configuraticonfig voipon guide - voice(v200r002c01_01)

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Huawei AR2200 Series Enterprise Routers

V200R002C01

Configuration Guide - Voice

Issue 01

Date 2012-04-20

HUAWEI TECHNOLOGIES CO., LTD.


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Copyright © Huawei Technologies Co., Ltd. 2012. All rights reserved.

No part of this document may be reproduced or transmitted in any form or by any means without prior written

consent of Huawei Technologies Co., Ltd.

Trademarks and Permissions

and other Huawei trademarks are trademarks of Huawei Technologies Co., Ltd.

All other trademarks and trade names mentioned in this document are the property of their respective holders.

Notice

The purchased products, services and features are stipulated by the contract made between Huawei and the

customer. All or part of the products, services and features described in this document may not be within the

purchase scope or the usage scope. Unless otherwise specified in the contract, all statements, information,and recommendations in this document are provided "AS IS" without warranties, guarantees or representations

of any kind, either express or implied.

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 all statements, information, and

recommendations in this document do not constitute the warranty of any kind, express or implied.

Huawei Technologies Co., Ltd.

Address: Huawei Industrial Base

Bantian, Longgang

Shenzhen 518129

People's Republic of China

Website: http://www.huawei.com

Email: [email protected]

Issue 01 (2012-04-20) Huawei Proprietary and Confidential

Copyright © Huawei Technologies Co., Ltd.

i

http://www.huawei.com/


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About This Document

Intended Audience

This document provides the basic concepts, configuration procedures, and configuration

examples in different application scenarios of the voice feature.

This document describes how to configure the voice feature.

This document is intended for:

l Data configuration engineers

l Commissioning engineers

l Network monitoring engineers

l System maintenance engineers

Symbol Conventions

The symbols that may be found in this document are defined as follows.

Symbol Description

DANGER

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

avoided, will result in death or serious injury.

WARNING

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

if not avoided, could result in minor or moderate injury.

CAUTION

Indicates a potentially hazardous situation, which if not

avoided, could result in equipment damage, data loss,

performance degradation, or unexpected results.

TIP Indicates a tip that may help you solve a problem or save

time.

NOTE Provides additional information to emphasize or supplement

important points of the main text.


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Command Conventions

The command conventions that may be found in this document are defined as follows.

Convention Description

Boldface The keywords of a command line are in boldface.

Italic Command arguments are in italics.

[ ] Items (keywords or arguments) in brackets [ ] are optional.

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

bars. One item is selected.

[ x | y | ... ] Optional items are grouped in brackets and separated by vertical

bars. One item is selected or no item is selected.

{ x | y | ... }* Optional items are grouped in braces and separated by vertical bars. A minimum of one item or a maximum of all items can be

selected.

[ x | y | ... ]* Optional items are grouped in brackets and separated by vertical

bars. Several items or no item can be selected.

& The parameter before the & sign can be repeated 1 to n times.

# A line starting with the # sign is comments.

Interface Numbering Conventions

Interface numbers used in this manual are examples. In device configuration, use the existing

interface numbers on devices.

Change History

Changes between document issues are cumulative. Therefore, the latest document issue contains

all updates made in previous issues.

Changes in Issue 01 (2012-04-20)

Initial commercial release.


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Contents

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

1 SIPAG Configuration...................................................................................................................1

1.1 SIP AG Overview...............................................................................................................................................2

1.2 SIP AG Features Supported by the AR2200......................................................................................................21.3 Configuring the AR2200 to Work in SIP AG Mode..........................................................................................6

1.4 Configuring a SIP AG Interface.........................................................................................................................7

1.4.1 Establishing the Configuration Task.........................................................................................................7

1.4.2 Configuring Media and Signaling IP Address Pools.................................................................................8

1.4.3 Setting Parameters on the SIP AG Interface.............................................................................................8

1.4.4 Checking the Configuration.....................................................................................................................15

1.5 (Optional) Setting Parameters for Voice Interfaces..........................................................................................16

1.5.1 Setting Parameters for an FXS Interface.................................................................................................16

1.5.2 Setting parameters for a BRA Interface...................................................................................................18

1.5.3 Setting Parameters for a PRA Interface...................................................................................................19

1.6 (Optional) Setting Voice Parameters on a SIP AG...........................................................................................21

1.6.1 Setting System Parameters......................................................................................................................21

1.6.2 Setting SIP Stack Parameters..................................................................................................................30

1.6.3 Setting DSP Parameters...........................................................................................................................32

1.7 Configuring a SIPAG User...............................................................................................................................43

1.7.1 Establishing the Configuration Task.......................................................................................................44

1.7.2 Setting Parameters for a SIP AG User.....................................................................................................44

1.7.3 Configuring a SIP AG User Group..........................................................................................................46

1.7.4 (Optional) Configuring Functions or Services in a SIP Service Data Profile.........................................47

1.7.5 Checking the Configuration.....................................................................................................................49

1.8 (Optional) Configuring the BEST Service.......................................................................................................50

1.9 Resetting a SIP AG...........................................................................................................................................52

1.10 Maintaining a SIP AG....................................................................................................................................53

1.10.1 Clearing SIP AG Statistics....................................................................................................................53

1.11 Configuration Examples.................................................................................................................................53

1.11.1 Example for Configuring a SIP AG......................................................................................................53

1.11.2 Using a PRA Trunk to Connect the SIP AG to a PBX..........................................................................57

1.11.3 Example for Configuring the BEST Service.........................................................................................61

2 PBX Configuration......................................................................................................................66


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2.1 PBX Overview..................................................................................................................................................68

2.2 PBX Features Supported by the AR2200.........................................................................................................69

2.3 PBX Configuration Procedures........................................................................................................................74

2.4 (Optional) Setting Parameters for Voice Interfaces..........................................................................................78

2.4.1 Setting Parameters for an FXS Interface.................................................................................................78

2.4.2 Setting Parameters for an FXO Interface.................................................................................................80

2.4.3 Setting parameters for a BRA Interface...................................................................................................81

2.4.4 Setting Parameters for a PRA Interface...................................................................................................83

2.5 Setting PBX Parameters...................................................................................................................................84

2.5.1 Configuring the AR2200 to Work in PBX Mode....................................................................................84

2.5.2 Configuring Media and Signaling IP Address Pools...............................................................................85

2.5.3 (Optional) Setting System Parameters.....................................................................................................86

2.5.4 (Optional) Setting SIP Stack Parameters.................................................................................................95

2.5.5 (Optional) Setting DSP Parameters.........................................................................................................98

