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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i

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.
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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.
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iii

Changes in Issue 01 (2011-03-30)
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 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
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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.
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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:
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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
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Service Type Introduction Configure
d on the
SIP AG or
Not
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 call
according 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 lInstant hotline service: After a user picks up a phone,
the SIP AG dials the hotline number for the user.
lNo 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
Anonymous call
service
The anonymous call service enables the called party not
to view information about incoming calls.
No
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.
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Applicable Environment
Before configuring SIP AG service features, configure the AR1200 to work in SIP AG mode.
You can run the display voice service-mode command to view the working mode of the
AR1200. If the AR1200 works in SIP AG mode, skip this configuration.
Pre-configuration Tasks
Before configuring the AR1200 to work in SIP AG mode, complete the following task:
lConfiguring IP addresses and routing protocols for interfaces 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 AR1200 is configured to work in SIP AG mode.
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 AR1200 is restarted.
NOTE
After the AR1200 is configured to work in SIP AG mode, restart the AR1200 to make the configuration
take effect.
----End
Checking the Configuration
After the configuration is complete, the AR1200 works in SIP AG mode.
Run the display voice service-mode command to check the working mode of the AR1200.
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<Huawei> display voice service-mode
The voice service mode is SIPAG
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.
Applicable Environment
On the IMS, when a media gateway controller (MGC) and a SIP AG exchange media and
signaling streams, set the media and signaling IP addresses, signaling port number, and
transmission protocol on the SIP AG.
Pre-configuration Tasks
Before configuring a SIP AG interface, complete the following task:
lConfiguring the AR1200 to work in SIP AG mode
Data Preparation
To configure a SIP AG interface, you need the following data.
No. Data
1Media 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 streams
and signaling streams can use the same IP address.
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Procedure
Step 1 Run:
system-view
The system view is displayed.
Step 2 Run:
voice
The voice view is displayed.
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:
lTo specify the ip-address parameter, ensure that an IP address has been configured for the
specified interface.
lTo 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.
Context
On the IMS, when the MGC and the SIP AG exchange media and signaling streams, set the
media and signaling IP addresses, signaling port number, and transmission protocol.
Procedure
Step 1 Run:
system-view
The system view is displayed.
Step 2 Run:
voice
The voice view is displayed.
Step 3 (Optional) Run:
local-digitmap name name-value { type { normal | emergency | scc | direct-centrex
| second-centrex } | body body-value [ append ] }*
The digitmap is configured.
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Step 4 Run:
sipag mgid
A SIP AG is created and the SIP AG view is displayed.
Step 5 Set mandatory parameters on the SIP AG interface.
Parameter Command Remarks
Media IP address media-ip { media-ip | media-addr-
name media-addr-name }
The media IP address
of the SIP AG interface
must be obtained from
the media IP address
pool.
Signaling IP address signalling-ip { signal-ip | signalling-
addr-name signal-addr-name }
The signaling IP
address of the SIP AG
interface must be
obtained from the
signaling IP address
pool.
Signaling port number signalling-port signalling-port-value By default, no
signaling port number
is configured for a SIP
AG interface.
Mode in which the IP
address of a proxy server
is obtained
proxy-address-mode { static | dns-a |
dns-srv | dns-naptr }
Mode in which the IP
address of a proxy
server is obtained:
lWhen the static
mode is used,
configure an IP
address and a port
number for the
primary proxy
server.
lWhen the DNS-A,
DNS-SRV, or
DNS-NAPTR
mode is used,
configure a domain
name for the
