interface vlanif2
[CE1-ospf-1] silent-interface gigabitethernet 3/0/0
[CE1-ospf-1] spf-schedule-interval millisecond 100
[CE1-ospf-1] lsa-originate-interval 0
[CE1-ospf-1] lsa-arrival-interval 0
[CE1-ospf-1] bandwidth-reference 10000
[CE1-ospf-1] vpn-instance-capability simple
[CE1-ospf-1-area-0.0.0.0] network 10.2.3.0 0.0.0.3
[CE1-ospf-1-area-0.0.0.0] quit
[CE1-ospf-1] quit
[CE1] ospf 2 router-id 1.1.1.2 vpn-instance media
[CE1-ospf-2]area 0
# Configure CE2.
[CE2]ospf 1 router-id 1.1.2.1 vpn-instance signal
[CE2-ospf-1]area 0
[CE2-ospf-1] silent-interface vlanif1
[CE2-ospf-1-area-0.0.0.0] quit
[CE2-ospf-1] quit
[CE2] ospf 2 router-id 1.1.2.2 vpn-instance media

[CE2-ospf-2]area 0
# Configure PE1.
[PE1]ospf 1 router-id 1.1.3.1 vpn-instance signal
[PE1-ospf-1]area 0
[PE1-ospf-1] spf-schedule-interval millisecond 100
[PE1-ospf-1] lsa-originate-interval 0
[PE1-ospf-1] lsa-arrival-interval 0
[PE1-ospf-1] bandwidth-reference 10000
[PE1-ospf-1] vpn-instance-capability simple
[PE1-ospf-1-area-0.0.0.0] network 11.4.1.0 0.0.0.3
[PE1-ospf-1-area-0.0.0.0] quit
[PE1-ospf-1] quit
[PE1] ospf 2 router-id 1.1.3.2 vpn-instance media
[PE1-ospf-2]area 0
[PE1-ospf-2]spf-schedule-interval millisecond 100
# Configure PE2.
[PE2]ospf 1 router-id 1.1.4.1 vpn-instance signal
[PE2-ospf-1]area 0
[PE2-ospf-1-area-0.0.0.0] quit
[PE2-ospf-1] quit
[PE2] ospf 2 router-id 1.1.4.2 vpn-instance media
[PE2-ospf-2] area 0
----End
7.3.4 Configuration File

This section describes configuration files of CEs and PEs.
Configuration File of CE1

#
sysname CE1
#
vlan batch 2 to 6
#
ip vpn-instance signal
route-distinguisher 100:1

vpn-target 100:1 export-extcommunity

vpn-target 100:1 import-extcommunity
#
ip vpn-instance media
#
interface Vlanif2
ip binding vpn-instance signal
ip address 10.2.3.1 255.255.255.252
#
interface Vlanif3
ip address 10.2.4.1 255.255.255.252
#
interface Vlanif4
ip binding vpn-instance media
ip address 10.2.5.1 255.255.255.252
#
interface Vlanif5
ip address 10.2.6.1 255.255.255.252
#
interface Vlanif6
ip address 10.2.7.1 255.255.255.252
#
interface Eth-Trunk1
portswitch
port link-type trunk
port trunk allow-pass vlan 3 to 4
#
portswitch
#
interface GigabitEthernet3/0/0
ip address 11.1.2.1 255.255.255.252
#
ip address 10.1.1.1 255.255.255.252
#
portswitch
port default vlan 2
#
portswitch
port default vlan 2
#
ospf 1 router-id 1.1.1.1 vpn-instance signal
silent-interface Vlanif2
silent-interface GigabitEthernet3/0/0
spf-schedule-interval millisecond 100
lsa-originate-interval 0
lsa-arrival-interval 0
bandwidth-reference 10000
vpn-instance-capability simple
area 0.0.0.0
network 10.2.3.0 0.0.0.3
network 10.2.4.0 0.0.0.3
network 10.2.6.0 0.0.0.3
network 11.1.2.0 0.0.0.3
#
ospf 2 router-id 1.1.1.2 vpn-instance media

area 0.0.0.0
network 10.2.5.0 0.0.0.3
network 10.2.7.0 0.0.0.3
network 11.1.1.0 0.0.0.3
#
return
Configuration File of CE2