2.6 Configuring an Enterprise..............................................................................................................................108

2.7 Configuring a Call Prefix...............................................................................................................................110

2.8 Configuring a PBX User.................................................................................................................................112

2.9 (Optional) Configuring a SIP Server..............................................................................................................113

2.10 (Optional)Configuring SBC Proxy Function................................................................................................115

2.10.1 Configuring the SBC Proxy Function (Public IP Address for the SIP Server and Private IP Addresses

for SIPUEs).....................................................................................................................................................116

2.10.2 Configuring the SBC Proxy Function (Private IP Address for the SIP Server and Public IP Addresses

for SIP UEs)....................................................................................................................................................118

2.10.3 Configuring the SBC Proxy Function (Private IP Address for the SIP Server and Private IP Addresses

for SIP UEs)....................................................................................................................................................121

2.11 (Optional)Configuring a CDR Server...........................................................................................................124

2.12 Configuring a Trunk Group..........................................................................................................................126

2.12.1 Establishing the Configuration Task...................................................................................................126

2.12.2 Configuring a Trunk Group.................................................................................................................127

2.12.3 Binding Trunks in a Trunk Group.......................................................................................................139

2.12.4 Resetting a SIP Trunk Group...............................................................................................................141

2.12.5 Checking the Configuration.................................................................................................................142

2.13 Configur ing a Call Route..............................................................................................................................1422.14 (Optional)Configuring Individualized PBX Services...................................................................................144

2.14.1 Configuring the Abbreviated Dialing Service.....................................................................................144

2.14.2 Configuring the Call-out Restriction Service......................................................................................145

2.14.3 Configuring the Call Forwarding Service...........................................................................................147

2.14.4 Configuring the Number Barring Service...........................................................................................148

2.14.5 Configuring the DND Service.............................................................................................................149

2.14.6 Configuring the RAC Service..............................................................................................................151

2.14.7 Configuring the Remote Office Service..............................................................................................152

2.14.8 Configuring the Secretary Service.......................................................................................................153

2.14.9 Configuring the Wake-up Service.......................................................................................................154





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2.14.10 Configuring the RBT Service............................................................................................................155

2.14.11 Configuring the SCR Service............................................................................................................157

2.14.12 Configuring the SCA Service............................................................................................................158

2.15 (Optional) Configuring Group Services on the PBX....................................................................................159

2.15.1 Configuring the Call Interception Service...........................................................................................159

2.15.2 Configuring the Distinctive Ringing Service......................................................................................161

2.15.3 Configuring the Enterprise RBT Service.............................................................................................162

2.15.4 Configure the IVR Service..................................................................................................................164

2.15.5 Configuring the Number Change Service...........................................................................................170

2.15.6 Configuring the Pre-routing Number Change Service........................................................................171

2.15.7 Configuring the Post-routing Number Change Service.......................................................................173

2.15.8 Configuring the PBX Line Selection Service......................................................................................175

2.15.9 Configuring the Co-Group Pickup Service.........................................................................................176

2.15.10 Configuring Simultaneous Ringing...................................................................................................177

2.15.11 Configuring Sequential Ringing........................................................................................................179

2.15.12 Configuring the One Number Link You Service...............................................................................180

2.15.13 Configuring the Blacklist or Whitelist Function...............................................................................181

2.15.14 Configuring the DISA Service..........................................................................................................183

2.15.15 Configuring the Immediate Conference Service...............................................................................186

2.16 Configuration Examples...............................................................................................................................188

2.16.1 Example for Configuring Voice Services for a Small- or Medium-sized Enterprise..........................188

2.16.2 Example for Configuring the AR to Implement Communication for Different Enterprises...............193

2.16.3 Example for Using an H.323 Trunk Group in Peer Mode to Configure Calls Between the Headquartersand Branch......................................................................................................................................................199

2.16.4 Example for Using an H.323 Trunk Group in Registration Mode to Configure Calls Between the

Headquarters and Branch................................................................................................................................208

2.16.5 Example for Using the SIP Trunk Group to Configure Calls Between the Headquarters and Branch

........................................................................................................................................................................213





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1 SIPAG ConfigurationAbout This Chapter

This chapter describes the concepts and configuration of SIPAG, and provides a configuration

example.

1.1 SIP AG Overview

This section describes the SIP AG in VoIP applications.

1.2 SIP AG Features Supported by the AR2200

This section describes SIP AG features supported by the AR2200.

1.3 Configuring the AR2200 to Work in SIP AG ModeBefore configuring SIP AG service features, configure the AR2200 to work in SIP AG mode.

1.4 Configuring a SIP AG Interface

This section describes how to configure a SIP AG interface.

1.5 (Optional) Setting Parameters for Voice Interfaces

This section describes how to set parameters for voice interfaces.

1.6 (Optional) Setting Voice Parameters on a SIP AG

This section describes how to set voice parameters on a SIP AG.

1.7 Configuring a SIPAG User

This section describes how to configure a SIPAG user.

1.8 (Optional) Configuring the BEST Service

The branch exchange for survivable telephony (BEST) service enables SIP AGs in enterprise

branches to manage local calls when communication with the headquarters voice server fails.

1.9 Resetting a SIP AG

This section describes how to reset a SIP AG.

1.10 Maintaining a SIP AG

This section describes how to maintain a SIP AG.

1.11 Configuration Examples

This section provides a SIPAG configuration example.


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1.1 SIP AG Overview

This section describes the SIP AG in VoIP applications.

VoIP

On the Public Switched Telephone (PSTN), telephone lines are occupied exclusively. The fees

for toll calls are high.

To reduce data and voice fees and meet service requirements, voice over IP (VoIP) is used.

VoIP transmits voice services over the IP network. In VoIP, a voice gateway encapsulates voice

signals into frames and transmits them as IP packets. Currently, IP phone gateways implement

communication between the PSTN network and IP network. As PC-to-phone, phone-to-PC, and

phone-to-phone technologies have developed, the voice quality is improved greatly. VoIP can

meet commercial requirements.

Voice Gateway

The IP network is a packet switched network. The transmission costs on the IP network are lower

than those on the PSTN; therefore, VoIP will take place traditional voice phones gradually.

Replacing all phone networks with VoIP will take high costs. The step-by-step migration

solution is recommended.

In most enterprise voice solutions, phone networks of branches are reserved and the IP network

is used between branches.

In the step-by-step migration solution, a voice gateway is required to connect the two types of networks. A router is often used as the voice gateway.

