How is important SIP trunking in Skype for Business Server
Tags: protocol, skype, business, communication
Published: 2021-04-26 16:20:17

Session Initiation Protocol (SIP) is used to initiate and manage Voice over IP (VoIP) communications sessions for basic telephone service and for additional real-time communication services, such as instant messaging, conferencing, presence detection, and multimedia. This section provides planning information for <a href="http://www.everydayvoip.uk">implementing SIP trunks</a>, a type of SIP connection that extends beyond the boundary of your local network.

<h2>What is SIP Trunking?</h2>
A SIP trunk is an IP connection that establishes a SIP communications link between your organization and an Internet telephony service provider (ITSP) beyond your firewall. Typically, a SIP trunk is used to connect your organization's central site to an ITSP. In some cases, you may also opt to use SIP trunking to connect your branch site to an ITSP.

<h2>Cost Savings</h2>
The cost savings associated with SIP trunking can be substantial:
Long-distance calls typically cost much less through a SIP trunk.
You can cut manageability costs and reduce the complexity of deployment.

Basic rate interface (BRI) and primary rate interface (PRI) fees can be eliminated if you connect a SIP trunk directly to your ITSP at a significantly lower cost. In TDM trunking, service providers charge for calls by the minute. The cost of SIP trunking may be based on bandwidth usage, which you can buy in smaller, more economical increments. (The actual cost depends on the service model of the ITSP you choose.

<h2>Expanded VoIP Services</h2>
Voice features are often the primary motivation for deploying SIP trunking, but voice support is just the first step. With SIP trunking, you can extend VoIP capabilities and enable Skype for Business Server to deliver a richer set of services. For example:
Enhanced presence detection for devices that are not running Skype for Business Server can provide better integration with mobile phones, enabling you to see when a user is on a mobile phone call.
E9-1-1 emergency calling enables the authorities who answer 911 calls to determine the caller's location from his or her telephone number.

<h2>SIP Trunks vs. Direct SIP Connections</h2>
The term trunk is derived from circuit-switched technology. It refers to a dedicated physical line that connects telephone switching equipment. Like their predecessor, time-division multiplexing (TDM) trunks, SIP trunks are connections between two separate SIP networks—the Skype for Business Server enterprise and the ITSP. Unlike circuit-switched trunks, SIP trunks are virtual connections that can be established over any of the supported SIP trunking connection types.

<h2>How do I implement SIP Trunking?</h2>
To implement SIP trunking, you must route the connection through a Mediation Server, which acts as a proxy for communications sessions between Skype for Business Server clients and the service provider and transcodes media, when necessary.

Each Mediation Server has an internal network interface and an external network interface. The internal interface connects to the Front End Servers. The external interface is commonly called the gateway interface because it has traditionally been used to connect the Mediation Server to a public switched telephone network (PSTN) gateway or an IP-PBX. To implement a SIP trunk, you connect the external interface of the Mediation Server to the external edge component of the ITSP. The external edge component of the ITSP could be a Session Border Controller (SBC), a router, or a gateway.

<h2>Centralized vs. Distributed SIP Trunking</h2>
Centralized SIP trunking routes all VoIP traffic, including branch site traffic, through your central site. The centralized deployment model is simple, cost-effective, and is generally the recommended approach for implementing SIP trunks with Skype for Business Server.

Distributed SIP trunking is a deployment model in which you implement local SIP trunks at one or more branch sites. VoIP traffic is then routed from the branch site directly to a service provider without going through the central site.

Distributed SIP trunking is required only in the following cases:
The branch-site requires survivable phone connectivity (for example, if the WAN goes down). This requirement should be analyzed for each branch site; some of your branches may require redundancy and failover, whereas others may not.

Resiliency is required between two central sites. You need to make sure that a SIP trunk terminates at each central site. For example, if you have Dublin and Tukwila central sites and both use only one site's SIP trunk, if the trunk goes down, the other site's users cannot make PSTN calls.

The branch site and central site are in different countries/regions. For compatibility and legal reasons, you need at least one SIP trunk per country/region. For example, in the European Union, communications cannot leave a country/region without terminating locally at a centralized point.

The decision whether to deploy centralized or distributed SIP trunking requires a cost-benefit analysis. In some cases, it may be advantageous to opt for the distributed deployment model even if it is not required. In a completely centralized deployment, all branch site traffic is routed over WAN links. 
Instead of paying for the bandwidth required for WAN linking, you may want to use distributed SIP trunking. For example, you may want to deploy a Standard Edition server at a branch site with the federation to the central site, or you may want to deploy a Survivable Branch Appliance or a Survivable Branch Server with a small gateway.