primary proxy
server.
Paramet
ers for
the
primary
proxy
server
IP address primary-proxy-ip primary-proxy-ip-
value1 [ primary-proxy-ip-value2 ]
By default, no IP
address is configured
for the primary proxy
server.
Domain name primary-proxy-domain primary-
proxy-domain-name
By default, no domain
name is configured for
the primary proxy
server.
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Parameter Command Remarks
Port number primary-proxy-port primary-proxy-
port-value
By default, no port
number is configured
for the primary proxy
server.
Transmission protocol transfer { tcp | udp | sctp } 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.
Parameter Command Remarks
Description description description By default, no
description is
configured for a SIP
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
paramet
ers
Authenticatio
n mode
auth-mode { interface | user } By default, the
authentication mode is
user.
Authenticatio
n user name
auth-username auth-username By default, no
authentication user
name is configured for
a SIP AG interface.
Authenticatio
n password
auth-password auth-password By default, no
authentication
password is configured
for a SIP AG interface.
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Parameter Command Remarks
Password
authenticatio
n mode
auth-password-mode { password |
ha1 }
By default, original
password
authentication is used.
Timer value digitmap { long-timer long-timer-
value | short-timer short-timer-value |
start-timer 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 is
transparent
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
transmission
mode
fax-modem common rtp-interval
{ 5ms | 10ms | 20ms | 30ms }
By default, the
packetization interval
in transparent
transmission mode is
10 ms.
VBD attribute
type
fax-modem common vbd-attribute-
type { chinatelecom |
chinatelecomext | 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 | T.38 }
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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Parameter Command Remarks
Paramet
ers of
the
seconda
ry proxy
server
IP address secondary-proxy-ip secondary-
proxy-ip-value1 [ secondary-proxy-ip-
value2 ]
By default, no IP
address, domain name,
or port number is
configured for the
secondary proxy
server.
Configure either the
domain name or IP
address for the
secondary proxy
server.
primary-proxy-ip-
value2 is optional and
takes effect only when
the Stream Control
Transmission Protocol
(SCTP) is used.
The secondary port
number must be
different from the
primary port number.
Domain name secondary-proxy-domain secondary-
proxy-domain-name
Port number secondary-proxy-port secondary-
proxy-port-value
Mode used for sending
DHCP Request messages
proxy-dhcp-option { none |
option120 | option21 }
By default, the mode
used for sending DHCP
Request messages is
none.
Software parameters mg number-parameter name value Different software
parameters may use
different value ranges.
For details, see mg
number-parameter.
String parameters mg string-parameter strpara-no
strpara-value
For details about string
parameters, see mg
string-parameter.
RFC
2833
transmis
sion
RFC 2833
DTMF
transmission
nte-dtmf { enable | disable } By default, RFC 2833
DTMF transmission is
enabled.
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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Parameter Command Remarks
Mode in
which RFC
2833
transmission
is started
based on
negotiation
nte-negotiation-mode { negostart |
fixedstart | no-initiative-start }
By default, RFC 2833
redundancy
transmission is started
based on negotiation in
no-initiative-start
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 dual
homed to
proxy servers
proxy-dual-home { disable | dual-
home | auto-switchover }
By default, a SIP AG
supports dual homing
but does not support
automatic switchover.
Mode used by
a SIP AG to
update the
proxy server
address
proxy-refresh-mode { no-switch |
defer | immediate }
By default, a SIP AG
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
{ negostart | fixedstart | no-initiative-
start }
By default, RFC 2198
redundancy
transmission is started
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
{ ordinary2198 | smart2198 }
By default, RFC 2198
redundancy
transmission is started
in ordinary mode.
Huawei AR1200 Series Enterprise Routers
Configuration Guide - Voice 1 SIPAG Configuration
Issue 03 (2011-09-30) Huawei Proprietary and Confidential
Copyright © Huawei Technologies Co., Ltd.
13

Parameter Command Remarks
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 }
By default, a SIP AG
does not use RFC 2198
redundancy
transmission to
transmit voice.
Registrar URI register-uri uri By default, no registrar
URI is configured for a
SIP AG.
Ringing parameters ringmode 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.
Huawei AR1200 Series Enterprise Routers
Configuration Guide - Voice 1 SIPAG Configuration
Issue 03 (2011-09-30) Huawei Proprietary and Confidential
Copyright © Huawei Technologies Co., Ltd.
14