IMS

The IP Multimedia Core Network Subsystem (IMS) is an architectural framework for providing

IP multimedia services, including audio, video, text, and instant messages. It was designed by

the wireless standards body 3rd Generation Partnership Project (3GPP) in Release 5.

SIP

The Session Initiation Protocol (SIP) is a text-based signaling protocol. SIP messages are

classified into request and response messages. SIP can be used for creating, modifying, and

terminating two-party or multiparty sessions. SIP can be used for multimedia conferences,

remote education, and Internet calls.

SIPAG

A SIP access gateway (SIP AG) is a voice gateway that exchanges SIP signals with other devices

between the PSTN/ISDN and IP network. It can implement VoIP functions.

1.2 SIP AG Features Supported by the AR2200

This section describes SIP AG features supported by the AR2200.





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Using a SIP AG as a Voice Gateway

As shown in Figure 1-1, a SIP AG functions as an IP voice gateway to implement

communication between the IP network and the PSTN or ISDN. The usage scenarios are as

follows:

l A user connects to the SIP AG through the PSTN. The SIP AG converts analog signals into

digital signals and compresses and packetizes the digital signals so that the voice packets

can be transmitted on the IP network.

l Voice packets are transmitted to the SIP AG through the IP network. The IP voice gateway

of the called party converts the voice packets into analog voice signals and transmits them

to the terminal of the called party through the PSTN.

NOTE

l The AR2200 series routers support the voice features only after a DSP module is installed.

l To provide voice services for POTS users, 4FXS1FXO board is required.

l To provide voice services for ISDN users, 2BST board is required.

Figure 1-1 SIP AG as a voice gateway

SIPAG Eth1/0/0

IMS/IP

POTS Modem FAX

SIP AG call flow

The SIP AG call flow is as follows:

1. A user picks up the phone and the SIP AG detects offhook signals.

2. The DSP module of the SIP AG generates a dial tone.

3. The user hears the dial tone played by the session application of the SIP AG and starts to

dial up.

4. The session application of the SIP AG collects the number dialed by the user.

5. The session application of the SIP AG matches the number with preconfigured called

number profiles.





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6. When the voice gateway of the calling party successfully matches the number with a

preconfigured called number profile, the number is mapped to the voice gateway of the

called party.

7. The SIP AG initiates a call over the IMS and establishes a logical channel for each call.

Then the channel is used to send and receive voice data.8. The IMS searches for the destination phone and initiates a call.

9. After the called party picks up the phone, a call is set up. After the calling party or called

party hangs up the phone, the call is ended.

Voice Services Supported by the SIP AG

The SIP AG supports the basic voice service and developed services such as the three-party

service and call waiting service. Among these services, some voice services need to be

configured on the SIP AG.

Service Type Introduction Configured on theSIP AG orNot

Basic voice service The basic voice service is the basic call connection

function, including intra-office calls, local calls,

national toll calls, international toll calls, and transit

calls.

Yes

Three-party service The third-party service allows a calling party or called

party in a conversation to call a third party without

ending the current conversation. Then the calling party

or original called party can implement a three-partyconversation or talk to the other two parties.

Yes

Call waiting

service

When user A is talking with user B over the phone and

at this moment user C is calling user A, user A hears a

call waiting tone, indicating that there is a call waiting

for user A.

Yes

MWI service The message waiting indicator (MWI) service allows a

user to read unread messages or leave messages. When

the called user is busy, the MWI is on, indicating that

there are leave messages.

Yes

Malicious call

identification

(MCID) service

The user that registers the MCID service with the carrier

can query the phone number of the attacker that initiates

malicious calls after performing relevant operations.

Yes

Call transfer

service

The call transfer service allows the called party to

transfer an incoming call to a third party by pressing the

hookflash so that the calling party establishes a

connection with a new called party.

Yes

Call conference

service

The call conference service allows more than three

parties to communicate together.

Yes





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Calling line

identification

presentation

(CLIP) service

The CLIP service displays the calling number in onhook

state or offhook state (for call waiting). The displayed

information includes the phone number, name, date, and

time.

No

Calling line

identification

restriction (CLIR)

service

The CLIR service displays the phone number of a

calling party on the terminal of a called party.

No

Distinctive ringing

service

The distinctive ringing service plays different ring tones

for incoming calls.

No

Differentiated

ringback tone

service

The differentiated ringback tone service enables the SIP

AG to play different ring tones for ringback tones.

No

Advice of charge

(AoC) service

The AoC service enables the SIP AG to display the

charge rate, fee notification during a call, and the total

fee of the call.

No

Polarity reversal

charging service

The polarity reversal charging service notifies the

accounting terminal of the starting point and stop point

of accounting using polarity reversal when users are in

a conversation or end the conversation.

No

Polarity reversal

pulse charging

service

When users are in a conversation or end the

conversation, a polarity reversal pulse is generated on

an interface, notifying the accounting terminal of the

starting point and stop point of accounting.

No

Urgent call process If the SIP AG finds an urgent call, the SIP AG inserts

the urgent call flag into the SIP message.

No

Completion of

Calls to Busy

Subscriber (CCBS)

service

When the called number is busy, the CCBS service

enables the SIP AG to monitor the called party status.

When the called party is idle, the SIP AG notifies the

calling party and determines whether to make a callaccording to the status of the calling and called parties.

No

Multiple MSN

numbers on a

POTS interface

Multiple MSN numbers can be configured on a POTS

interface.

No

Hotline service l Instant hotline service: After a user picks up a phone,

the SIP AG dials the hotline number for the user.

l No dialing within a long time after picking up the

phone: If a user does not dial any number within the

specified period of time after picking up the phone,

the SIP AG dials the hotline number for the user.

No





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Anonymous call

service

The anonymous call service enables the called party not

to view information about incoming calls.

No

License Support

The SIPAG function is used with a license. To use the SIPAG function, apply for and purchase

the following license from the Huawei local office:

AR2200 value-added service package for voice services

To use the BEST function, you must purchase the CM&BEST License in addition to the precedng

licenses.

1.3 Configuring the AR2200 to Work in SIP AG Mode

Before configuring SIP AG service features, configure the AR2200 to work in SIP AG mode.

Applicable Environment

You can configure the AR2200 to work in SIP AG or PBX mode. Before configuring SIP AG

service features, configure the AR2200 to work in SIP AG mode. You can run the display voiceservice-mode command to view the working mode of the AR2200. If the AR2200 works in

PBX mode, delete the PBX configurations and configure the AR2200 to work in SIP AG mode.

If the AR2200 works in SIP AG mode, skip this configuration.

