What is SIP Trunking?
Chances are you use your work phone on a daily basis and rarely stop to think about what makes your calls or messaging work. However, if you are shopping for a phone system, you may be greeted with new terms, such as VoIP, SIP, and SIP trunking. These terms may be unfamiliar to you, but they make up the backbone of many modern telephony systems.
Traditional telephones make calls by connecting to the Public Switched Telephone Network (PSTN). The PSTN is a network of phone lines, satellites, cables, and more that connect calls through operators, or telephony service providers, both on a local and international scale. However, IP phones are like small computers that use an ethernet cable to connect to your company’s Private Branch Exchange (PBX). This is your private telephone network that is set up for your phone system with either an on-site server or with a hosted PBX service that is maintained remotely. Phone calls, messaging, or video transfers are then completed using Voice over Internet Protocol (VoIP).
VoIP is a way to send voice, and even video data, over the internet. SIP trunking is a method of VoIP intended specifically for sending and receiving call data using the Session Initiation Protocol (SIP). SIP is the protocol that signals the data connection between two endpoints, allows for changes to the session, and terminates the session when the call is complete.
Simply put, VoIP is a way to transfer voice and video data over the internet, SIP trunking is a method of VoIP meant specifically for calls, and SIP is the protocol that initiates, manages, and ends a calling session.
What are the benefits of SIP Trunking?
Cut Costs & Increase Scalability
Many companies already use phone systems that connect their phones to a Local Area Network (LAN). Telephone connections can be made to the PBX server using a LAN, making the move to SIP trunking affordable and painless. Because SIP trunking is installed using the internet, there is no additional hardware or wiring required, reducing the costs that you would normally see with implementing a new traditional phone system.
You can even save money when updating a legacy phone system to use IP phones and VoIP, as you can connect SIP trunking to your analog system using a gateway, reducing the need to replace your telephony infrastructure.
SIP trunking’s affordable pricing models also permit you to only purchase the number of channels you need with one low monthly rate (Channelized Rate Plan) or only pay for the minutes you use each month (Metered Rate Plan) allowing for flexibility in your monthly call volume. This permits you to cut unnecessary spending and wasted resources, and it lets you combine your voice and data plans into one network. This makes scaling your business simple and affordable, as it is easy to add more channels or use more minutes when your number of phone system users grows.
Every successful business incorporates a level of redundancy and has a disaster recovery plan in place. If you have a backup plan for your business operations, shouldn’t you have one for your phone system, too? Fortunately, SIP trunking offers a level of redundancy out of the box. When a phone call fails to reach your IP phone, SIP trunking providers can reroute your PBX services to a backup line, or even to your mobile phone. This allows your phone system to demonstrate optimum reliability and allows your business to never miss a call, no matter the situation.
Is your network ready?
Because VoIP systems can not transfer call data without an internet connection, it is important to make sure your network is prepared to handle a SIP trunking installation. There are three things to consider when assessing your network’s VoIP readiness: upload and download speeds, latency, and jitter.
Upload and download speeds refer to the bandwidth speeds of your network’s LAN, internet connection method, or internet provider’s network. You may need to upgrade your internet service to maintain the recommended bandwidth of your network.
Latency refers to the amount of time it takes for data that is sent from one end-point to reach the receiving endpoint. For example, latency in a call session would measure the amount of time it takes for a person’s voice message to reach the person on the other end of the phone call. Although latency does not directly affect the quality of a phone call connection, it can affect the calling experience because the delay may become evident to callers.
Jitter refers to the varying degrees of delay of voice data transfer in a phone call session. Jitter is often influenced by network traffic and can result in poor call quality.
The following equation can be used to determine the necessary bandwidth to support your calls:
number of concurrent calls at your company’s peak X 100 kilobits per second = bandwidth in Megabits per second (Mbps) needed for calls
You will also need to consider the amount of bandwidth you use for other business processes, such as sending and receiving emails, videos, messages, and more. Digium offers a simple Network Assessment Tool that can give you an idea about whether your network is prepared to move to a VoIP-based phone system using SIP trunks.
Check Out Digium’s Network Assessment Tool
Long-gone are the days of telephone switchboard operators manually routing your calls. SIP trunking is a method of VoIP, which sends voice and video data over the internet using the SIP protocol to initiate, manage, and terminate calling sessions. Although SIP trunking requires a strong and stable internet network connection, it can benefit a business by cutting phone system costs and providing an extra level of redundancy to keep operations moving when disaster strikes. It requires little hardware and wiring, and offers a scalable solution to give you flexibility in your phone system.
Learn more about SIP trunking, and watch an on-demand webinar to see how SIP trunking can save you money and improve business – Watch the Webinar Now