Web Page Item
Call Disconnection
Terminate Call After Delay
Audio Codec Selection
Codec
RTP Settings
RTP Port (even)
Asymmetric RTP
Jitter Buffer
RTP Encryption (SRTP)
Call Disconnection
Terminate Call After Delay
Codec Selection
Force Selected Codec
Codec
Note
Note
Operations Guide
Table 2-13. SIP Page Parameters (continued)
Description
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.
Automatically terminate an active call after a given delay in seconds. A value of 0 will
disable this function. Enter up to 8 digits.
When configured, this option will allow you to force the device to negotiate for the
selected codec. Otherwise, the device will perform codec negotiation using the default
list of supported codecs.
Select the desired codec (only one may be chosen).
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
The maximum number of total characters in the dial-out field is 64.
931976A
Installing the InformaCast Enabled Paging Amplifier
InformaCast Enabled Paging Amplifier Setup
CyberData Corporation
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