AltiGen comm ACM 5.1 사용자 설명서
Chapter 11: Board Configuration
142
AltiWare ACM 5.1 Administration Manual
Configuring Virtual Boards SIPSP and H323SP
A VoIP connection typically consists of two parts:
•
Signal Channel—responsible for setting up and tearing down a call using protocol.
For example, SIP protocol is used in AltiWare 5.1 to build a signal channel between
the server and the IP phone.
For example, SIP protocol is used in AltiWare 5.1 to build a signal channel between
the server and the IP phone.
•
Media Path—responsible for encoding, transmitting, and decoding voice for both
parties. For example, when an IP phone user makes a call to an outside number, the
voice will be encoded at the IP phone, transmitted to the system via the IP network,
decoded by the VoIP codec, and passed to a trunk port so that the external party will
hear the voice.
parties. For example, when an IP phone user makes a call to an outside number, the
voice will be encoded at the IP phone, transmitted to the system via the IP network,
decoded by the VoIP codec, and passed to a trunk port so that the external party will
hear the voice.
The purpose of virtual boards SIPSP and H323SP is to build signal channels for different
connection types, IP extensions, SIP Tie Trunks, SIP Trunking from ITSP, and H323 Tie
Trunks. Each channel will have its channel ID similar to channels on a Triton extension
or trunk board. When an IP phone registers to the system, a channel ID will be assigned
to the IP extension. However, these channels are only responsible for processing
protocol and call control signals. They require a media path from a VoIP board or from
the IP phone to establish a voice steam so that both sides can hear.
Notes:
connection types, IP extensions, SIP Tie Trunks, SIP Trunking from ITSP, and H323 Tie
Trunks. Each channel will have its channel ID similar to channels on a Triton extension
or trunk board. When an IP phone registers to the system, a channel ID will be assigned
to the IP extension. However, these channels are only responsible for processing
protocol and call control signals. They require a media path from a VoIP board or from
the IP phone to establish a voice steam so that both sides can hear.
Notes:
•
Make sure you have enough VoIP resource boards.
•
The more signal channels, the more system memory and CPU power required.
Proper planning is essential.
Proper planning is essential.
•
Changing the number of signal channels requires that you stop and restart the
switching and gateway services.
switching and gateway services.
•
SIP Trunking Channel requires a license to activate.
Configuring the SIPSP Board
Double-clicking a SIPSP board in Boards view and then clicking the Board
Configuration button opens this dialog box:
Configuration button opens this dialog box:
Figure 17. SIP Signaling Channel Configuration dialog box
If you change the
number of SIP extension
or tie trunk channels,
you must stop and
restart the switching and
gateway services.
number of SIP extension
or tie trunk channels,
you must stop and
restart the switching and
gateway services.