CyberData InformaCast Enabled 011406 Operation Manual page 72

Web Page Item
RTP Settings
RTP Port (even)
Jitter Buffer
Call Disconnection
Terminate Call After Delay
Codec Selection
Force Selected Codec
Codec
Note
Note
Operations Guide
Table 2-17. SIP Page Parameters (continued)
Description
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 the size of the jitter buffer (in milliseconds) used for SIP calls. Valid values are
50-1000.
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.
Note: You need to reboot for changes to take effect.
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.
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CyberData Corporation