Huawei AR1200 Series User manual

Huawei AR1200 Series Enterprise Routers
V200R001C00
Configuration Guide - Voice
Issue 03
Date 2011-09-30
HUAWEI TECHNOLOGIES CO., LTD.

Copyright © Huawei Technologies Co., Ltd. 2011. 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]
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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 supported by the AR1200 device.
This document describes how to configure the voice feature.
This document is intended for:
lData configuration engineers
lCommissioning engineers
lNetwork monitoring engineers
lSystem 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.
Huawei AR1200 Series Enterprise Routers
Configuration Guide - Voice About This Document
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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.
&<1-n> The parameter before the & sign can be repeated 1 to n times.
# A line starting with the # sign is comments.
Change History
Changes between document issues are cumulative. Therefore, the latest document version
contains all updates made to previous versions.
Changes in Issue 03 (2011-09-30)
Based on issue 02 (2011-06-25), the document is updated as follows:
The following information is modified:
lThe configuration guide optimizes the SIP AG configuration.
Changes in Issue 02 (2011-06-25)
Based on issue 01 (2011-03-30), the document is updated as follows:
The following information is modified:
lExample for Configuring a SIP AG
l(Optional) Configuring the Three-Party Service for a SIPAG User
The following information is deleted:
lThe configuration guide of PBX is deleted.
Huawei AR1200 Series Enterprise Routers
Configuration Guide - Voice About This Document
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iii

Changes in Issue 01 (2011-03-30)
Initial commercial release.
Huawei AR1200 Series Enterprise Routers
Configuration Guide - Voice About This Document
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Copyright © Huawei Technologies Co., Ltd.
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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 AR1200......................................................................................................2
1.3 Configuring the AR1200 to Work in SIP AG Mode..........................................................................................5
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.................................................................................7
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..........................................................................................18
1.5.1 Setting Parameters for an FXS Interface.................................................................................................18
1.5.2 Setting Parameters for an FXO Interface.................................................................................................20
1.6 (Optional) Setting Voice Parameters on a SIP AG...........................................................................................22
1.6.1 Setting System Parameters......................................................................................................................22
1.6.2 Setting SIP Stack Parameters..................................................................................................................31
1.6.3 Setting DSP Parameters...........................................................................................................................34
1.7 Configuring a SIPAG User...............................................................................................................................46
1.7.1 Establishing the Configuration Task.......................................................................................................46
1.7.2 Setting Parameters for a SIP AG User.....................................................................................................47
1.7.3 Configuring a SIP AG User Group..........................................................................................................49
1.7.4 (Optional) Configuring Functions or Services in a SIP Service Data Profile.........................................51
1.7.5 Checking the Configuration.....................................................................................................................53
1.8 Resetting a SIP AG...........................................................................................................................................54
1.9 Maintaining a SIP AG......................................................................................................................................55
1.9.1 Clearing SIP AG Statistics......................................................................................................................55
1.10 Configuration Examples.................................................................................................................................56
1.10.1 Example for Configuring a SIP AG......................................................................................................56
Huawei AR1200 Series Enterprise Routers
Configuration Guide - Voice Contents
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1 SIPAG Configuration
About 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 AR1200
This section describes SIP AG features supported by the AR1200.
1.3 Configuring the AR1200 to Work in SIP AG Mode
Before configuring SIP AG service features, configure the AR1200 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 FXS interfaces and FXO 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 Resetting a SIP AG
This section describes how to reset a SIP AG.
1.9 Maintaining a SIP AG
This section describes how to maintain a SIP AG.
1.10 Configuration Examples
This section provides a SIPAG configuration example.
Huawei AR1200 Series Enterprise Routers
Configuration Guide - Voice 1 SIPAG Configuration
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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.
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
on the PSTN/ISDN and IP network. It can implement VoIP functions.
1.2 SIP AG Features Supported by the AR1200
This section describes SIP AG features supported by the AR1200.
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:
Huawei AR1200 Series Enterprise Routers
Configuration Guide - Voice 1 SIPAG Configuration
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lA 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.
lVoice 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.
Figure 1-1 SIP AG as a voice gateway
SIPAG Eth1/0/0
IMS
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.
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.
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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 Configure
d on the
SIP AG or
Not
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-party
conversation 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
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
Huawei AR1200 Series Enterprise Routers
Configuration Guide - Voice 1 SIPAG Configuration
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Copyright © Huawei Technologies Co., Ltd.
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