Documenti di Didattica
Documenti di Professioni
Documenti di Cultura
V100R006C05
Issue 02
Date 2008-04-20
Part Number 00399775
Website: http://www.huawei.com
Email: support@huawei.com
and other Huawei trademarks are the property of Huawei Technologies Co., Ltd.
All other trademarks and trade names mentioned in this document are the property of their respective holders.
Notice
The information in this document is subject to change without notice. Every effort has been made in the
preparation of this document to ensure accuracy of the contents, but the statements, information, and
recommendations in this document do not constitute a warranty of any kind, express or implied.
Contents
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
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
9 FAQ...............................................................................................................................................9-1
Index.................................................................................................................................................i-1
Figures
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
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.
MSOFTX3000 V100R006C05
Intended Audience
The intended audiences of this document are:
Organization
This document consists of nine sections and is organized as follows.
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.
Conventions
Symbol Conventions
The following symbols may be found in this document. They are defined as follows.
Symbol Description
Symbol Description
TIP Indicates a tip that may help you solve a problem or save
your time.
General Conventions
Convention Description
Boldface Names of files, directories, folders, and users are in boldface. For
example, log in as user root.
Command Conventions
Convention Description
GUI Conventions
Convention Description
Boldface Buttons, menus, parameters, tabs, window, and dialog titles are in
boldface. For example, click OK.
Convention Description
> 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.
1 Introduction to VoIP
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
After the IP bearer network is introduced, you must pay attention to the security issue of the
mobile network.
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.
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.
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
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.
(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.
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%.
Figure 1-3 shows the protocol stack used when the data on the A interface is transmitted through
the IP bearer network.
BSSAP
SCCP
M3UA RTP
SCTP UDP
IP IP
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).
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.
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.
AMR
The AMR supports voice transmission of different rates. There are eight compression methods.
The corresponding rates are as follow.
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.
The QoS covers the unidirectional transmission delay, packet drop rate and jitter.
The major problem that causes the decrease of QoS can be:
The system needs to perform QoS control concerning equipment, services, subscribers, and
reducing jitter.
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.
(4) (1)
(4)
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.
MGW BR BR MGW
CR CR
A B C A B C A B C
RTP time
10 30 50 10 30 50 10 30 50
stamp
Jitter Buffer
PSTN RNC
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.
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.
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.
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.
SDH
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 remote router networking mode can be adopted in the following conditions:
Nc Interface
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.
BICC
M3UA
SCTP
IP
Mc Interface
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.
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.
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.
BSSAP
SCCP
M3UA RTP
SCTP UDP
IP IP
Iu Interface
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.
RANAP
SCCP IuUP
M3UA RTP
SCTP UDP
IP IP
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%.
2 VoIP Planning
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.
Good 5 0.01% 1
Normal 50 0.1% 10
Bad 200 5% 40
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.
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.
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.
CR
CR
CR
CR
CR
BR BR
AR AR AR AR
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.
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
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
Roundup (value, 0) indicates that the value is rounded up to an integral number.
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 Growth rate = Number of subscribers on the target network (traffic) Number of
subscribers on the original network (traffic)
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)
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
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 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
Formulas for Calculating the Parameters Required for the Target Network
To calculate the parameters required for the target network, use the following formulas based
on the preceding parameters of the original network:
Input Parameters
To calculate the traffic over the Nb interface, the following input parameters are required:
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:
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
Formulas for Calculating the Parameters Required for the Target Network
To calculate the parameters required for the target network, use the following formulas based
on the preceding parameters of the original network:
Input Parameters
To calculate BICC CICs, the following input parameters are required:
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
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)
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)
Input Parameters
To calculate the number of TDM signaling links, the following input parameters are required:
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 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.
Formulas for Calculating the Parameters Required for the Target Network
To calculate the parameters required for the target network, use the following formulas based
on the preceding parameters of the original network:
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
To calculate the parameters required for the original network, use the following formulas based
on the preceding input parameters:
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
To calculate the parameters required for the target network, use the following formulas based
on the preceding parameters of the original network:
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.
HLR SCP
MSC Server
MSC Server
MGW MGW
TDM
IP
BSC BSC
Start
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.
This section describes the VoIP-related data that must be configured when the VoIP service is
used in the multi-MGW networking by taking a typical multi-MGW networking mode as an
example.
3.3 CMN Networking
This section describes the VoIP-related data that must be configured when the VoIP service is
used in the CMN networking by taking a typical CMN 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.
MSC Server
HLR
STP
CE1 CE2
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
Figure 3-4 Procedure of the data configuration on the single-MGW VoIP networking
Start
End
For the networking shown in Figure 3-3, the interworking data to be negotiated is as follows.
Country code 86
Table 3-2 Interworking data between the MSC server and other NEs (single-MGW)
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.
Procedure
Step 1 Load a license file.
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.
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.
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.
Desc Add a VLR configuration data record by setting the MSRN/HON number
ripti allocation mode.
on
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.
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
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.
Descr Add an MGW data record where MGW name is MGW1, Transport
iptio protocol is SCTP protocol, and Server/Client is Server.
n
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.
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.
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,
riptio Signaling point code of local entity is A0001, and Local entity type is
n Application server.
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.
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.
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.
Desc Add an M3UA link set data record where Linkset name is MGW1-A, and
riptio Adjacent entity name is MGW1-A.
n
Desc Add an M3UA route data record. The name of the destination entity of the
riptio UMG8900 (SG) route is MGW1-A.
n
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.
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.
1. Add an M3UA local entity data record.
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
n Application server.
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.
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 the 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.
Rem To ensure the reliability of M3UA links, those M3UA links to the same
arks destination signaling point must be assigned to different WBSGs.