Pre-configuration Tasks

Before configuring the AR2200 to work in SIP AG mode, complete the following task:

l Configuring IP addresses for interfaces and routing protocols to ensure connectivity

Procedure

Step 1 Run:system-view

The system view is displayed.

Step 2 Run:voice

The voice view is displayed.

Step 3 Run:service-mode sipag

The AR2200 is configured to work in SIP AG mode.





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Step 4 Run:quit

Return to the system view.

Step 5 Run:quit

Return to the user view.

Step 6 Run:save

The configuration is saved.

Step 7 Run:reboot

The AR2200 is restarted.

NOTE

After the AR2200 is configured to work in SIP AG mode, restart the AR2200 to make the configuration

take effect.

----End

Checking the Configuration

After the configuration is complete, the AR2200 works in SIP AG mode.

Run the display voice service-mode command to check the working mode of the AR2200.

1.4 Configuring a SIP AG Interface

This section describes how to configure a SIP AG interface.

1.4.1 Establishing the Configuration Task

Before configuring a SIP AG interface, familiarize yourself with the applicable environment,

complete the pre-configuration tasks, and obtain the data required for the configuration. This

will help you complete the configuration task quickly and accurately.


To allow the SIP AG and IMS network to exchange media and signaling streams, the media and

signaling IP addresses, signaling port number, and transmission protocol need to be configured.


Before configuring a SIP AG interface, complete the following task:

l Configuring the AR2200 to work in SIP AG mode

Data Preparation

To configure a SIP AG interface, you need the following data.





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No. Data

1 Media and signaling IP addresses

2 SIP AG interface number, media and signaling IP addresses, signaling port number,

IP addresses and port numbers of primary and secondary proxy servers, transmission protocol, and home domain name

1.4.2 Configuring Media and Signaling IP Address Pools

Media and signaling IP address pools provide media and signaling IP addresses for SIP AG

interfaces.

Context

A SIP AG interface must obtain media and signaling IP addresses from media and signaling IP

address pools respectively. The signaling IP address pool stores IP addresses of SIP AG

interfaces and the media IP address pool stores IP addresses of media streams. Media and

signaling IP address pools can contain the same IP addresses.

Procedure



Step 2 Run:voice


Step 3 Run:voip-address media interface interface-type interface-number { ip-address |dynamic }

A media IP address pool is configured.

Step 4 Run:voip-address signalling interface interface-type interface-number { ip-address |dynamic }

A signaling IP address pool is configured.

When configuring media and signaling IP address pools:

l To specify the ip-address parameter, ensure that an IP address has been configured for the

specified interface.

l To specify the dynamic parameter, ensure that an IP address has been dynamically assigned

to the specified interface.

----End

1.4.3 Setting Parameters on the SIP AG Interface

This section describes how to set parameters on a SIP AG interface.





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Context

To allow the SIP AG and IMS network to exchange media and signaling streams, set the media

and signaling IP addresses, signaling port number, and transmission protocol.

Procedure

Step 1 Run:

system-view


Step 2 Run:

voice


Step 3 (Optional) Run:

local-digitmap name-value { append | type { normal | emergency | scc | direct-centrex | second-centrex } }body-value

The digitmap is configured.

Step 4 Run:

sipag sipag-interface-id

A SIP AG interface is created and the SIP AG interface view is displayed.

Step 5 Set mandatory parameters on the SIP AG interface.

Parameter Command Remarks

Media IP address media-addr { media-ip | addr-name

addr-name }

The media IP address

of the SIP AG interface

must be obtained from

the media IP address

pool.

Signaling IP address and

signaling port number

signalling-addr { signal-ip | addr-

name signal-addr-name } signal-port

The signaling IP

address of the SIP AG

interface must be

obtained from the

signaling IP address pool.

By default, no

signaling port number

is configured for a SIP

AG interface.

Paramet

ers for

the

primary

proxy

server

Static IP

address

primary-proxy-addr static static-ip-

address static-ip-address-port

By default, no static IP

address is configured

for the primary proxy

server.





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DNS-A

domain name

and port

number

primary-proxy-addr dns-a dns-a-

domain-name dns-a-port-number

By default, no DNS-A

domain name or port

number is configured

for the primary proxy

server.

DNS-NAPTR

domain name

primary-proxy-addr dns-naptr dns-

naptr-domain-name

By default, no DNS-

NAPTR domain name

is configured for the

primary proxy server.

DNS-SRV

domain name

primary-proxy-addr dns-srv dns-

srv-domain-name

By default, no DNS-

SRV domain name is

configured for the

primary proxy server.

Transmission protocol transfer transfer-protocol By default, a SIP AG

interface uses the User

Datagram Protocol

(UDP) protocol as the

transmission protocol.

Home domain name home-domain home-domain-value By default, no home

domain name is

configured for a SIP

AG interface.

Step 6 Set optional parameters on the SIP AG interface.


Description description description By default, no

description is


AG interface.

Profile index profile profile-index By default, the profile

index of a SIP AG

interface is 1.

Domain name ag-domain ag-domain-name By default, no domain

name is configured for

a SIP AG interface.

Conference factory URI conference-factory-uri uri By default, no

conference factory

URI is configured for a

SIP AG interface.

Authent

ication

Authenticatio

n mode

auth mode mode-value By default, the

authentication mode is

user.





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paramet

ers

Authenticatio

n user name

and password

auth username auth-username

password { cipher | ha1 { cipher |

simple password-value1 } | simple

password value2 }

By default, no

authentication user

name or password is


AG user.

Timer value digitmap-timer { long long-timer-

value | short short-timer-value | start

start-timer-value }

By default, the value of

a digitmap long timer is

20s, the value of a

digitmap short timer is

4s, and the value of a

digitmap start timer is

20s.

DTMF parameters dtmf-transmission-mode

{ thoroughly | erase }

By default, the DTMF

transmission mode istransparent

transmission.

Fax/

Modem

paramet

ers

Fax/Modem

codec

negotiation

mode

fax-modem common negotiation-

mode { negotiate | self-switch }

By default, the fax/

modem codec

negotiation mode is

negotiation.

Packetization

interval in

transparent

transmissionmode

fax-modem common rtp-interval

{ 5ms | 10ms | 20ms | 30ms }

By default, the

packetization interval

in transparent

transmission mode is10 ms.

VBD attribute

type

fax-modem common vbd-attribute-

type { chinatelecom |

extchinatelecom | ietf | v152 }

By default, the VBD

attribute type is V.152.

VBD codec

mode

fax-modem common vbd-codec

{ g711a | g711u }

By default, the VBD

codec mode is A-law

algorithm in G.711.

