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Item Description
Route Description Enter the description of the call route.
Call Route Type
SIP
Proxy Server
Use a SIP proxy server to complete
calling.
Required
to use one
method.
IP Routing
Use the SIP protocol to perform direct
calling. It you select this option, you
must provide the destination address
and port number.
Binding Server
Group
Select a server group from the
Server
Group
list. You can add SIP server
groups into the list in
Voice
Management
>
Call Connection
>
SIP Server Group Management
.
Trunk
Trunk Route
Line
Select a trunk routing line from the list
that displays all available voice
subscriber lines.
Transport Layer
Protocol for Call
Route
Select one of the following transport layer protocols:
• UDP.
• TCP.
• TLS.
By default, UDP is selected.
URL Scheme for
Call Route
• SIP—Specifies the SIP scheme.
• SIPS—Specifies the SIPS scheme.
By default, the SIP scheme is selected.
Register Function
• Enable. After you select the Enable option, you can configure the authentication
related options.
• Disable.
IMPORTANT:
The trunk routing mode supports register function. Authentication related options and
their meanings are the same as those of local number and therefore are not included
here.
Jitter-buffer
Adaptive Mode
• Enable—Select this option to buffer the voice packers received from the IP side,
so that the received voice packets can be played out evenly.
• Disable—Select this option to not buffer the voice packers received from the IP
side.
Jitter-buffer Initial
Delay
Specify the initial duration for buffering voice packers received from the IP side. The
default value is 30 seconds.
Jitter-buffer
Maximum Delay
Specify the maximum duration for buffering voice packers received from the IP side.
The default value is 160 seconds.
Status Enable or disable the call route.
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