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
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
Desc Add two M3UA link data records. For one link, set Client/Server to Client,
riptio WBSG module number to 141, Link name to MSX01-1, Local port
n number to 7000, and Peer port number to 7000; for the other link, set Client/
Server to Client, WBSG module number to 142, Link name to MSX01-2,
Local port number to 7001, and Peer port number to 7001.
Rem To ensure the reliability of M3UA links, those M3UA links to the same
arks destination signaling point must be assigned to different WBSGs.
Desc Add an office direction data record where Office direction name is MSX01,
riptio and Signaling type is M3UA.
n
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.
Desc Add a route analysis data record where Route selection name is RSSN1.
riptio
n
Rem Usually, set Caller category to All, and Address information indicator and
arks Transmission capability of the command to All categories.
Desc Add a BICC trunk group data record where Trunk group name is MSX01-
riptio MGW1, and MGW name is MGW1.
n
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.
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.
Desc Add SCCP sub-system numbers where SCCP sub-system numbers are
riptio SCMGMSCVLR.
n
Desc Add an office direction data record where Office direction name is MSX02,
riptio and DPC1 is C0002.
n
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.
Desc Add a route analysis data record where Route selection name is RSSN1.
riptio
n
Rem Usually, set Caller category to All, and Address information indicator and
arks Transmission capability of the command to All categories.
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 SCCP destination signaling point where DSP name is BSC1DPC is
riptio A01.
n
6. Add a BSC.
Desc Add a BSC where Route selection source name is RSSN1 and LAI
riptio number is 460000003.
n
Desc Add an A-interface circuit pool where Circuit pool name is BSC1 .
riptio
n
Desc Add an A-interface trunk group where Trunk group name is BSC1 and MGW
riptio name is MGW1.
n
Desc Add A-interface trunk circuits where Start CIC is 640 and End CIC is 703.
riptio
n
----End
Procedure
Step 1 Configure local office information.
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.
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.
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.
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
Desc Set a virtual MGW data record where Virtual media gateway id is 0, and
riptio Virtual media gateway MID type is IP.
n
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.
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.
Rem l The numbers of the H.248 links of the SCTP type configured on the
arks UMG8900 must be greater than 47.
l To ensure reliability and availability, you must configure multiple H.248
links between the MSOFTX3000 and the MGW.
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.
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.
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.
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.
Desc Add an M3UA local entity data record where Local Entity Index is 1, Local
riptio Entity Name is MGW1-B, Source signaling Point Code is 101, and Local
n Entity Type is signaling_Gateway.
Desc Add an M3UA destination entity data record. Destination Entity Index is set
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.
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.
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.
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.
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.
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
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.
Rem The destination signaling point of the MTP3 route is BSC1. Therefore, the route
arks index is 0.
Rem To ensure the reliability of MTP2 links, those MTP2 links must be assigned to
arks different SPFs.
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.
Rem The signaling link code must be negotiated with BSC1. Otherwise, the MPT3
arks link cannot be set up properly.
Desc Add an MTP3 destination signaling point data record where DSP index is 1,
riptio DSP name is MSX02, and OPC index is 1.
n
Rem The value of STP function on the UMG8900 must be consistent with the value
arks of STP function on MSX02. Otherwise, the MTP3 link cannot be set up
properly.
Desc Add an MTP3 link set data record where Linkset index is 1, Linkset name is
riptio MSX02, and Adjacent DSP index is 1.
n
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 MSX02. Here, Adjacent DSP index is set to
1.
Desc Add an MTP3 route data record where Route index is 1, and Route name is
riptio MSX02.
n
Rem The destination signaling point of the MTP3 route is MSX02. Therefore, the
arks route index is 1.
Rem To ensure the reliability of MTP2 links, those MTP2 links must be assigned to
arks different SPFs.
Rem The signaling link code must be negotiated with TDMMSC. Otherwise, the
arks MPT3 link cannot be set up properly.
----End
Networking Description
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.
MSC Server
HLR
STP
CE1 CE2
MSC1
MGW1 MGW2 MGW3
TDM
IP
BSC MSC2
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/
SCCP/MTP3/MTP2/MTP1.
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 Signaling point name
l Signaling point name
l Office direction name
l IP address
l SCTP port number
Figure 3-6 Procedure of the data configuration on the multi-MGW VoIP networking
Start
End
For the networking shown in Figure 3-5, the interworking data to be negotiated is as follows.
Country code 86
Table 3-4 Interworking data between the MSC server and other NEs (multi-MGW)
Name MSC MGW1 MGW2 MGW3 BSC MSC1 MSC2
Server
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.
Procedure
Step 1 Load a license file.
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.
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.
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.
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.
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.
Rem The preceding is the default configuration, which can be modified according
arks to the actual networking.
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.
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.
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.
1. Add an M3UA local entity data record.
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
n Application server.
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.
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 the 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.
Rem To ensure the reliability of M3UA links, those M3UA links to the same
arks destination signaling point must be assigned to different WBSGs.
Desc Add an M3UA destination entity data record where Destination entity name
riptio is MSX01, indicating the IP MSC directly connected to the MSOFTX3000.
n
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
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
Rem To ensure the reliability of M3UA links, those M3UA links to the same
arks destination signaling point must be assigned to different WBSGs.
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.
Desc Add a route analysis data record where Route selection name is RSSN1.
riptio
n
Rem Usually, set Caller category to All, and Address information indicator and
arks Transmission capability of the command to All categories.
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.
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.
Desc Add an M3UA route data record. The name of the destination entity of the
riptio MSX02 route is MSC2.
n
Desc Add an SCCP destination signaling point where DSP name is MSX02 and
riptio DPC is C0002.
n
Desc Add SCCP sub-system numbers where SCCP sub-system numbers are
riptio SCMGMSCVLR.
n
Desc Add an office direction data record where Office direction name is MSX02,
riptio and DPC1 is C0002.
n
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.
Desc Add a route analysis data record where Route selection name is RSSN1.
riptio
n
Rem Usually, set Caller category to All, and Address information indicator and
arks Transmission capability of the command to All categories.