VBD payload

type

fax-modem common vbd-pt-type

{ dynamic | static }

By default, the VBD

payload type is static.

Fax

transmission

mode

fax-modem fax transmission-mode

{ thoroughly | t38 }

By default, the fax

transmission mode of a

SIP AG interface is

transparent

transmission.

Modem

transmission

mode

fax-modem modem transmission-

mode { thoroughly | relay }

By default, the modem

transmission mode of a

SIP AG interface is

transparent

transmission.





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Paramet

ers of

the

seconda

ry proxy

server

Static IP

address and

port number

secondary-proxy-addr static static-

ip-address static-ip-address-port

By default, no static IP

address or port number


secondary proxy

server.

DNS-A

domain name

and port

number

secondary-proxy-addr dns-a dns-a-

domain-name dns-a-port-number

By default, no DNS-A

domain name or port

number is configured

for the secondary proxy

server.

DNS-NAPTR

domain name

secondary-proxy-addr dns-naptr

dns-naptr-domain-name

By default, no DNS-

NAPTR domain name


secondary proxyserver.

DNS-SRV

domain name

secondary-proxy-addr dns-srv dns-

srv-domain-name

By default, no DNS-

SRV domain name is

configured for the

secondary proxy

server.

Mode used for sending

DHCP Request messages

proxy-dhcp-option option-value By default, the mode

used for sending DHCP

Request messages is

none.

Software parameters ag number-parameter name value Different software

parameters may use

different value ranges.

For details, see ag

number-parameter.

String parameters ag string-parameter strpara-name

strpara-value

For details about string

parameters, see ag

string-parameter.

RFC

2833transmis

sion

RFC 2833

DTMFtransmission

nte-dtmf { enable | disable } By default, RFC 2833

DTMF transmission isenabled.

RFC 2833 fax

and modem

transmission

nte-fax-modem { enable | disable } By default, RFC 2833

fax and modem

transmission is

enabled.

RFC 2833

hookflash

transmission

nte-flash-hook { enable | disable } By default, RFC 2833

hookflash transmission

is disabled.





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Mode in

which RFC

2833

transmission

is started

based on

negotiation

nte-negotiation-mode mode-value By default, RFC 2833

redundancy

transmission is started

based on negotiation in

passive-nego mode.

Proxy

function

s

Mode used by

a SIP AG to

detect a proxy

server

proxy-detect-mode { option | probe |

register }

By default, a SIP AG

uses the probe mode to

detect a proxy server.

Mode in

which a SIP

AG is dualhomed to

proxy servers

proxy-dual-home { disable | manual-

switchover | auto-switchover }


supports dual homing

but does not supportautomatic switchover.

Mode used by

a SIP AG to

update the

proxy server

address

proxy-refresh-mode { no-switch |

defer | immediate }


updates the proxy

server IP address after

a delay (defer mode).

Registration of a SIP AG

with the specified proxy

server

proxy-switchover { primary |

secondary }

To maintain or

troubleshoot the

current proxy server,run this command.

RFC

2198

redunda

ncy

transmis

sion

Mode in

which RFC

2198

redundancy

transmission

is started

redundancy-negotiation-mode

negotiation-mode

By default, RFC 2198

redundancy


in no-initiative-start

mode.

Transmission

of RFC 2833

audio data

using RFC

2198

redundancy

transmission

redundancy-nte { enable | disable } By default, a SIP AG is

disabled from using

RFC 2198 redundancy

transmission to

transmit the RFC 2833

audio data.

Mode in

which RFC

2198

redundancy

transmission

is started

redundancy-start-mode mode-value By default, RFC 2198

redundancy


in ordinary mode.





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VBD

transmission

using RFC

2198

redundancy

transmission

redundancy-vbd { enable | disable } By default, a SIP AG is

enabled to transmit

VBD using RFC 2198

redundancy

transmission.

Voice

transmission

using RFC

2198

redundancy

transmission

redundancy-voice { enable |

disable }


does not use RFC 2198

redundancy

transmission to

transmit voice.

Registrar URI register-server-uri uri By default, no registrar

URI is configured for aSIP AG.

Ringing parameters ring-mode index ringmode-name

[ cadence-ring cadencering-value |

initial-ring initialring-value ] *

By default, ringing

parameters of a SIP AG

is empty.

SCTP stack parameters sctp { assoc-maximum-

retransmission assoc-max-retrans |

checksum-arithmetic checksum-arith

| heartbeat-interval hb-interval |

high-congestion-level high-

congestion-level-value | in-stream-number in-stream-num | low-

congestion-level low-congestion-

level-value | maximum-init-

retransmission max-init-retrans | no-

congestion-level no-congestion-level-

value | out-stream-number out-

stream-num | path-maximum-

retransmission path-max-retrans |

rto-alpha rto-alpha-value | rto-beta

rto-beta-value | rto-init rto-init-value |

rto-maximum rto-max | valid-cookie-

life valid-cookie-life-value | }

After SCTP stack

parameters are set,

reset the SIP AG to

make SCTP stack

parameters take effect.

SDP negotiation mode sdp-negotiation-mode { local |

remote }

By default, the remote

priority mode is used

for SDP negotiation.

Index of the service logic

used by a SIP AG

service-logic service-logic-index By default, the service

logic index of a SIP AG

is 0.





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Mapping between a call

scenario and a status code

status-code-mapping scene status-

code

After configuring the

status code of a call

scenario, reset the SIP

AG to make the

configuration take

effect.

MWI

subscrip

tion

function

MWI

subscription

function

subscribe mwi { enable | disable } By default, the MWI

subscription function is

disabled.

Register

status

subscription

function

subscribe reg { enable | disable } By default, the register

status subscription

function is disabled.

UA profile

subscription

function

subscribe ua-profile { enable |

disable }

By default, the UA

profile subscription

function is enabled on a

SIP AG.

Codec priority, codec

value, and packetization

interval

user-defined-profile codec-priority

pri-value codec-value pt-value

If you change the

priority of a codec

value, priorities of

other codec values are

also changed.

Software parameter with

a specified sequence

number

user-defined-profile number-

parameter strpara-no strpara-value

–

Priority of a service type user-defined-profile service-priority

pri-value srvid-value

If you change the

priority of a service

type, priorities of other

service types are also

changed.

----End

1.4.4 Checking the Configuration

Procedure

l Run the display voice sipag [ sipag-interface-id { running | config } ] command to check

the configuration of SIP AG parameters.

l Run the display voice voip-address command to check the signaling and media IP

addresses pool.

