What is SIP Trunking?

SIP Trunking

SIP, or Session Initiation Protocol, is the standard communications protocol for voice and video in a Unified Communications (UC) solution across a data network. A SIP trunk replaces the need for traditional analog, T1-based Public Switched Telephone Network (PSTN) connections with termination instead provided over a company’s public or private Internet connection through a SIP provider. These SIP providers, often referred to as Internet Telephony Service Providers (ITSP), provide PSTN service on a per minute or channelized pricing model.

The per-minute pricing model is fairly self-explanatory, with a set rate per minute of usage. A channelized pricing model typically provides nearly unlimited minutes on a set number of channels, or call paths. For example, a company can purchase 10 channels and make use of unlimited minutes on those channels, but can only have 10 simultaneous calls.

Many companies already use VoIP within their PBX on the Local Area Network (LAN) to connect to IP phones. SIP Trunking also uses VoIP to take advantage of shared lines, such as a company’s Internet connection, to allow more flexibility in communications. Traditional legacy systems, that aren’t already VoIP-capable, can be connected using common VoIP gateways to take advantage of SIP trunking and reap the significant cost benefits.

The Benefits of SIP Trunking

The cost savings and communications benefits of SIP trunking are substantial. Your company is most likely experiencing high costs with an existing PBX, while still having the constraints of the limited communications technology provided by your current Telco. High costs may be incurred through a combination of monthly phone bills, which include charges for incoming phone lines, long distance charges and IT and maintenance fees, all of which can be drastically reduced or eliminated by a SIP Trunking provider.

SIP Trunking allows companies to only pay for the number of lines they need as opposed to getting locked in to excess analog lines or partially-used T1s and PRIs. The savings are realized either by purchasing only the necessary number of channels, or by paying only for minutes used. This allows companies to make more efficient use of communications costs and reduce or eradicate wasted resources.

SIP Trunking eliminates the physical connection to a phone company. There are no hardware, wiring, or circuit boxes to maintain for connection to the PSTN. Reducing multiple phone lines into a single point of entry drastically reduces charges for incoming lines and the IT cost associated with the maintenance of those lines. (Some organizations prefer to maintain standard lines for faxes and alarms.)

A phone number, or Direct Inward Dialing number (DID), is less expensive when purchased with a SIP Trunk. Traditionally, when a DID is obtained from a phone company, charges are applied for the DID, IT and maintenance services, and the hardware connecting the shared physical lines or channels. A DID provided without these infrastructure costs is more affordable.

SIP Trunking with VoIP increases reliability of services by providing a level of redundancy. When system failures and emergencies occur, SIP Trunking providers can reroute services to a redundant data line or forward the PBX to mobile phones to keep your business up and running.

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Considerations Needed for SIP Trunking

To prepare your company for SIP Trunking, you need to assess the usage rates of your business communications. Specifically, you should consider how many people are on the phone at the same time during your busiest hours. The answer to this question will determine how many channels you will need. Remember, SIP trunks allow for quick and easy scaling, so you may add or remove channels as needed, if you under or over estimate. You can also decide which method to purchase SIP services. Some organizations may benefit purchasing a set number of trunks while others would benefit from a set number of minutes.

Network considerations that must be examined include total available bandwidth, Quality of Service (QoS), and firewalls. Upgrading your Internet connection may be necessary to ensure sufficient bandwidth to carry UC on top of typical Internet usage for your company.

Use the following simple equation to determine the necessary bandwidth to support your calls:

(number of concurrent calls at your company’s peak) x 85kilobits per second = bandwidth in Megabits per second (Mbps) needed for calls

Equally important to bandwidth is QoS. QoS prioritizes your voice traffic and ensures that your phone calls are going to get the bandwidth needed, regardless of what else is happening on the network. The vast majority of business grade network routers will provide QoS for your network.

A firewall is critical to maintain security both within a LAN and Wide Area Network (WAN). Though firewalls are a critical component to any business network they must be configured correctly to work well with SIP trunking.

For safety, it is essential to add Enhanced 911 (E911). E911 is a feature of the 911 emergency-calling systems that places VoIP emergency callers with the appropriate resources by associating a physical address with the calling party's telephone number.

Potential Network Issues

On occasion, a PBX with SIP Trunks may experience potential network issues such as Jitter, Latency, and Packet Loss. Jitter can occur when voice packets arrive with varying delays. This is typically caused with changes in network traffic. Latency (delay) doesn’t typically cause audio quality issues, but when the latency is over 150ms, the delay is noticeable to your users. Packet loss, the failure of one or more transmitted packets to arrive at their destination, can cause noticeable effects in all types of digital communications.

These infrequent network issues can usually be fixed by QoS, reducing the amount of traffic your network equipment is handling, and simple network tuning.

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