What Is SIP Trunking
What is SIP Trunking
Session Initiation Protocol, or SIP, is the fundamental communications protocol for voice and video across a network. SIP trunking is designed to replace conventional analog, T1-based PSTN (public switched telephone network) telephone services with termination that is delivered over a public or private Internet connection. This service is typically provided by a VOIP or SIP solutions provider on a per minute or packaged pricing structure.
Per minute billing refers to a service that is billed based on a set charge per minute of talk time. Packaged pricing, however, is much more practical for business use. SIP providers offering this type of pricing model typically allow customers to enjoy an unlimited amount of minutes, on a predetermined number of call channels, for a fixed monthly fee. For example, an organization can purchase 15 channels, and utilize an uncapped amount of minutes on each channel, however, they will only be able to have 15 simultaneous calls.
Several organizations already make use of VoIP (Voice-over-IP) to communicate. SIP trunking works in a similar manner by utilizing VoIP to leverage shared phone networks, like a business Internet connection, to create a more flexible communications environment. Legacy systems, and dated communications platforms that are not VoIP-ready, can be fitted with gateways that enable the system to harness the advantages of SIP trunking.
The Advantages of SIP Trunking
The benefits of SIP trunking are substantial. Particularly when taking into account the strong potential for cost savings. Businesses still deploying standard PBX solutions often report high costs and difficulty keeping pace with competitors who have more advanced communications systems. The most common causes of higher charges under the conventional telco model include things like charges for incoming phone lines, long distance calling charges, IT costs, maintenance fees and system management costs. Most of these fees can be significantly reduced or even eliminated by migrating to a SIP-powered platform.
In a typical SIP trunking scenario customer pay only for the number of lines (or seats) that they require, unlike the ‘locked-in’ plans with extra analog line or unused T1s / PRIs offered by standard phone service provider.
SIP trunking removes the requirement for a physical connection to a telephone provider. Furthermore, there a no requirements for hardware, copper wiring, bulky phone closest or even connections to the PSTN. Limiting the number of phone lines into a single entry point significantly lessens charges for incoming calls and also reduces (and in some situations eliminates) IT management costs for line maintenance.
Direct Inward Dialing (a phone number) is less costly when obtained from a SIP trunking provider. In the majority of cases, when a phone number is purchased from a traditional phone service provider, there are costs incurred for the number, IT and maintenance services, as well as for all of the hardware needed to connect shared lines or multiple channels.
Combining the technologies of SIP trunking and VoIP optimizes service reliability by enhancing redundancy. In the event of system failure, SIP trunking solutions providers have the ability to route services to another number or location, or forward incoming calls to any predefined number, voicemail or device so that business operations continue.
What is needed to prepare for SIP Trunking?
In order to prepare your business for SIP trunking it is necessary to first assess your typical communication usage rates. More specifically, you will need to take into account how many users are on the phone simultaneously at peak. This will give you a better understanding of how may channels you will require. Keep in mind that SIP trunking service are designed to allow for easy scalability, meaning you can add or remove channels with minimal effort.
It is also necessary to investigate the capabilities of the corporate network, taking care to examine things like Quality of Service, firewalls and bandwidth availability. Some businesses find that they need to upgrade their Internet service in order to receive optimal results from SIP and VoIP.
Here is an easy equation to help you determine how much bandwidth will be required to support your communications needs.
(number of concurrent voice calls at peak) x 85kbps = bandwidth in mbps required for high quality calls
Quality of Service, or QoS, is just as important as bandwidth. QoS will prioritize voice calls to ensure that they are receiving the bandwidth required despite whatever other activities are happening on the network.
Possible network issues
In certain situations, a PBX with SIP trunk may experience issues such as Latency, Jitter, and Packet Loss. These types of issues have a direct impact on call quality and clarity. However, they can typically be resolved with QoS prioritization, and in some cases, upgrading the network equipment.