----End





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1.5 (Optional) Setting Parameters for Voice Interfaces

This section describes how to set parameters for voice interfaces.

1.5.1 Setting Parameters for an FXS Interface

This section describes how to set parameters for an FXS interface.


An FXS interface connects to a POTS phone. To achieve high transmission efficiency on an

FXS interface, properly set parameters for the FXS interface on the AR2200, including physical

attributes, electrical attributes, and KC attributes.


Before setting parameters for an FXS interface, complete the following task:

l Configuring the AR2200 to work in SIPAG mode

l Ensuring that the voice 4FXS1FXO board is working properly

Data Preparation

To set parameters for an FXS interface, you need the following data.

No. Data

1 Polarity reversal pulse level width, polarity reversal mode, and dialing mode

2 Current, impedance, send gain, and receive gain

3 High-level pulse width, low-level pulse width, KC accounting mode, and voltage

operating

Procedure



Step 2 Run:voice


Step 3 Run: port fxs slotid/subcardid/portid

The FXS interface view is displayed.





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Step 4 Set physical attributes for the FXS interface.

l Run:reverse-pole-pulse enable

The polarity reversal pulse function is enabled.

l Run:reverse-pole-pulse level level-value

The polarity reversal pulse level width is set.

l Run: polarity-reverse-mode { hard | soft }

The polarity reversal mode is set.

l Run:clip-reverse-pole-pulse enable

The CLIP polarity reversal pulse function is enabled.

l Run:dial-mode { dtmf | pulse | both }

The dialing mode is set.

l Run:line-lock enable

Port locking is enabled.

l Run:nlp-mode { disable | normal | ec-gain-adjust | enhance | reduce }

The non-linear processing (NLP) mode is set.

Step 5 Set electrical attributes for the FXS interface.

l Run:electric current current-value

The current is set.

l Run:electric impedance impedance-value

The impedance is set.

l Run:electric send-gain sendgain-value

The send gain is set.

l Run:electric receive-gain recvgain-value

The receive gain is set.

Step 6 Set KC attributes for the FXS interface.

l Run:kc high-level highlevel

The high-level pulse width is set.

l Run:kc low-level lowlevel

The low-level pulse width is set.

l Run:kc type { 16kc | 12kc }

The KC accounting mode is set.

l Run:





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kc voltage voltage

The voltage operating is set.

----End


Run the display voice port fxs [ state ] [ slotid/subcardid/portid ] command to check the FXS

interface configuration.

1.5.2 Setting parameters for a BRA Interface

This section describes how to set parameters for a BRA interface.


A basic rate access (BRA) interface connects to an ISDN phone. On the AR2200, you can enable

the BRA interface Layer 2 monitoring, remote power supply, automatic deactivation, and alarm

functions, and set the working mode and Layer 1 activation mode on a BRA interface.


Before setting parameters for a BRA interface, complete the following task:


l Ensuring that the 2BST board is working properly

Data Preparation

To set parameters for a BRA interface, you need the following data.

No. Data

1 Interface working mode, automatic deactivation delay, and Layer 1 activation mode

Procedure

Step 1 Run:

system-view


Step 2 Run:voice


Step 3 Run: port bra slotid/subcardid/portid

The BRA interface view is displayed.

Step 4 Run:l2-monitor enable





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BRA interface Layer 2 monitoring is enabled.

Step 5 Run: work-mode { p2p | p2mp }

The working mode is set.

Step 6 Run:remote-power enable

Remote power supply is enabled.

Step 7 Run:auto-deactive enable

Automatic deactivation is enabled.

Step 8 Run:

auto-deactive delay delay

The automatic deactivation delay is set.

Step 9 Run:active-mode { unstable | stable }

The Layer 1 activation mode is set.

Step 10 Run:alarm enable

The alarm function is enabled.

----End


Run the display voice port bra [ state ] [ slotid/subcardid/portid ] command to check the BRA


1.5.3 Setting Parameters for a PRA Interface

This section describes how to set parameters for a PRA interface.


A PRA interface connects to a PBX or PSTN network. On the AR2200, you can enable the

CRC4 check, E1 interface Layer 2 monitoring, and E1 interface pulse code modulation (PCM)

alarm functions, and set the CRC alarm threshold and E1 interface signaling mode on a PRA

interface.


Before setting parameters for a PRA interface, complete the following task:


l Ensuring that the 1E1T1-M/2E1T1-M board is working properly





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Data Preparation

To set parameters for a PRA interface, you need the following data.

No. Data

1 Access digital section mode

2 CRC alarm threshold and signaling mode of the E1 interface

Procedure



Step 2 Run:set workmode slot slot-id e1t1 e1-voice

The E1 interface board is configured as an E1 voice card.

Step 3 Run:voice


Step 4 Run: port ve1 slotid/subcardid/portid

The PRA interface view is displayed.

Step 5 Run:access-mode { digital | direct }

The access digital section mode is set.

Step 6 Run:crc4 enable

CRC4 check is enabled.

Step 7 Run:

crc-alarm-threshold { es es-threshold | cses cses-threshold | dm dm- threshold }

The CRC alarm threshold is set.

Step 8 Run:l2-monitor enable

E1 interface Layer 2 monitoring is enabled.

Step 9 Run: pcm-alarm

The PCM alarm function is enabled.

Step 10 Run:signal { CCS | CAS }





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The signaling mode is set.

----End

Checking the ConfigurationRun the display voice port ve1 [ state ] [ slotid/subcardid/portid ] command to check the PRA


1.6 (Optional) Setting Voice Parameters on a SIP AG

This section describes how to set voice parameters on a SIP AG.

1.6.1 Setting System Parameters

This section describes how to set system parameters.

Establishing the Configuration Task

Before setting system parameters, familiarize yourself with the applicable environment,




The AR2200 working in SIP AG mode can exchange information with a softswitch device using

SIP. Different countries and regions use different voice parameter standards; therefore, set voice

parameters on the SIP AG in accordance with local standards.


Before setting system parameters, complete the following task:


l Configuring IP addresses and routing protocols for interfaces to ensure connectivity

Data Preparation

To set system parameters, you need the following data.