Desc Add an SS7 trunk group data record where Trunk group name is MSX02, and
riptio MGW name is MGW1.
n
Desc Add an M3UA destination entity data record where Destination entity name
riptio is BSC1, indicating the BSC.
n
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
Desc Add an SCCP destination signaling point where DSP name is BSC1DPC is
riptio A01.
n
6. Add a BSC.
Desc Add a BSC where Route selection source name is RSSN1 and LAI
riptio number is 460000003.
n
Desc Add 2G LAI or GCI informations where Global cell ID are 460000003 and
riptio 4600000030003, Call source name is CALL_SRC1.
n
Desc Add an A-interface circuit pool where Circuit pool name is BSC1 .
riptio
n
Desc Add an A-interface trunk group where Trunk group name is BSC1 and MGW
riptio name is MGW1.
n
Desc Add A-interface trunk circuits where Start CIC is 640 and End CIC is 703.
riptio
n
----End
Procedure
Step 1 Configure 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.
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.
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.
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
Rem l The numbers of the H.248 links of the SCTP type configured on the
arks UMG8900 must be greater than 47.
l To ensure reliability and availability, you must configure multiple H.248
links between the MSOFTX3000 and the MGW.
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.
Desc Add an M3UA route data record. The index of the destination entity of the
riptio MSOFTX3000 (AS) is 0.
n
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.
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.
Desc Add an M3UA local entity data record where Local Entity Index is 1, Local
riptio Entity Name is MGW1-B, Source signaling Point Code is 101, and Local
n Entity Type is signaling_Gateway.
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.
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.
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.
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.
Rem The destination signaling point of the MTP3 route is BSC1. Therefore, the route
arks index is 0.
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.
Rem To ensure the reliability of MTP2 links, those MTP2 links must be assigned to
arks different SPFs.
Rem The signaling link code must be negotiated with BSC1. Otherwise, the MPT3
arks link cannot be set up properly.
Rem The value of STP function on the UMG8900 must be consistent with the value
arks of STP function on MSX02. Otherwise, the MTP3 link cannot be set up
properly.
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 MSX02. Here, Adjacent DSP index is set to
1.
Rem The destination signaling point of the MTP3 route is MSX02. Therefore, the
arks route index is 1.
Desc Add two MTP2 signaling link data records. For one MTP2 signaling link, set
riptio Interface board No. to 2, Interface board type to E32, E1T1 No. to 1, Start
n time slot to 16, and Link type to MTP3 64K LINK. For the other MTP2
signaling link, set Interface board No. to 3, Interface board type to E32,
E1T1 No. to 1, Start time slot to 16, and Link type to MTP3 64K LINK.
Rem To ensure the reliability of MTP2 links, those MTP2 links must be assigned to
arks different SPFs.
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
Networking Description
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.
CMN
MSC MSC
Server1 Server2
TDM
BSC IP MSC
Interworking Data
For the CMN networking, you must plan and collect the following data:
l Signaling point name
l Signaling point name
l Office direction name
l IP address
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
End
For the networking shown in Figure 3-7, the interworking data to be negotiated is as follows.
Table 3-5 Interworking data between the MSC server and other NEs (CMN)
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
Procedure
Step 1 Load a license file.
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.
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.
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.
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.
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
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.
Desc Add an M3UA local entity data record where Local entity name is MSX,
riptio Signaling point code of local entity is A0001, and Local entity type is
n Application server.
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.
Desc Add an M3UA destination entity data record where Destination entity name
riptio is MSX01, indicating the IP MSC directly connected to the MSOFTX3000.
n
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
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
Desc Add two M3UA link data records. For one link, set Client/Server to Client,
riptio WBSG module number to 141, Link name to MSX01-1, Local port
n number to 7000, and Peer port number to 7000; for the other link, set Client/
Server to Client, WBSG module number to 142, Link name to MSX01-2,
Local port number to 7001, and Peer port number to 7001.
Rem To ensure the reliability of M3UA links, those M3UA links to the same
arks destination signaling point must be assigned to different WBSGs.
Desc Add an office direction data record where Office direction name is MSX01,
riptio and Signaling type is M3UA.
n
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.
Desc Add a route analysis data record where Route selection name is RSSN1.
riptio
n
Rem Usually, set Caller category to All, and Address information indicator and
arks Transmission capability of the command to All categories.
Desc Add a BICC trunk group data record where Trunk group name is MSX01,
riptio and MGW name is CMN.
n
Desc Add a BICC CIC module where Office direction name is MSX01.
riptio
n
Desc Add an M3UA destination entity data record where Destination entity name
riptio is MSX02, indicating the MSC2.
n
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.
Desc Add an M3UA link set data record where Linkset name is MSX02, and
riptio Adjacent entity name is MSX02.
n
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
riptio MSX02 route is MSC2.
n
Desc Add two M3UA link data records. For one link, set Client/Server to Client,
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.
Rem To ensure the reliability of M3UA links, those M3UA links to the same
arks destination signaling point must be assigned to different WBSGs.
Desc Add an office direction data record where Office direction name is MSX02,
riptio and Signaling type is M3UA.
n
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 supports the TrFo function, and the peer office supports
the redundancy function.
Desc Add a route analysis data record where Route selection name is RSSN1.
riptio
n
Rem Usually, set Caller category to All, and Address information indicator and
arks Transmission capability of the command to All categories.
----End
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
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.
4.1 Single-MGW Networking
This section describes the data configuration in the single-MGW VoIP networking mode.
4.2 Multi-MGW Networking
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.
Original Network
Figure 4-1 shows the typical single-MGW networking.
STP
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 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/
SCCP/MTP3/MTP2/MTP1.
Destination Network
Figure 4-2 shows the typical single-MGW VoIP networking.
