CyberData 011146 Operation Manual page 41

Web Page Item
SIP Auth ID
SIP Auth Password
Re-registration Interval (in seconds)
Relay rings to multicast
Multicast Address
Multicast Port
Call Disconnection
Terminate Call After Delay
Audio Codec Selection
Codec
RTP Settings
RTP Port (even)
Asymmetric RTP
Jitter Buffer
RTP Encryption (SRTP)
Note
1. Enter the IP address of the SIP Server.
CyberData Corporation
Table 2-7. SIP Configuration Parameters (continued)
Description
Specify the Authenticate ID for the SIP Server. This parameter is required for SIP
registration authentication. Enter up to 64 alphanumeric characters.
Specify the Authenticate Password for the SIP Server. This parameter is required for
SIP registration authentication. Enter up to 64 alphanumeric characters.
The SIP Re-registration Interval (in seconds) is the SIP Registration lease time, also
known as the expiry. The supported range is 30-3600 seconds. Enter up to 4 digits.
When selected, the device will play ring tones to the specified multicast address and
port.
The multicast address used for nightring audio.
The multicast port used for nightring audio.
Automatically terminate an active call after a given delay in seconds. A value of 0 will
disable this function. Enter up to 8 digits.
Select desired codec (only one may be chosen).
Specify the port number used for the RTP stream after establishing a SIP call. This
port number must be an even number and defaults to 10500. The supported range is
0-65536. Enter up to 5 digits.
Specify if the remote endpoint will send and receive RTP packets on different ports. If
set to false, the device will track the address/port that is sending RTP packets during a
SIP call. If the address/port changes mid-stream, the device will disregard the SDP
and send all further RTP packets to this new address.
If set to true, this device will ignore the sending address/port and send RTP as
specified in the SDP. Warning! Enabling asymmetric RTP can cause the RTP stream
to be lost.
Most installations should not enable asymmetric RTP.
Specify the size of the jitter buffer (in milliseconds) used for SIP calls. Valid values are
50-1000.
When enabled, a SIP call's audio streams are encrypted using SRTP.
Click the Save button to save your configuration settings.
Click on the Reboot button to reboot the system.
Click on the Toggle Help button to see a short description of some of the web page
items. First click on the Toggle Help button, and you will see a question mark (
appear next to some of the web page items. Move the mouse pointer to hover over a
question mark to see a short description of a specific web page item.
For specific server configurations, go to the following website address:
https://www.cyberdata.net/pages/connecting-to-ip-pbx-servers
931803A
Setting Up the SIP Paging Server
Configure the SIP Parameters
)
Operations Guide
33