#
sysname CE2
#
vlan batch 2 to 6
#
#
#
interface Vlanif2
ip address 10.2.3.2 255.255.255.252
#
interface Vlanif3
ip address 10.2.4.2 255.255.255.252
#
interface Vlanif4
ip address 10.2.5.2 255.255.255.252
#
interface Vlanif5
ip address 10.2.8.1 255.255.255.252
#
interface Vlanif6
ip address 10.2.9.1 255.255.255.252
#
portswitch
#
portswitch
#
ip address 11.2.2.1 255.255.255.252
#
ip address 10.2.1.1 255.255.255.252
#

portswitch
port default vlan 2
#
portswitch
port default vlan 2
#
silent-interface Vlanif1
area 0.0.0.0
network 10.2.3.0 0.0.0.3
network 10.2.4.0 0.0.0.3
network 10.2.8.0 0.0.0.3
network 11.2.2.0 0.0.0.3
#
area 0.0.0.0
network 10.2.5.0 0.0.0.3
network 10.2.9.0 0.0.0.3
network 11.2.1.0 0.0.0.3
#
return
Configuration File of PE1

#
sysname PE1
#
vlan batch 5 to 6
#
#
#
interface Vlanif5
ip address 10.2.6.2 255.255.255.252
#
interface Vlanif6
ip address 10.2.7.2 255.255.255.252
#
portswitch
#

area 0.0.0.0
network 11.4.1.0 0.0.0.3
network 10.2.6.0 0.0.0.3
#
area 0.0.0.0
network 11.4.1.0 0.0.0.3
network 10.2.7.0 0.0.0.3
#
return
Configuration File of PE2

#
sysname PE2
#
vlan batch 5 to 6
#
#
#
interface Vlanif5
ip address 10.2.8.2 255.255.255.252
#
interface Vlanif6
ip address 10.2.9.2 255.255.255.252
#
portswitch
#
area 0.0.0.0
network 11.4.1.0 0.0.0.3
network 10.2.8.0 0.0.0.3
#
area 0.0.0.0
network 11.4.1.0 0.0.0.3
network 10.2.9.0 0.0.0.3
#
return

VoIP User Manual 8 VoIP Network Test
8 VoIP Network Test
About This Chapter
You can detect network problems in the test of the VoIP network. The quality of network can
be ensured in the formal operation. This section describes the tests related to the IP bearer
networks.
8.1 Core Network Test

This section describes the IP bearer tests for the MSOFTX3000.
8.2 Bearer Network Test
This section describes the tests for the VoIP bearer network.

8 VoIP Network Test VoIP User Manual
8.1 Core Network Test

This section describes the IP bearer tests for the MSOFTX3000.
8.1.1 IP Bearer Between Multiple MGWs (WCDMA)
Check whether the IP links between multiple MGWs in the WCDMA system can bear services
normally.
8.1.2 IP Bearer Between Multiple MGWs (GSM)
Check whether the IP links between multiple MGWs can bear services normally in the GSM
system.
Check whether service operation is normal when the IP bearer is used to bear the data of the Iu
interface.
8.1.4 TDM/IP Dual Bearer
Check whether the TDM/IP dual bearer function can be achieved.
8.1.1 IP Bearer Between Multiple MGWs (WCDMA)

Check whether the IP links between multiple MGWs in the WCDMA system can bear services
normally.
Prerequisites
l Subscribers A and B are served by the same MSC server in the WCDMA system. A and B
can perform location update successfully. The data of A and B is stored in the HLR and
VDB.
l Configure the system to use the IP bearer between internal MGWs.
Check whether Internal MGW media type is IP by using LST SRVNODE. If yes, no
processing is required. If not, set Internal MGW media type to IP and Internal MGW
connection type to All MGW connection by using MOD SRVNODE.
l Subscribers A and B connect the network from different MGWs in the same MSC server.
Procedure
Subscriber A calls subscriber B.
Expected Result
l The call is set up successfully.
l The conversation is stable and voice is clear.
l The resources are released successfully after the call ends.
8.1.2 IP Bearer Between Multiple MGWs (GSM)

Check whether the IP links between multiple MGWs can bear services normally in the GSM
system.