STP
CE1 CE2
TDM
BSC IP
transfer point. The structure of the protocol stack at the MSC server side is BSSAP/SCCP/
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 Signaling point name
l Signaling point code
l Office direction name
l IP address
l SCTP port
Figure 4-3 Flow of the data configuration for the single-MGW VoIP networking
Start
End
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.
Procedure
Step 1 Modify MGW data.
Descr Modify the MGW data record to support the IP bearer and codec.
iption
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
Desc Add a BICC trunk group data record where Trunk group name is MSX01,
riptio and MGW name is MGW1.
n
Desc Add a BICC CIC module where Office direction name is MSX01.
riptio
n
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.
Desc Add a local office signaling point code where National network code is
riptio A0002.
n
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.
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 two M3UA link data records. For one link, set Client/Server to Client,
riptio WBSG module number to 141, Link name to SX02-1, Local port number
n to 7004, and Peer port number to 7004; for the other link, set Client/Server
to Client, WBSG module number to 142, Link name to SX02-2, Local port
number to 7005, and Peer port number to 7005.
Desc Add an M3UA route data record. The name of the destination entity of the
riptio MSX02 route is MSC2.
n
Desc Add an office direction data record where Office direction name is MSX02,
riptio Signaling type is M3UA, BICC call source name is CALL_SRC1, and
n Multi-area name is 1.
Rem BICC call source name and Multi-area name should be same to original
arks network.
Desc Add a route analysis data record where Route selection name is RSN2, and
riptio Signaling priority is Prefer to select BICC IP.
n
Rem Usually, set Caller category to All, and Address information indicator and
arks Transmission capability of the command to All categories.
Desc Add a BICC trunk group data record where Trunk group name is MSX02,
riptio and MGW name is MGW1.
n
Desc Add a BICC CIC module data record where Office direction name is
riptio MSX02.
n
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.
Step 4 Modify the M2UA signalling to M3UA signalling of national reserved network between MSC
Server and MGW.
1. Add an M3UA local entity data record.
Desc Add an M3UA local entity data record where Local entity name is MSXd,
riptio Signaling point code of local entity is 001, and Local entity type is
n Application server.
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.
Desc Add an M3UA route data record. The name of the destination entity of the
riptio UMG8900 (SG) route is MGW1.
n
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.
1. Add an M3UA destination entity data record.
Desc Add an M3UA destination entity data record where Destination entity name
riptio is BSC1, indicating the BSC.
n
Desc Add an M3UA route data record. The name of the destination entity of BSC1
riptio route is BSC1.
n
----End
Procedure
Step 1 Configure data for IP bearer network on MGW.
1. Add an IP 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
Configure M3UA data that is used for the national reserved network.
Desc Add an M3UA local entity data record where Local Entity Index is 1, Local
riptio Entity Name is MGW1-B, Source signaling Point Code is 101, and Local
n Entity Type is signaling_Gateway.
Desc Add an M3UA destination entity data record. Destination Entity Index is set
riptio to 1, and Destination Entity Name is set to MSXd, indicating the
n MSOFTX3000 (AS) that is directly connected to the UMG8900. The local
entity index corresponding to the destination entity index is 1.
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.
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
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.
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
Original Network
Figure 4-4 shows the typical multi-MGW networking.
MSC Server
HLR
STP
MGW2
TDM
BSC IP
M2UA/SCTP/IP. The structure of the protocol stack at the BSC side is BSSAP/SCCP/
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.
MSC Server
HLR
STP
CE1 CE2
TDM
IP
BSC
transfer point. The structure of the protocol stack at the MSC server side is BSSAP/SCCP/
M3UA/SCTP/IP. The structure of the protocol stack at the BSC side is BSSAP/SCCP/
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
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.
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 Signaling point name
l Signaling point code
l Office direction name
l IP address
l SCTP port
Figure 4-6 Flow of the data configuration for the multi-MGW VoIP networking
Start
End
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.
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
Rema When the codec UMTS AMR/AMR2 is supported, set Support codec rate
rks configuration in Special attributes.
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.
Rem The preceding is the default configuration, which can be modified according
arks to the actual networking.
Desc Add a local office signaling point code where National network code is
riptio A0002.
n
Desc Add an M3UA local entity data record where Local entity name is MSXc,
riptio Signaling point code of local entity is A00021, and Local entity type is
n Application server.
Desc Add an M3UA destination entity data record where Destination entity name
riptio is MSX02, indicating the MSC2.
n
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.
Desc Add an M3UA link set data record where Linkset name is MSX02, and
riptio Adjacent entity name is MSX02.
n
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 two M3UA link data records. For one link, set Client/Server to Client,
riptio WBSG module number to 141, Link name to SX02-1, Local port number
n to 7004, and Peer port number to 7004; for the other link, set Client/Server
to Client, WBSG module number to 142, Link name to SX02-2, Local port
number to 7005, and Peer port number to 7005.
Desc Add an M3UA route data record. The name of the destination entity of the
riptio MSX02 route is MSC2.
n
Desc Add an office direction data record where Office direction name is MSX02,
riptio Signaling type is M3UA, BICC call source name is CALL_SRC1, and
n Multi-area name is 1.
Rem BICC call source name and Multi-area name should be same to original
arks network.
Desc Add a route analysis data record where Route selection name is RSN2, and
riptio Signaling priority is Prefer to select BICC IP.
n
Rem Usually, set Caller category to All, and Address information indicator and
arks Transmission capability of the command to All categories.
Desc Add a BICC trunk group data record where Trunk group name is MSX02,
riptio and MGW name is MGW1.
n
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.
Step 3 Modify the M2UA signalling to M3UA signalling of national reserved network between MSC
Server and MGW1.
1. Add an M3UA local entity data record.
Desc Add an M3UA local entity data record where Local entity name is MSXd,
riptio Signaling point code of local entity is 001, and Local entity type is
n Application server.
