User guide
Table Of Contents
- About This Manual
- Overview
- Technology Enhancements in Release 5.1
- SIP Implementation
- Multi-Site Enterprise Management - AltiEnterprise Manager
- Multi-Chassis Gateway Support (ACM only)
- Capacity Improvements
- Voice Processing New Features and Enhancements
- PBX New Features and Enhancements
- Call Center New Features and Enhancements
- Client Application New Features and Enhancements
- IP Phone New Features and Enhancements (IP600, IP710, IP705)
- Key System Features
- Technology Enhancements in Release 5.1
- Software Installation & License Registration
- Getting Around AltiWare Administrator 5.1
- System Configuration
- Setting General Parameters
- Setting a System Number Plan
- Setting Business Hours
- Routing Calls on Holidays
- Configuring System Speed Dialing
- Defining System Call Restrictions
- Creating Account Codes
- Setting up Call Reports
- Country-Relevant Settings
- Audio Peripheral Configuration
- Activity
- Feature Profiles
- CT-Bus Configurations
- Voice Mail Configuration
- Auto Attendant Configuration
- Board Configuration
- Using the Triton Resource Board
- Using the Triton MeetMe Conference Board
- Configuring the Quantum Board
- Configuring the Triton Analog Station Board
- Configuring the Triton Analog Trunk LS/GS and LS Boards
- Configuring the Triton VoIP Board
- Configuring the Triton T1/E1 Board
- Configuring Virtual Boards SIPSP and H323SP
- Configuring the MAX Board
- Configuring the Virtual MobileExt Board
- Trunk Configuration
- Trunks Out of Service
- Channel Identification
- Opening the Trunk Configuration Window
- Selecting Trunks to Set Attributes
- Configuring One or Multiple Trunks
- Setting General Trunk Attributes
- H323 Tie Trunk Properties
- SIP Tie Trunk Properties
- SIP Trunk Properties
- Triton T1/E1 Trunk Properties
- Triton Analog Trunk GS/LS Properties
- Quantum Trunk Properties
- Incoming Call Routing
- Outgoing Call Blocking
- In Call Routing Configuration
- Out Call Routing Configuration
- Extension Configuration
- Setting Up IP Extensions
- AltiGen IP Phone Configuration
- Mobile Extension Configuration
- Hunt Group Configuration
- Paging Group Configuration
- Line Park Configuration
- Workgroup Configuration
- Managing and Using MeetMe Conference
- Network Configuration Guidelines for VoIP
- Enterprise VoIP Network Management
- Understanding VoIP Bandwidth Requirements
- Opening AltiEnterprise Manager
- Setting VoIP Codec Profiles
- Assigning Codec Profiles to IP Addresses
- Defining IP Networks
- Defining the IP Dialing Table
- The Multi-site VoIP Domain
- Working with Servers in the VoIP Domain
- Managing VoIP Domain Users
- Configuring Global Least Cost Routing
- When Information May Be Out of Sync
- System Report Management
- Tools and Applications
- E1-R2 and E1 ISDN PRI Installations
- Required Service Parameters
- Network Ports
- Technical Support & Product Repair Services
- Troubleshooting
- Index
278 AltiWare ACC 5.1 Administration Manual
Assigning Codec Profiles to IP Addresses
You can specify what codec profile to use when connecting to the following VoIP devices:
• IP phones on the LAN
• a remote IP phone over WAN
• a remote AltiGen system over WAN
• SIP Trunk service provider over WAN
• multiple AltiGateways on the LAN
The codec profile assigned in the IP Device Range table (shown below) supersedes the
codec profile defined in the IP dialing table if the IP address is duplicated in both tables.
G.711/G.723/G.729
Jitter Buffer Range
(ms)
Indicates the delay, in milliseconds, used to buffer
G.711/G.723/G.729 voice packets received from the
IP network. Voice packets sent over the IP network
may incur different delays due to network load or
congestion. The jitter buffer helps to smooth out the
delay variation in the arriving voice packets and
maintain voice quality at the receiving end.
The default values for the jitter buffer for G.711 is 10
min. to 100 max milliseconds.
The default values for the jitter buffer for G.723 is 30
min. to 480 max milliseconds.
The default values for the jitter buffer for G.729 is 10
min. to 480 max milliseconds.
G.711 RTP Packet
Length (ms)
Lets you configure the length of the RTP packets for
G.711 in milliseconds. The RTP packet length can be
set to 10, 20 or 30 milliseconds. The smaller the
packet length, the larger the bandwidth required.
G.729 RTP Packet
Length (ms)
Lets you configure the length of the RTP packets for
G.729 in milliseconds. The RTP packet length can be
set to 10, 20 or 30 milliseconds.
DTMF Delivery
(Applies to SIP protocol
only)
Default—If SIP INFO is used to deliver DTMF.
RFC 2833—The DTMF pay load is embedded with
RTP. Most 3rd-party SIP gateways support this
standard. Applies to SIP TRUNK only.
In band—If deliver DTMF tone over the voice band.
It’s not reliable over G.711 codec and will not work
over G.729/G.723 codec
SIP Early Media
(Applies to SIP protocol
and SIP trunk only)
SIP Early Media allows two SIP devices to
communicate before a SIP call is actually
established. It is important for interoperability with
the SIP trunk carrier’s PSTN gateway. If SIP Early
Media is not checked, the caller may not hear the
exact ringback tone provided by the CO (the caller
may not hear any ringback tone at all).
Parameter Description