Prerequisites
l Subscribers A and B are served by the same MSC server in the GSM system. Subscribers
A and B can perform location update successfully. The data of A and B is stored in the
HLR and VLR.
l Configure the system to use the IP bearer between internal MGWs.
Check whether Internal MGW media type is IP by using LST SRVNODE. If yes, no
processing is required. If not, set Internal MGW media type to IP and Internal MGW
connection type to All MGW connection by using MOD SRVNODE.
l Subscribers A and B are accessed from different MGWs served by the same MSC server.
Procedure
Expected Result
l The call is created successfully.
l The conversation is stable and the voice is clear.
l The resources are released successfully after the call ends.

Check whether service operation is normal when the IP bearer is used to bear the data of the Iu
interface.
Prerequisites
l Subscribers A and B are WCDMA subscribers accessing the network through the same
RNC.
l The IP bearer is used between this RNC and the MGW.
Procedure
Expected Result
The call is connected normally. The voice is clear.
8.1.4 TDM/IP Dual Bearer

Check whether the TDM/IP dual bearer function can be achieved.
Prerequisites
l Subscribers A and B are in different MSC servers.
l Check whether Bearer first select mode is Invalid mode, TDM first select percent is
50%, and IP first select percent is 50% by using LST SRT. If yes, no processing is
required. If not, set the parameters by using MOD SRT.
l Check whether the IP and TDM circuits between MSC servers are idle.

Procedure
Subscriber A calls subscriber B twice.
Expected Result
l From the message trace, TDM bearer and IP bearer are selected equally for the signaling
trunk in the mixed office direction.
l The call is connected normally. The voice is clear.
8.2 Bearer Network Test

This section describes the tests for the VoIP bearer network.
Normally, the network must provide the switchover function in case of failure. If any link or NE
in the Figure 8-1 is faulty, the service operation is not affected.
Figure 8-1 Access bearer network of the core network
MSC CE1 AR1 AR3 CE3 MSC

Server A Server B
OSPF OSPF
MGW A CE2 AR2 AR4 CE4 MGW B
Active signaling link Active media link

Standby signaling link Standby media link
8.2.1 Checking System Operation When the Active Signaling Link of MSC Server Is Faulty
Check whether the switchover is normal and service operation is normal when the active
signaling link of the MSC server is faulty.
8.2.2 Checking System Operation When the Standby Signaling Link of MSC Server Is Faulty
Check whether the service operation is normal when the standby signaling link of the MSC
server is faulty.
8.2.3 Checking System Operation When the Active Signaling Link of MGW Is Faulty
Check whether the switchover and service operations are normal when the active signaling link
of the MGW is faulty.
8.2.4 Checking System Operation When the Standby Signaling Link of MGW Is Faulty
Check whether the service operation is normal when the standby signaling link of the MGW is
faulty.
8.2.5 Checking System Operation When the Active Media Link of MGW Is Faulty
Check whether the switchover is normal and service operation is normal when the active media
link of the MGW is faulty.

8.2.6 Checking System Operation When the Standby Media Link of MGW Is Faulty
Check whether the service operation is normal when the standby media link of the MGW is
faulty.
8.2.7 Checking System Operation When CE1 Is Faulty
Check whether the switchover is normal and service operation is normal when CE1 is faulty.
Check whether the service operation is normal when CE2 is faulty.
8.2.9 Checking System Operation When the Link Between CE1 and CE2 Is Faulty
Check whether the service operation is normal when the link between CE1 and CE2 is faulty.
8.2.10 Checking System Operation When the Link Between CE1 and AR1 Is Faulty
Check whether the link switchover is normal and service operation is normal when the link
between the CE1 and AR1 is faulty.
8.2.11 Checking System Operation When the Link Between CE2 and AR2 Is Faulty
8.2.1 Checking System Operation When the Active Signaling Link

of MSC Server Is Faulty
Check whether the switchover is normal and service operation is normal when the active
signaling link of the MSC server is faulty.
Prerequisites
l When the network works normally, the link between MSC server A and CE1 is
disconnected.
l The SCTP multi-homing function is enabled at the Nc interface and the Mc interface.
Procedure
1. The subscriber served by MSC server A calls the subscriber served by MSC server B.
2. After the call is connected, reconnect the link between MSC server A and CE1.
Expected Result
l After the link between MSC server A and CE1 is connected, the SCTP switchover is
performed. The data on the Nc and Mc interface is switched over to the active link again.
The SCTP association is not interrupted.
8.2.2 Checking System Operation When the Standby Signaling

Link of MSC Server Is Faulty
Check whether the service operation is normal when the standby signaling link of the MSC
server is faulty.