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 MGW1. Here, it is null.
Desc Add an M3UA link set data record where Linkset name is BSC1, and Adjacent
riptio entity name is MGW1.
n
Desc Add two M3UA link data records. For one link, set Client/Server to Client,
riptio WBSG module number to 141, Link name to BSC1_01, Local port
n number to 7006, and Peer port number to 7006; for the other link, set Client/
Server to Client, WBSG module number to 142, Link name to BSC1_02,
Local port number to 7007, and Peer port number to 7007.
Desc Add an M3UA route data record. The name of the destination entity of the
riptio UMG8900 (SG) route is MGW1.
n
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.
Desc Add an M3UA destination entity data record where Destination entity name
riptio is BSC1, indicating the BSC.
n
Desc Add an M3UA route data record. The name of the destination entity of BSC1
riptio route is BSC1.
n
----End
Procedure
Step 1 Configure data for IP bearer network on MGW.
1. Add an IP 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
Step 2 Configure M3UA data for interworking between the MSC Server and MGW1.MSOFTX3000.
Configure M3UA data that is used for the national reserved network.
Desc Add an M3UA local entity data record where Local Entity Index is 1, Local
riptio Entity Name is MGW1-B, Source signaling Point Code is 101, and Local
n Entity Type is signaling_Gateway.
Desc Add an M3UA destination entity data record. Destination Entity Index is set
riptio to 1, and Destination Entity Name is set to MSXd, indicating the
n MSOFTX3000 (AS) that is directly connected to the UMG8900. The local
entity index corresponding to the destination entity index is 1.
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.
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
Desc Add an M3UA route data record. The index of the destination entity of the
riptio MSOFTX3000 (AS) is 1.
n
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.
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
Original Network
Figure 4-7 shows the typical TDM networking.
TMSC
MSC MSC
Server1 Server2
TDM
BSC IP MSC
Destination Network
Figure 4-8 shows the typical CMN networking.
CMN
MSC MSC
Server1 Server2
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
with MSC server 1 by using the M3UA direct connection mode. The structure of the
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
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.
The BICC 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 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
the new BICC signaling between the two NEs.
l Maintain the ISUP signaling between the TMSC and MSC server 2. Configure the data for
the new BICC signaling between the two NEs.
Interworking Data
In the above evolution methods, you must plan and collect the following data:
l Signaling point name
l Signaling point name
l Office direction name
l IP address
l SCTP port number
4.3.1 Configuration Flow
This section introduces the flow of the data configuration in the CMN VoIP networking mode.
4.3.2 Interworking Data
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.
Figure 4-9 Flow of the data configuration for the CMN networking mode
Start
End
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
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.
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
Procedure
Step 1 Configure BICC signalling data for interworking with MSC1.
1. Add a local office signaling point code.
Desc Add a local office signaling point code where National network code is
riptio A0002.
n
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.
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 two M3UA link data records. For one link, set Client/Server to Client,
riptio WBSG module number to 141, Link name to MSX01_01, Local port
n number to 7000, and Peer port number to 70040; for the other link, set Client/
Server to Client, WBSG module number to 142, Link name to
MSX01_02, Local port number to 7001, and Peer port number to 7001.
Desc Add an M3UA route data record. The name of the destination entity of the
riptio MSX01 route is MSC1.
n
Desc Add an office direction data record where Office direction name is MSX01,
riptio Signaling type is M3UA, BICC call source name is CALL_SRC1, and
n Multi-area name is 1.
Rem BICC call source name and Multi-area name should be same to original
arks network.
Desc Add a route analysis data record where Route selection name is RSN1, and
riptio Signaling priority is Prefer to select BICC IP.
n
Desc Add a BICC trunk group data record where Trunk group name is MSX01,
riptio and MGW name is CMN.
n
Desc Add a BICC CIC module data record where Office direction name is
riptio MSX01.
n
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.
Desc Add an M3UA destination entity data record where Destination entity name
riptio is MSX02, indicating the MSC2.
n
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.
Desc Add an M3UA link set data record where Linkset name is MSX02, and
riptio Adjacent entity name is MSX02.
n
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 two M3UA link data records. For one link, set Client/Server to Client,
riptio WBSG module number to 141, Link name to SX02-1, Local port number
n to 7004, and Peer port number to 7004; for the other link, set Client/Server
to Client, WBSG module number to 142, Link name to SX02-2, Local port
number to 7005, and Peer port number to 7005.
Desc Add an M3UA route data record. The name of the destination entity of the
riptio MSX02 route is MSC2.
n
Desc Add an office direction data record where Office direction name is MSX02,
riptio Signaling type is M3UA, BICC call source name is CALL_SRC1, and
n Multi-area name is 2.
Rem BICC call source name and Multi-area name should be same to original
arks network.
Desc Add a route analysis data record where Route selection name is RSN2, and
riptio Signaling priority is Prefer to select BICC IP.
n
Desc Add a BICC trunk group data record where Trunk group name is MSX02,
riptio and MGW name is CMN.
n
Desc Add a BICC CIC module data record where Office direction name is
riptio MSX02.
n
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.
----End
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.
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.
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
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.
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.
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.
The performance measurement entities are as follows:
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.
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.
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.
The performance measurement entities are as follows:
Interface Received Flow, Interface Sent Flow, Use Rate of Interface Sent Bandwidth, and Use
Rate of Interface Received Bandwidth
The involved formulas are as follows:
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.
The performance measurement entities are as follows:
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
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.
The performance measurement entities are as follows:
IP Send Establish Request Times, IP Receive Establish Confirm Times, IP Receive Establish
Request Times, and IP Send Establish Confirm Times.
6 VoIP Troubleshooting
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.
Start
Yes
End
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
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.
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.
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.
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.
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
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.
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.
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.
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.
l If the call setup is successful, go to the end.
l If the call setup fails, go to Step 1.6.