No. Data

1 Country/Region identifier

2 Lower threshold and upper threshold for hookflash pressing

3 MWI mode

4 Codec mode for voice services

5 AC amplitude of the ringing current, frequency of the ringing current, and cadence

ratio





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No. Data

6 CLIP parameters in offhook state:

l Interval between the time when the ACK message is received and the time when

the FSK is transmittedl Maximum duration between the time when the DT-AS is transmitted and the time

when the ACK message is received

l Duration of the DT-AS

l Level of the DT-AS

l Level of the FSK

l Number of bits of the FSK synchronization mask

7 CLIP parameters in onhook state:

l Number of bits seized by a channel

l Duration of the DT-AS

l Interval between the time when the DT-AS is transmitted and the time when the

FSK is transmitted

l Level of the DT-AS

l Level of the FSK

l Number of bits of the FSK synchronization mask

8 Maximum uplink bandwidth and reserved bandwidth for emergency calls

Configuring a Country/Region Identifier

This section describes how to configure a country/region identifier.


A country/region identifier is configured on a SIP AG so that user terminals connect to the SIP

AG can comply with the local standard.

Procedure



Step 2 Run:voice


Step 3 Run:country { brazil | britain-msfuk | britain-etsi | bulgaria | china-hongkong |china-mainland | egypt | france | singapore | thailand }

A country/region identifier is configured.





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By default, a country/region identifier is China mainland.

----End

Setting Upper and Lower Thresholds for Hookflash Pressing The upper and lower thresholds for hookflash pressing determine the hookflash duration.

Context

Hookflash or flash is a button on a telephone that simulates quickly hanging up and then picking

up again (a quick off-hook/on-hook/off-hook cycle). The hookflash can be pressed by a calling

party or a called party:

l Hookflash pressed by a called party: If the called party user A wants to transfer an incoming

call to user B, user A can press the hookflash and dial the number of user B.

l Hookflash pressed by a calling party: User A calls user B. User B answers the call and talkswith user A. User A can press the hookflash and dial the number of user C after hearing a

special dial tone.

After user A presses the hookflash:

l If user C is busy, user A can press the hookflash and talk with user B.

l If user C does not respond for a long period of time, user A can press the hookflash and

talk with user B.

l If the phone of user C rings, user A hangs up and user B hears the ringback tone. User C

picks up the phone and talks with user B.

l Whether a called party can be transferred to a toll call is restricted by the outgoing right of

the called party.

Procedure



Step 2 Run:voice


Step 3 Run:flash-hook lower lower-value

The lower threshold for hookflash pressing is set.

By default, the lower threshold for hookflash pressing is 100 ms.

Step 4 Run:flash-hook upper upper-value

The upper threshold for hookflash pressing is set.

By default, the upper threshold for hookflash pressing is 350 ms.





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NOTE

The lower threshold for hookflash pressing must be 50 ms less than the upper threshold for hookflash

pressing.

----End

Setting the MWI Mode

You need to set the MWI mode according to user habits.

Context

If there are leave messages, the user device configured with the MWI function makes the

indicator on or plays a tone, indicating that there are leave messages. You can set the MWI mode

according to user habits.

ProcedureStep 1 Run:

system-view


Step 2 Run:voice


Step 3 Run: mwi-mode { fsk-with-ring | fsk-without-ring | high-voltage }

The MWI mode is set.

----End

Configuring the Codec Mode for Voice Services

You need to configure the codec mode for voice services so that the user device can work in

compliance with the local standard.

Context

G.711, also known as Pulse Code Modulation (PCM), is a commonly used waveform codec. G.711 defines two main compression algorithms, the µ-law algorithm (used in North America &

Japan) and A-law algorithm (used in Europe and China). A-law encoding takes a 13-bit signed

linear audio sample as input. μ-law encoding takes a 15-bit signed linear audio sample as input.

Procedure



Step 2 Run:voice





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Step 3 Run: pcm { a-law | u-law }

The codec mode is configured for voice services.

----End

Enabling the AR2200 to Check the Checksum in Packets

You can enable the AR2200 to check the checksum in received messages or encapsulate the

checksum in messages to be sent.

Procedure



Step 2 Run:voice


Step 3 Run the following commands as required.

l Run:

sctp checksum receive enable

The AR2200 is enabled to check the checksum in received messages.

l Run:

sctp checksum send enable

The AR2200 is enabled to encapsulate the checksum in messages to be sent.

----End

Setting Ringing Parameters

You can set the AC amplitude of the ringing current, frequency of the ringing current, cadence

ratio, and initial ringing function on the AR2200 to meet local standards.

Context

Different countries and regions use different ringing standards. You can set the AC amplitude

of the ringing current to adjust the ringing tone volume, voice pitch, cadence ratio, and initial

ringing function on the AR2200 to meet local standards.

Procedure







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Step 2 Run:voice


Step 3 Run the following command according to user requirements.l Run:

ring frequency { 16hz | 25hz | 50hz }

The frequency of the ringing current is set.

l Run:

user-defined-ring ring-index { first-ring first-ring-period | first-interval first-interval | second-ring second-ring-period | second-interval second-

interval | third-ring third-ring-period | third-interval third-interval }*

The cadence ratio of a specified user-defined ring type is set.

l Run:

stop-initial-ring { enable | disable }

The initial ringing function is enabled.

----End

Enabling the Function That Reduces the Feed on Locked Ports

Procedure



Step 2 Run:voice


Step 3 Run: park-feed enable

The function that reduces the feed on locked ports is enabled.

This function takes effect for only FXS ports.

----End

Setting CLIP Parameters in Offhook State

You can set CLIP parameters in offhook state on the AR2200 so that the AR2200 can work with

the phone terminal.

Context

When the CLIP service is registered, CLIP parameters in offhook state need to be configured

on the AR2200 so that the AR2200 can work with the phone terminal. Generally, default

parameter settings are used. If CLIP parameters are not set properly, change relevant CLIP parameters.





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Procedure



Step 2 Run:voice



l Run:

clip offhook ack-fsk-interval ack-fsk-interval

The interval between the time when the ACK message is received and the time when the

frequency-shift keying (FSK) is transmitted in offhook state is set.

l Run:clip offhook dtas-ack-interval dtas-ack-interval

The maximum duration between the time when the dual tone-alerting signal (DT-AS) is

transmitted and the time when the ACK message is received in offhook state is set.

l Run:

clip offhook dtas-duration dtas-dur-value

The duration of the dual tone-alerting signal (DT-AS) in offhook state is set.

l Run:

clip offhook dtas-level dtas-level

The level of the DT-AS in offhook state is configured.

l Run:

clip offhook fsk-level fsk-level

The level of the FSK in offhook state is configured.

l Run:

clip offhook mark-signal-bit mark-signal-bit

The number of bits of the FSK synchronization mask in offhook state is set.

----End

Setting CLIP Parameters in Onhook State

You can set CLIP parameters in onhook state on the AR2200 so that the AR2200 can work with

the phone terminal.