Prerequisites
l The network works normally.
l The SCTP multi-homing function is enabled at Nc interface and Mc interface.
Procedure
1. The subscriber in MSC server A calls the subscriber in MSC server B.
2. After the call is connected, disconnect the link between the MSC server and CE2.
3. Query the disconnection alarm of the standby link of the SCTP association at Nc and Mc
interface at the alarm console.
4. Re-connect the link between the MSC server and CE2.
Expected Result
l After the link between the MSC server and CE2 is disconnected, the disconnection alarm
of the standby link of the SCTP association at Nc and Mc interface can be queried at the
alarm console.
l After the link between the MSC server and CE2 is reconnected, the disconnection alarm
of the standby link of the SCTP association at Nc and Mc interface is removed.
8.2.3 Checking System Operation When the Active Signaling Link

of MGW Is Faulty
Check whether the switchover and service operations are normal when the active signaling link
of the MGW is faulty.
Prerequisites
l When the network works normally, disconnect the signaling link between the MGW and
CE1.
Procedure
1. The subscriber served by MSC server A calls the subscriber served by MSC server B.
2. After the call is connected, re-establish the signaling link between the MGW and CE1.
Expected Result
l After the signaling link between the MGW and CE1 is connected, the SCTP switchover is
performed. The data at the Nc and Mc interface is switched over to the active link again.
The SCTP association is not interrupted.
8.2.4 Checking System Operation When the Standby Signaling

Link of MGW Is Faulty
Check whether the service operation is normal when the standby signaling link of the MGW is
faulty.

Prerequisites
Procedure
2. After the call is connected, disconnect the link between the MGW server and CE2.
3. The disconnection alarm of the standby link of the SCTP association at Nc and Mc interface
can be queried at the alarm console.
4. Re-connect the link between the MGW and CE2.
Expected Result
l After the link between the MGW and CE2 is disconnected, the disconnection alarm of the
standby link of the SCTP association at Nc and Mc interface can be queried at the alarm
console.
l After the link between the MGW and CE2 is reconnected, the disconnection alarm of the
standby link of the SCTP association at Nc and Mc interface is removed.
8.2.5 Checking System Operation When the Active Media Link of

MGW Is Faulty
Check whether the switchover is normal and service operation is normal when the active media
link of the MGW is faulty.
Prerequisites
l When the network works normally, disconnect the media link between MGW and CE1.
Procedure
2. After the call is connected, reconnect the media link between the MGW and CE1.
Expected Result
l After the media link between the MGW and CE1 is reconnected, the SCTP switchover is
performed. Nc and Mc interface are switched over to the active link again. The SCTP
association is not interrupted.
8.2.6 Checking System Operation When the Standby Media Link of

MGW Is Faulty
Check whether the service operation is normal when the standby media link of the MGW is
faulty.

Prerequisites
Procedure
2. After the call is connected, disconnect the link between the MGW server and CE2.
Expected Result

Check whether the switchover is normal and service operation is normal when CE1 is faulty.
Prerequisites
Procedure
2. After the call is connected, power off CE1.
3. Restart CE1 after five minutes.
Expected Result
l When CE1 is powered off,
The SCTP multi-homing switchover is performed at Nc and Mc interfaces and the SCTP
CE2 is the active gateway of Nb interface. The media link of MGW A is the standby
media link.
l After CE1 is restarted,
CE1 is the active gateway of Nb interface. The media link of MGW A is the active
media link.