6. Deactivate and activate the SCTP link by using the DEA BSCTP and ACT BSCTP
command. 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.
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 is successful, go to the end.
l If the call setup fails, go to Step 2.
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.
l If it is correct, go to Step 8.
l If it is incorrect, go to Step 7.
Step 7 Modify the configuration data. 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 8.
Step 8 Check whether the data in the route analysis table is correct by using LST RTANA.
l If it is correct, go to Step 10.
l If it is incorrect, go to Step 9.
Step 9 Modify the configuration data. 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 10.
Step 10 Check the configuration of the route, sub-route and gateway of the BICC trunk group in the
trunk group table.
l If it is correct, go to Step 12.
l If it is incorrect, go to Step 11.
Step 11 Modify the configuration data. 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 12.
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 14 Modify the configuration data. 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 27.
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.
l If the call setup is successful, go to the end.
l If the call setup fails, go to Step 18.
Step 18 Check whether the configuration of the trunk resources of the UMG8900 is correct.
l If it is incorrect, go to Step 19.
l If it is correct, go to Step 27.
Step 19 Modify the configuration data. 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 27.
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).
l If not, go to Step 22.
Step 22 Delete the codec type other than the ones listed previously. 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 27.
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.
l If it is correct, go to Step 27.
l If it is incorrect, go to Step 26.
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.
l If it is correct, go to Step 27.
l If it is incorrect, go to Step 26.
Step 26 Modify the configuration data. 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 27.
Step 27 Check whether the back board of the HRU board of the UMG8900 is an E8T, an E1G or a PAL.
l If yes, go to Step 29.
l If not, go to Step 28.
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.
l If the call setup is successful, go to the end.
l If the call setup fails, go to Step 29.
Step 29 Check whether the bearer bandwidth of the interface is zero by using LST IFBW.
l If not, go to Step 31.
l If yes, go to Step 30.
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.
Step 31 Check whether the bearer bandwidth of the interface or the VLAN is configured.
l If yes, go to Step 33.
Step 32 Configure the bearer bandwidth of the interface or the VLAN. 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 33.
Step 33 Check whether the port for none-bearer purpose is reserved board of the specified IP address by
using LST RSVPORT.
l If yes, go to Step 34.
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 is successful, go to the end.
l If the call setup fails, contact Huawei's technical support.
----End
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.
3. Ping the bearer IP address of the gateway of UMG8900 1 on UMG8900 2.
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.
Step 5 If the RTP traffic of the IP port is not normal, handle the related fault. 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 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 removed, go to the end.
l If the fault is not removed, contact Huawei's technical support.
----End
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 1 on UMG8900 2.
3. Ping the bearer IP address of the gateway of UMG8900 1 on UMG8900 2.
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.
l If the fault is removed, go to the end.
l If the fault is not removed, go to Step 4.
Step 4 Check whether voice codec uses G.711 by using LST MG.
l If yes, go to Step 5.
l If not, go to Step 6.
Step 5 Disable the VAD function. 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 6 Set Special attributes to Not support echo cancellation by using MOD MGW. 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 7.
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
This section describes the principles and data configuration of the VoIP bearer network.
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.
POS1/0/0 11.4.1.2/30
PE1 PE2
POS1/0/0 11.4.1.1/30
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
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
[CE1] interface gigabitethernet 6/0/0
[CE1-GigabitEthernet6/0/0] undo shutdown
[CE1-GigabitEthernet6/0/0] portswitch
[CE1-GigabitEthernet6/0/0] 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
# Configure VLANIF interface.
[CE1] interface vlanif 3
3. Assign VLAN5 and VLAN6 for connecting signal stream interface and media stream
interface of PE1 on CE1.
# Create VLAN5.
[CE1] vlan 5
[CE1-Vlan5] quit
# Configure VLANIF interface.
[CE1] interface vlanif 5
[CE1-Vlanif5] undo shutdown
[CE1-Vlanif5] ip address 10.2.6.1 30
[CE1-Vlanif5] quit
# Create VLAN6.
[CE1] vlan 6
[CE1-Vlan6] quit
# Configure VLANIF interface.
[CE1] interface vlanif 6
[CE1-Vlanif6] undo shutdown
[CE1-Vlanif6] ip address 10.2.7.1 30
[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
[CE2] interface gigabitethernet 5/0/0
[CE2-GigabitEthernet5/0/0] undo shutdown
[CE2-GigabitEthernet5/0/0] portswitch
[CE2-GigabitEthernet5/0/0] quit
[CE2] interface gigabitethernet 6/0/0
[CE2-GigabitEthernet6/0/0] undo shutdown
[CE2-GigabitEthernet6/0/0] portswitch
[CE2-GigabitEthernet6/0/0] quit
# 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
# Configure VLANIF interface.
[CE2] interface vlanif 2
[CE2-Vlanif2] undo shutdown
[CE2-Vlanif2] ip address 10.2.3.2 30
[CE2-Vlanif2] quit
2. Assign VLAN3 and VLAN4 for connecting signal stream interface and media stream
interface of CE1 on CE2.
# Create VLAN3.
[CE2] vlan 3
[CE1-Vlan3] quit
# Configure VLANIF interface.
[CE2] interface vlanif 3
[CE2-Vlanif3] undo shutdown
[CE2-Vlanif3] ip address 10.2.4.2 30
[CE2-Vlanif3] quit
# Create VLAN4.
[CE2] vlan 4
[CE2-Vlan4] quit
# Configure VLANIF interface.
3. Assign VLAN5 and VLAN6 for connecting signal stream interface and media stream
interface of PE2 on CE2.