Context

When the CLIP service is registered, CLIP parameters in onhook state need to be configured on

the AR2200 so that the AR2200 can work with the phone terminal. Generally, default parameter

settings are used. If CLIP parameters are not set properly, change relevant CLIP parameters.

Procedure






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Step 2 Run:voice



l Run:

clip onhook channel-seize-bit channel-seize-bit

The number of bits seized by a channel in onhook state is set.

l Run:

clip onhook dtas-duration dtas-dur-value

The duration of the DT-AS in onhook state is set.

l Run:

clip onhook dtas-fsk-interval dtas-fsk-interval

The interval between the time when the DT-AS is transmitted and the time when the FSK

is transmitted in onhook state is set.

l Run:

clip onhook dtas-level dtas-level

The level of the DT-AS in onhook state is configured.

l Run:

clip onhook fsk-level fsk-level

The level of the FSK in onhook state is configured.

l Run:

clip onhook mark-signal-bit marksignal-bit

The number of bits of the FSK synchronization mask in onhook state is set.

----End

Setting Uplink Bandwidth Parameters

This section describes how to set uplink bandwidth parameters.

Context

The AR2200 provides uplink bandwidth control. When the system detects that the uplink bandwidth usage reaches the configured upper threshold, it restricts calls and generates an alarm.

If the uplink bandwidth is insufficient, the system processes calls based on user levels. Common

users may not obtain services.

Procedure



Step 2 Run:voice





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l Run:

media-bandwidth-control enableThe AR2200 is enabled to restrict calls when the uplink bandwidth is enabled.

l Run:

media-bandwidth-control maximum max-bandwidth

The maximum uplink bandwidth is set.

l Run:

media-bandwidth-control reserved-for-emergency reserved-bandwidth

The reserved bandwidth is set for emergency calls.

NOTE

The reserved bandwidth for emergency calls must be smaller than the maximum uplink bandwidth

configured by the media-bandwidth-control maximum command.

----End

Setting the Number of SIP Register Messages Sent Every Second

Procedure



Step 2 Run:voice


Step 3 Run:sip-reg-count-per-second reg-count-value

The number of SIP register messages sent every second is set.

----End


After the configuration of system parameters is complete, you can view the system parameters

such as the SIP interface's IP address, upper and lower thresholds for hookflash pressing, and

CLIP parameters.

Procedure

l Run the display voice configuration command to check the voice configuration.

l Run the display voice user-defined-ring [ring-index ] command to check user-defined

ring information.

l Run the display voice clip command to check CLIP parameters.





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l Run the display voice sip-reg-count-per-second command to check the number of SIP

Register messages initiated per second.

----End

1.6.2 Setting SIP Stack ParametersThis section describes how to set SIP stack parameters.


Before setting SIP stack parameters, familiarize yourself with the applicable environment,



Applicable EnvironmentSIP is an IETF-defined signaling protocol widely used for controlling communication sessions

such as voice and video calls over Internet Protocol (IP). SIP, RTP, RTCP, RTSP, and other

protocols constitute a SIP protocol stack.


Before setting SIP protocol stack parameters, complete the following task:



Data Preparation

To set SIP protocol stack parameters, you need the following data.

No. Data

1 Minimum value of the session timer, value of the unreliable provisional

response retransmission timer, value of the reliable provisional response

retransmission timer, value of the T1 timer, value of the T2 timer, value of the

T4 timer, and value of the TD timer

2 Values of the Max-forwards field, Organization header field, Server header field, and User-Agent header field

Setting the Values of Timers

This section describes how to set values of timers of the SIP stack.

Procedure






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Step 2 Run:voice


Step 3 Run:sip

The SIP view is displayed.


l Run:

entity-based-sessions-timer enable

The session timer defined by RFC 4028 based on the UA is defined.

l Run:

min-se min-se–period

The minimum value of the session timer is set.

l Run:

session-progress session-progress–period

The value of the unreliable provisional response retransmission timer is set.

l Run:

session-rel-progress session-rel-progress–period

The value of the reliable provisional response retransmission timer is set.

l Run:

t1 t1–period

The value of the T1 timer is set.

l Run:

t2 t2–period


l Run:

t4 t4–period


l Run:

td td–period

The value of the TD timer is set.

----End

Setting Values of Header Fields

This section describes how to set values of header fields.

Procedure






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Step 2 Run:voice


Step 3 Run:sip

The SIP view is displayed.


l Run:

header-folding enable

Header folding in the authentication header field is enabled.

l Run:

field-header max-forwards max-forwards-value

The value of the Max-forwards header field is set.

l Run:

field-header organization organization-head

The value of the Organization header field is set.

l Run:

field-header server server-head

The value of the Server header field is set.

l Run:

field-header user-agent user-agent-head

The value of the User-agent header field is set.

----End


After the configuration is complete, you can view the configured SIP stack parameters.

Procedure

lRun the display voice sip command to check SIP parameters.

----End

1.6.3 Setting DSP Parameters

This section describes how to set DSP parameters.


Before setting DSP parameters, familiarize yourself with the applicable environment, complete

the pre-configuration tasks, and obtain the data required for the configuration. This will helpyou complete the configuration task quickly and accurately.





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The digital signal processing (DSP) collects, converts, filters, measures, enhances, compresses,

or identifies signals and coverts the signal from an analog to a digital form.

The DSP module converts analog voice signals into digital signals and stores a certain number

of digital signals into packets for transmission. To improve the voice communication quality,

the DSP needs to further process voice signals.


Before setting DSP parameters, complete the following task:



Data Preparation

To set DSP parameters, you need the following data.

No. Data

1 VBD switching mode

2 Default DSP channel code type and default interval at which the DSP channel

packetizes RTP packets

3 T.30 redundancy parameter value of the T.38 fax, T.4 redundancy parameter value

of the T.38 fax, fax training mode, and maximum fax training rate

4 Alarm threshold of the dynamic jitter buffer, initial value of the dynamic jitter buffer,

maximum value of the dynamic jitter buffer, maximum value of the static jitter buffer,

minimum value of the dynamic jitter buffer, minimum value of the static jitter buffer,

and initial value of the static jitter buffer of a DSP channel

5 RTP payload type value, G.726-16k payload type value, G.726-24k payload type

value, G.726-32k payload type value, G.726-40k payload type value, NTE payload

type value, redundancy payload type value, and VBD payload value of a DSP channel

6 Interval at which a DSP channel sends RTCP packets and threshold for the number

configuraticonfig voipon guide - voice(v200r002c01_01)

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