Check whether the service operation is normal when CE2 is faulty.
Prerequisites

Procedure
2. After the call is connected, power off CE2.
3. Restart CE2 after five minutes.
Expected Result
l When CE2 is powered off,
The disconnection alarm of the standby link of the SCTP association at Nc and Mc
interfaces can be queried at the alarm console.
l After CE2 is restarted, the disconnection alarm of the standby link of the SCTP association
at Nc and Mc interface is cleared.
8.2.9 Checking System Operation When the Link Between CE1 and
CE2 Is Faulty
Check whether the service operation is normal when the link between CE1 and CE2 is faulty.
Prerequisites
Procedure
2. After the call is connected, disconnect the link between CE1 and CE2.
Expected Result
AR1 Is Faulty
Prerequisites
Procedure

2. After the call is connected, disconnect the link between CE1 and AR1.
3. Reconnect the link between CE1 and AR1 after five minutes.
Expected Result
l After the link between CE1 and AR1 is disconnected,
The SCTP multi-homing switchover is performed at Nc and Mc interface and the SCTP
The media link of MGW A is switched over to the standby media link.
l After the link between CE1 and AR1 is reconnected,
The media link of MGW A is switched over to the active media link.
AR2 Is Faulty
Prerequisites
Procedure
2. After the call is connected, disconnect the link between CE2 and AR2.
3. Reconnect the link between CE2 and AR2 after five minutes.
Expected Result
l After the link between CE2 and AR2 is disconnected,
The disconnection alarm of the standby link of the SCTP association at Nc and Mc
interface can be queried at the alarm console.
l After the link between CE2 and AR2 is reconnected, the disconnection alarm of the standby
link of the SCTP association at Nc and Mc interface is removed.

VoIP User Manual 9 FAQ
9 FAQ
This section describes FAQs related to the IP bearer networks.
How is DTMF applied in the IP/TDM dual bearers?

The TDM bearer uses inband Dual Tone Multi-Frequency (DTMF) to transmit data. The IP
bearer uses outband DTMF to transmit data. If the DTMF signal of the TDM bearer is transmitted
in the IP bearer without conversion, the transmission quality cannot be assured and the DTMF
signal may be lost. Therefore, the DTMF conversion is required in the case of IP/TDM dual
bearers.
l When the data is transmitted from the TDM bearer to the IP bearer, the system must check
the DTMF signal at the TDM node and convert the intraband DTMF signal to the outband
DTMF signal.
l When the data is transmitted from the IP bearer to the TDM bearer, the MSC server must
convert the outband DTMF signal to the inband DTMF signal.
l When only the IP bearer is used, only the outband DTMF can be used.
How to assure the RBT quality and control the VAD function during conversation?
The VAD function can save bandwidth and improve utilization ratio of transmission resources
when this function is enabled during conversation. The VAD function, however, can reduce the
RBT quality greatly. Therefore, the MSC server must control the MGW to enable or disable the
VAD function based on conversation type and phrase.
How does the MSC server perform flow control in the IP bearer network?
The MSC server can perform flow control in the IP bearer network with the following methods:
l Active flow control: The MSC server can obtain the supported calls based on the bandwidth
of the IP bearer network. The MSC server can perform flow control to calls based on the
calls.
l Feedback flow control: The MSC server requires the MGW to sample calls and call quality
in the IP bearer network. The MSC server performs the flow control based on the sampling
result reported by the MGW.
In the feedback flow control, the MSC server can perform IP QoS tests to the nodes that have
established calls between MGWs. When QoS loss is found between MGWs, the MSC server

9 FAQ VoIP User Manual
performs flow control to subsequent calls. In this way, the voice quality of the ongoing calls is
not affected by QoS loss.

VoIP User Manual Index
Index
C based on performance measurement, 6-7

call setup failure, 6-9
concept determine the scope and category, 6-4
BICC CMN, 1-5 fault information collecting, 6-4
IP-based A interface, 1-8 find out the cause, 6-5
IP-based Iu interface, 1-9 one-way audio and no audio, 6-13
SCTP multi-homing, 1-4 speech noise, 6-14
speech codec, 1-10
TDM/IP dual bearers, 1-7 V
TrFO, 1-6
VoIP bearer network
I BFD, 7-2
BGP/MPLS IP VPN, 7-3
interface fault detection technology, 7-4
A, 1-18 FRR, 7-2
Iu, 1-19 local router, 1-13
Mc, 1-17 remote router, 1-14
Nb, 1-18 resources isolation VPN, 7-4
Nc, 1-17 solution, 7-5
VRRP, 7-2
P VoIP network test
bearer network test, 8-4
planning core network test, 8-2
bandwidth planning, 2-5
networking planning, 2-4
system requirement, 2-2
Q
QoS control
equipment-based, 1-11
jitter reduction-based, 1-12
priority-based, 1-12
R
routine maintenance
MSOFTX3000 performance entity measurement,
5-2
UMG8900 performance entity measurement, 5-6
T
troubleshooting
based on alarm, 6-6
Issue 02 (2008-04-20) Huawei Proprietary and Confidential i-1


