SIP Settings

Before you start creating extensions for SIP phones or SIP trunks, you should review options configured for SIP devices.

To configure common options used for all SIP devices (phones and trunks), perform the following steps:

  1. Open the UCx Web-based Configuration Utility
  2. From the PBX tab, select PBX Configuration
  3. From the left side column, select SIP Settings
  4. Review the configured defaults
  5. If you made any changes in the settings for SIP phones, press the Submit Changes button to save the configuration

You should review and change the default values for some of the options as required: 

NAT Settings

Update the NAT specific configuration options to match your network configuration.

Click on Auto Configure to let the system detect and automatically populate the values for External IP and Local Networks.
Click on Add Local Network Field to add more fields for multiple networks.


Audio Codecs


Video Codecs


Media & RTP Settings


Notification & MWI


Registration Settings


Jitter Buffer Settings

If you have SIP external devices (phones or trunks) that are not located on your local subnet, you should set the Jitter Buffer option to Enabled.


Advanced General Settings

If you want to change the default port number (5060) that is used by the UCx system for communication with SIP phones (for example for security reasons), enter the desired port number in the Bind Port field.

For interoperability with recent releases of Nortel/Avaya CS1000 and BCM50 switches, the Legacy User Option Parsing must be set to Yes.


If you are using an FXS16 at a remote location via NAT, then the setting for Check User Domain must be set to No.

It is highly recommended that you leave the fields Allow SIP Guests and Allow Anonymous Inbound SIP Calls at the default values of No for security reasons.

You can override some of the default settings configured on this page with device specific settings configured on the SIP extension or SIP trunk configuration page of each SIP device.
Other SIP Settings:

By default, the UCX sends SIP media (RTP) packets associated with SIP trunks with a special marking in the IP packet that routers can use to treat them with a higher priority as they traverse the internet. This prioritization marking is commonly referred to as DSCP (Diff Serv Code Point) tagging. By default, the UCX sends the SIP media packets with the industry standard DSCP value of EF (Expedited Forwarding). Some Internet Service Providers do not support the use of the EF value for DSCP and may actually treat it as lowest priority, which will cause intermittment packet loss and impact voice quality. In these circumstances, use the Other SIP Settings to change the DSCP for the SIP trunk media packets.

The value of the first field must be entered as "tos_audio" while the second field will be the new DSCP value to be used instead of the default.  
You may use any of the recognized DSCP Names (i.e. EF, CS0, etc) defined in RFC4594 in the value field:



Self Assessment Quiz

  1. Do you know what happens if you click on the Auto Configure button under NAT Settings?
  2. Do you know when to enable Jitter Buffer?
  3. What is the default port number for SIP devices?
  4. Do you understand the security risks of setting Allow Anonymous Inbound SIP Calls to Yes?
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