# Create VLAN5.
[CE2] vlan 5
[CE2-Vlan5] quit
# Configure VLANIF interface.
[CE2] interface vlanif 5
[CE2-Vlanif5] undo shutdown
[CE2-Vlanif5] ip address 10.2.8.1 30
[CE2-Vlanif5] quit
# Create VLAN6.
[CE2] vlan 6
[CE2-Vlan6] quit
# Configure VLANIF interface.
[CE2] interface vlanif 6
[CE2-Vlanif6] undo shutdown
[CE2-Vlanif6] ip address 10.2.9.1 30
[CE2-Vlanif6] quit
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
# Configure VLANIF interface.
[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
# Configure VLANIF interface.
[PE1] interface vlanif 6
[PE1-Vlanif6] undo shutdown
[PE1-Vlanif6] ip address 10.2.7.2 30
[PE1-Vlanif6] quit
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
# Configure VLANIF interface.
[PE2] interface vlanif 5
[PE2-Vlanif5] undo shutdown
[PE2-Vlanif5] ip address 10.2.8.2 30
[PE2-Vlanif5] quit
# Create VLAN6.
[PE2] vlan 6
[PE2-Vlan6] quit
# Configure VLANIF interface.
[PE2] interface vlanif 6
[PE2-Vlanif6] undo shutdown
[PE2-Vlanif6] ip address 10.2.9.2 30
[PE2-Vlanif6] 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] interface gigabitethernet 3/0/0
[CE1-GigabitEthernet3/0/0] ip binding vpn-instance signal
[CE1-GigabitEthernet3/0/0] ip address 11.1.2.1 30
[CE1-GigabitEthernet3/0/0] quit
[CE1] interface gigabitethernet 4/0/0
[CE1-GigabitEthernet4/0/0] ip binding vpn-instance media
[CE1-GigabitEthernet4/0/0] ip address 10.1.1.1 30
[CE1-GigabitEthernet4/0/0] quit
[CE1] interface vlanif 2
[CE1-Vlanif1] ip binding vpn-instance signal
[CE1-Vlanif1] ip address 10.2.3.1 30
[CE1-Vlanif1] quit
[CE1] interface vlanif 3
[CE1-Vlanif2] ip binding vpn-instance signal
[CE1-Vlanif2] ip address 10.2.4.1 30
[CE1-Vlanif2] quit
[CE1] interface vlanif 4
[CE1-Vlanif3] ip binding vpn-instance media
[CE1-Vlanif3] ip address 10.2.5.1 30
[CE1-Vlanif3] quit
[CE1] interface vlanif 5
[CE1-Vlanif4] ip binding vpn-instance signal
[CE1-Vlanif4] ip address 10.2.6.1 30
[CE1-Vlanif4] quit
[CE1] interface vlanif 6
[CE1-Vlanif5] ip binding vpn-instance media
[CE1-Vlanif5] ip address 10.2.7.1 30
[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] interface gigabitethernet 3/0/0
[CE2-GigabitEthernet3/0/0] ip binding vpn-instance signal
[CE2-GigabitEthernet3/0/0] ip address 11.2.2.1 30
[CE2-GigabitEthernet3/0/0] quit
[CE2] interface gigabitethernet 4/0/0
[CE2-GigabitEthernet4/0/0] ip binding vpn-instance media
[CE2-GigabitEthernet4/0/0] ip address 10.2.1.1 30
[CE2-GigabitEthernet4/0/0] quit
[CE2] interface vlanif 2
[CE2-Vlanif1] ip binding vpn-instance signal
[CE2-Vlanif1] ip address 10.2.3.2 30
[CE2-Vlanif1] quit
[CE2] interface vlanif 3
[CE2-Vlanif2] ip binding vpn-instance signal
[CE2-Vlanif2] ip address 10.2.4.2 30
[CE2-Vlanif2] quit
[CE2] interface vlanif 4
[CE2-Vlanif3] ip binding vpn-instance media
[CE2-Vlanif3] ip address 10.2.5.2 30
[CE2-Vlanif3] quit
[CE2] interface vlanif 5
[CE2-Vlanif4] ip binding vpn-instance signal
[CE2-Vlanif4] ip address 10.2.8.1 30
[CE2-Vlanif4] quit
[CE2] interface vlanif 6
[CE2-Vlanif5] ip binding vpn-instance media
[CE2-Vlanif5] ip address 10.2.9.1 30
[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] interface vlanif 5
[PE1-Vlanif4] ip binding vpn-instance signal
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] network 10.2.4.0 0.0.0.3
[CE1-ospf-1-area-0.0.0.0] network 10.2.6.0 0.0.0.3
[CE1-ospf-1-area-0.0.0.0] network 11.1.2.0 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
[CE1-ospf-2] silent-interface gigabitethernet 4/0/0
[CE1-ospf-2] spf-schedule-interval millisecond 100
[CE1-ospf-2] lsa-originate-interval 0
[CE1-ospf-2] lsa-arrival-interval 0
[CE1-ospf-2] bandwidth-reference 10000
[CE1-ospf-2] vpn-instance-capability simple
[CE1-ospf-2-area-0.0.0.0] network 10.2.5.0 0.0.0.3
[CE1-ospf-2-area-0.0.0.0] network 10.2.7.0 0.0.0.3
[CE1-ospf-2-area-0.0.0.0] network 11.1.1.0 0.0.0.3
# 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] silent-interface gigabitethernet 3/0/0
[CE2-ospf-1] spf-schedule-interval millisecond 100
[CE2-ospf-1] lsa-originate-interval 0
[CE2-ospf-1] lsa-arrival-interval 0
[CE2-ospf-1] bandwidth-reference 10000
[CE2-ospf-1] vpn-instance-capability simple
[CE2-ospf-1-area-0.0.0.0] network 10.2.3.0 0.0.0.3
[CE2-ospf-1-area-0.0.0.0] network 10.2.4.0 0.0.0.3
[CE2-ospf-1-area-0.0.0.0] network 10.2.8.0 0.0.0.3
[CE2-ospf-1-area-0.0.0.0] network 11.2.2.0 0.0.0.3
[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
[CE2-ospf-2] silent-interface gigabitethernet 4/0/0
[CE2-ospf-2] spf-schedule-interval millisecond 100
[CE2-ospf-2] lsa-originate-interval 0
[CE2-ospf-2] lsa-arrival-interval 0
[CE2-ospf-2] bandwidth-reference 10000
[CE2-ospf-2] vpn-instance-capability simple
[CE2-ospf-2-area-0.0.0.0] network 10.2.5.0 0.0.0.3
[CE2-ospf-2-area-0.0.0.0] network 10.2.9.0 0.0.0.3
[CE2-ospf-2-area-0.0.0.0] network 11.2.1.0 0.0.0.3
# 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] network 10.2.6.0 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
