Create a new collection of trunk configuration settings in Skype for Business Server 2015
Summary: Learn how to create a new collection of trunk configuration settings by using the Skype for Business Server Control Panel.
SIP trunk configuration settings define the relationship and capabilities between a Mediation Server and the Public Switched Telephone Network (PSTN) gateway, an IP-Public Branch eXchange (PBX), or a Session Border Controller (SBC) at the service provider. These settings do such things as specify:
Whether media bypass should be enabled on the trunks.
The conditions under which Realtime Transport Control Protocol (RTCP) packets are sent.
Whether or not Secure Realtime Transport Protocol (SRTP) encryption is required on each trunk.
When you install Skype for Business Server, a global collection of SIP trunk configuration settings is created for you. In addition, administrators can create custom setting collections at the site scope or at the service scope (for the PSTN gateway service, only).
When creating SIP trunk configuration settings using Skype for Business Server Control Panel, the following options are available to you.
|UI Setting||PowerShell Parameter||Description|
||Unique identifier for the collection. This property is read-only; you cannot change the Identity of a collection of trunk configuration settings.
||Provides a way for administrators to store addition information about the settings (for example, the purpose of the trunk configuration).
|Maximum early dialogs supported
||The maximum number of forked responses a PSTN gateway, IP-PBX, or SBC at the service provider can receive to an Invite that it sent to the Mediation Server.
|Encryption support level
||Indicates the level of support for protecting media traffic between the Mediation Server and the PSTN Gateway, IP-PBX, or SBC at the service provider. For media bypass cases, this value must be compatible with the EncryptionLevel setting in the media configuration. Media configuration is set by using the New-CsMediaConfiguration and Set-CsMediaConfiguration cmdlets.
Allowed values are:
Required: SRTP encryption must be used.
Optional: SRTP will be used if the gateway supports it.
Not Supported: SRTP encryption is not supported and therefore will not be used.
SRTPMode is used only if the gateway is configured to use Transport Layer Security (TLS). If the gateway is configured with Transmission Control Protocol (TCP) as the transport, SRTPMode is internally set to Not Supported.
|If set to Enable sending refer to the gateway, indicates that the trunk supports receiving Refer requests from the Mediation Server.
If set to Enable refer using third-party call control, indicates that the 3pcc protocol can be used to allow transferred calls to bypass the hosted site. 3pcc is also known as "third party control," and occurs when a third-party is used to connect a pair of callers (for example, an operator placing a call from person A to person B).
|Enable media bypass
||Indicates whether media bypass is enabled for this trunk. Media bypass can only be enabled if Centralized media processing is also enabled.
|Centralized media processing
||Indicates whether there is a well-known media termination point. (An example of a well-known media termination point would be a PSTN gateway where the media termination has the same IP as the signaling termination.)
|Enable RTP latching
||Indicates whether or not the SIP trunks support RTP latching. RTP latching is a technology that enables RTP/RTCP connectivity through a NAT (network address translator) device or firewall.
|Enable forward call history
||Indicates whether call history information will be forwarded through the trunk.
|Enable forward P-Asserted-Identity data
||Indicates whether the P-Asserted-Identity (PAI) header will be forwarded along with the call. The PAI header provides a way to verify the identity of the caller.
|Enable outbound routing failover timer
||Indicates whether outbound calls that are not answered by the gateway within 10 seconds will be routed to the next available trunk; if there are no additional trunks then the call will automatically be dropped. In an organization with slow networks and gateway responses, that could potentially result in calls being dropped unnecessarily.
|Associated PSTN usages
||Collection of PSTN usages assigned to the trunk.
|Translated number to test
||Phone number that can be used to do an ad hoc test of the trunk configuration settings.
|Associated translation rules
||Collection of phone number translation rules that apply to calls handled by Outbound Routing (calls routed to PBX or PSTN destinations).
|Called number translation rules
||Collection of outbound calling number translation rules assigned to the trunk.
|Phone number to test
||Phone number that can be used to do an ad hoc test of the translation rules.
||Indicates that the phone number to test is the phone number of the caller.
||Indicates that the phone number to test is the phone number of the person being called.
The Skype for Business Server CsTrunkConfiguration cmdlets support additional properties not shown in Skype for Business Server Control Panel. For more information, see the help topic for the New-CsTrunkConfiguration cmdlet.
To create new trunk configuration settings by using Skype for Business Server Control Panel
In Skype for Business Server Control Panel, click Voice Routing, and then click Trunk Configuration.
On the Trunk Configuration tab, click New, and then click Site trunk to create the new settings at the site scope, or Pool trunk to create the new settings at the service scope.
In the Select a Site or the Select a Service dialog box (the dialog box that appears will depend on whether you are creating site-scoped or service-scoped settings) select the location for the new configuration settings and then click OK. If the dialog box is blank, that means there is no place to create the new settings; for example, if the Select a Site dialog box is blank that means that all of your sites have already been assigned a collection of trunk configuration sites, and each site (and each service) can only host one such collection. In that case, you can either delete the existing collection and create a new collection, or simply modify the existing collection.
In the New Trunk Configuration dialog, make the appropriate selections and then click OK.
The State property for the collection will be updated to Uncommitted. To commit the changes, and to delete the collection, click Commit and then click Commit All.
In the Uncommitted Voice Configuration Settings dialog box, click OK.
In the Skype for Business Server Control Panel dialog box click OK.