Basic Configuration Settings

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Check the Enable VoIP checkbox to enable the SIP proxy functionality of the ClariLink.

VoIP WAN Configuration

By Default all existing WANs will be used for the ClariLink VoIP device.  The default IP that is used for each WAN is its base WAN IP address. Optionally, you may enter another available address within your WAN subnet.  To delete a WAN to be used, press the corresponding Delete button.

To add a WAN to be used select the WAN alias in the WAN selection box. This cannot be a WAN that is already listed below. The new row will have the alias and the base WAN IP displayed. If the IP listed in the IP field is not the IP you wish to use for this WAN, place a different IP in this area.

The Public IP field should be filled in if the ClariLink is behind a NAT device. This can contain either an IP address or an FQDN host name. If you want all the VoIP traffic to only go out this specific WAN, then check the Primarycheckbox. If you have multiple WANs and you have chosen a Primary WAN then there will be no outbound load balancing. In the event of a failure the other WANs will be used.

Port Settings

The SIP Port Range is the range of ports on which the ClariLink will listen for, and send SIP traffic. This range should be large enough to accommodate users that have either phones with multiple lines configured to the same user or multiple phones on one or more LANs that are configured to the same user. If either of the previous cases apply, add a range where the number of ports that are configured is the desired number of different devices/locations from which a user may register.

The RTP Port Range is the range of ports on which the ClariLink will send and receive RTP traffic. Real-time Transport Protocol is used to transport real-time data such as audio and video over the Internet. This range should be large enough to accommodate your expected number of concurrent calls at any point in time , keeping in mind that each call uses two ports.

Failover Settings

The VoIP Traffic Failover Mode, if enabled, will monitor the flow of voice (RTP) traffic for all active calls. If there is no traffic flow during the configured Testing Interval, then the ClariLink will attempt to failover the call to a different WAN. The Testing Interval tells the ClariLink how often (in seconds) to check the traffic flow of all calls. If your VoIP provider uses the silence suppression feature, inbound voice traffic will stop when the other party is silent. This may cause false failovers, so you may want to disable this feature in that case.

The VoIP Latency Failover Mode, if enabled, will monitor the round-trip latency between each configured VoIP WAN and the remote endpoint of your calls. If you have a VoIP provider, the remote address of your calls will always be your VoIP provider’s server. If you have an Outbound Proxy configured then the latency to that proxy will also be monitored. Otherwise, the ClariLink will learn other remote call addresses and monitor those additionally. If the ClariLink detects that the current WAN latency for a call exceeds another WAN by more than the configured Latency Delta Time, then it will fail the call over to the WAN with a lower latency.

The VoIP Bandwidth Utilization Failover Mode, if enabled, will monitor bandwidth utilization percentage of all VoIP WANs. If the ClariLink detects a call is on a WAN with a bandwidth utilization percentage which is greater than the configured Utilization Threshold, then the ClariLink will fail the call over to a WAN with lower utilization. TheUtilization Threshold is based on the Link Speed specified on the Configure WAN page.

SIP Redirect Configuration

By default the ClariLink listens for, and handles, all SIP traffic that is received on the ports which are specified in theSIP Port Range field. If there are devices on the LAN network which are not able to specify the Outbound Proxy, or you do not wish to configure the devices to specify the ClariLink as their outbound proxy then the information about those devices should be entered here so that the ClariLink can intercept SIP traffic from those devices. From the device’s point of view the ClariLink will not exist, since this allows for the ClariLink to act in transparent mode for SIP traffic. You must either configure all SIP devices to use the ClariLink as their outbound SIP proxy, or you must add aSIP Redirect Entry which define all SIP traffic on your LANs. Note that for SIP devices using TCP, the source port of outbound SIP traffic will most likely be a random high port. If all traffic on the selected LAN is VoIP traffic then you can use 1-65535 as the Source Port or if all SIP traffic is sent to a specific port you can only that port in the Destination Port field.

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