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HUAWEI MSOFTX3000 Mobile SoftSwitch Center

VoIP User Manual

Huawei Proprietary and Confidential

Huawei Technologies Co., Ltd.

Copyright Huawei Technologies Co., Ltd. 2008. All rights reserved.

Trademarks and Permissions

Huawei Proprietary and Confidential

About This Document.....................................................................................................................1

3 Newly Established VoIP Networking....................................................................................3-1

Issue 02 (2008-04-20) Huawei Proprietary and Confidential i

3.1 Single-MGW Networking...............................................................................................................................3-5

4 VoIP Networking Evolving from Network in Use..............................................................4-1

5 Routine Maintenance on the VoIP Networking.................................................................. 5-1

ii Huawei Proprietary and Confidential Issue 02 (2008-04-20)

6.6.3 Speech Noise........................................................................................................................................6-14

7 Principles and Data Configuration of Bearer Network......................................................7-1

8 VoIP Network Test....................................................................................................................8-1

Issue 02 (2008-04-20) Huawei Proprietary and Confidential iii

Figure 1-1 The paths related to the SCTP association ........................................................................................1-4

Issue 02 (2008-04-20) Huawei Proprietary and Confidential v

Figure 4-8 CMN networking..............................................................................................................................4-30

vi Huawei Proprietary and Confidential Issue 02 (2008-04-20)

Issue 02 (2008-04-20) Huawei Proprietary and Confidential vii

About This Document

Product Name Version

l Mobile network maintenance engineers

Issue 02 (2008-04-20) Huawei Proprietary and Confidential 1

9 FAQ This section describes FAQs related to the IP bearer networks.

Indicates a hazard with a high level of risk which, if not

Indicates a hazard with a medium or low level of risk which,

Indicates a potentially hazardous situation that, if not

2 Huawei Proprietary and Confidential Issue 02 (2008-04-20)

NOTE Provides additional information to emphasize or

Times New Roman Normal paragraphs are in Times New Roman.

Italic Book titles are in italics.

Boldface The keywords of a command line are in boldface.

Italic Command arguments are in italic.

[] Items (keywords or arguments) in square brackets [ ] are

{ x | y | ... } Alternative items are grouped in braces and separated by

[ x | y | ... ] Optional alternative items are grouped in square brackets

{ x | y | ... } * Alternative items are grouped in braces and separated by

Issue 02 (2008-04-20) Huawei Proprietary and Confidential 3

Updates in Issue 02 (2008-04-20)

Updates in Issue 01 (2008-02-25)

4 Huawei Proprietary and Confidential Issue 02 (2008-04-20)

About This Chapter

Issue 02 (2008-04-20) Huawei Proprietary and Confidential 1-1

1-2 Huawei Proprietary and Confidential Issue 02 (2008-04-20)

1.2 Basic Concepts of VoIP

1.2.1 SCTP Multi-Homing

Issue 02 (2008-04-20) Huawei Proprietary and Confidential 1-3

1.2.6 IP-based Iu Interface

1.2.1 SCTP Multi-Homing

Figure 1-1 The paths related to the SCTP association

Path Active and Standby Path Priority

[IP1,IP3] Active High

[IP2,IP4] Standby Low

1-4 Huawei Proprietary and Confidential Issue 02 (2008-04-20)

Note the following for transmission paths:

1.2.2 BICC CMN

Issue 02 (2008-04-20) Huawei Proprietary and Confidential 1-5

1-6 Huawei Proprietary and Confidential Issue 02 (2008-04-20)

Figure 1-2 Principles of the TrFO function