[PE1-ospf-2] lsa-originate-interval 0
[PE1-ospf-2] lsa-arrival-interval 0
[PE1-ospf-2] bandwidth-reference 10000
[PE1-ospf-2] vpn-instance-capability simple
[PE1-ospf-2-area-0.0.0.0] network 11.4.1.0 0.0.0.3
[PE1-ospf-2-area-0.0.0.0] network 10.2.7.0 0.0.0.3
# Configure PE2.
[PE2]ospf 1 router-id 1.1.4.1 vpn-instance signal
[PE2-ospf-1]area 0
[PE2-ospf-1] spf-schedule-interval millisecond 100
[PE2-ospf-1] lsa-originate-interval 0
[PE2-ospf-1] lsa-arrival-interval 0
[PE2-ospf-1] bandwidth-reference 10000
[PE2-ospf-1] vpn-instance-capability simple
[PE2-ospf-1-area-0.0.0.0] network 11.4.1.0 0.0.0.3
[PE2-ospf-1-area-0.0.0.0] network 10.2.8.0 0.0.0.3
[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
[PE2-ospf-2] spf-schedule-interval millisecond 100
[PE2-ospf-2] lsa-originate-interval 0
[PE2-ospf-2] lsa-arrival-interval 0
[PE2-ospf-2] bandwidth-reference 10000
[PE2-ospf-2] vpn-instance-capability simple
[PE2-ospf-2-area-0.0.0.0] network 11.4.1.0 0.0.0.3
[PE2-ospf-2-area-0.0.0.0] network 10.2.9.0 0.0.0.3
----End
silent-interface GigabitEthernet4/0/0
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 10.2.5.0 0.0.0.3
network 10.2.7.0 0.0.0.3
network 11.1.1.0 0.0.0.3
#
return
portswitch
port default vlan 2
#
interface GigabitEthernet6/0/0
portswitch
port default vlan 2
#
ospf 1 router-id 1.1.2.1 vpn-instance signal
silent-interface Vlanif1
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 10.2.8.0 0.0.0.3
network 11.2.2.0 0.0.0.3
#
ospf 2 router-id 1.1.2.2 vpn-instance media
silent-interface GigabitEthernet4/0/0
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 10.2.5.0 0.0.0.3
network 10.2.9.0 0.0.0.3
network 11.2.1.0 0.0.0.3
#
return
bandwidth-reference 10000
vpn-instance-capability simple
area 0.0.0.0
network 11.4.1.0 0.0.0.3
network 10.2.6.0 0.0.0.3
#
ospf 2 router-id 1.1.3.2 vpn-instance media
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 11.4.1.0 0.0.0.3
network 10.2.7.0 0.0.0.3
#
return
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.
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.
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
Subscriber A calls subscriber B.
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.
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
Subscriber A calls subscriber B.
Expected Result
The call is connected normally. The voice is clear.
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.
8.2.1 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.
8.2.8 Checking System Operation When CE2 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
Check whether the link switchover is normal and service operation is normal when the link
between the CE2 and AR2 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 The call is connected normally. The voice is clear.
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.
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 The call is connected normally. The voice is clear.
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.
Prerequisites
l When the network works normally, disconnect the signaling link between the MGW and
CE1.
l The SCTP multi-homing function is enabled at Nc interface and 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, re-establish the signaling link between the MGW and CE1.
Expected Result
l The call is connected normally. The voice is clear.
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.
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 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 The call is connected normally. The voice is clear.
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.
Prerequisites
l When the network works normally, disconnect the media link between MGW and CE1.
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, reconnect the media link between the MGW and CE1.
Expected Result
l The call is connected normally. The voice is clear.
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.
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 MGW server and CE2.
Expected Result
The call is connected normally. The voice is clear.
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, 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
association is not interrupted.
CE2 is the active gateway of Nb interface. The media link of MGW A is the standby
media link.
The call is connected normally. The voice is clear.
l After CE1 is restarted,
The SCTP multi-homing switchover is performed at Nc and Mc interfaces and the SCTP
association is not interrupted.
CE1 is the active gateway of Nb interface. The media link of MGW A is the active
media link.
Prerequisites
l The network works normally.
Procedure
1. The subscriber in MSC server A calls the subscriber in MSC server B.
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.
The call is connected normally. The voice is clear.
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
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 CE1 and CE2.
Expected Result
The call is connected normally. The voice is clear.
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.
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 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
association is not interrupted.
The media link of MGW A is switched over to the standby media link.
The call is connected normally. The voice is clear.
l After the link between CE1 and AR1 is reconnected,
The SCTP multi-homing switchover is performed at Nc and Mc interfaces and the SCTP
association is not interrupted.
The media link of MGW A is switched over to the active media link.
8.2.11 Checking System Operation When the Link Between CE2 and
AR2 Is Faulty
Check whether the link switchover is normal and service operation is normal when the link
between the CE2 and AR2 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 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.
The call is connected normally. The voice is clear.
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.
9 FAQ
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
performs flow control to subsequent calls. In this way, the voice quality of the ongoing calls is
not affected by QoS loss.
Index
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