Here's a checklist for switching from copper-based lines to SIP connections.
SIP is now the global standard in telephony, so organisations that are not using it will need to rethink their telephony systems. Switching to SIP is a meticulous process that can be handled by either the internal IT team or the SIP service provider. The following are some steps and considerations to take when upgrading a PABX to SIP trunk lines.
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Check SIP interoperability with existing PABX.
The organisation will be moving from copper to a digital telephone network, so the existing PABX should support SIP connections or alternative solutions will be needed.
Organisations should consider using either the Atos Unify or Office 365 platforms. The benefit of Atos Unify is that it can work with any PABX, including analog, and provide SIP connections. Alternatively, Office 365's Phone System uses Teams and Skype For Business to connect internal calls and link the organisation to the Public Switched Telephone Network (PSTN ).
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Does the PSTN contract allow a conversion to SIP?
This is a simple check usually done by the internal IT department which involves reviewing the service contract with the current PSTN provider, negotiating a new SIP contract, and comparing it with market offerings.
Using the provider that owns the physical delivery medium to the premises and provides the accompanying SIP trunking could ensure quality of service. Should the organisation opt for a different provider, then it should first verify that it can transfer its existing Direct Inward Dialling (DID) numbers to the chosen provider. Keeping these numbers is important for customer confidence and business continuity.
Considerations should also be made on how long the transfer can take. An organisation without a means to receive and send communication can be severely handicapped, if it has not been planned.
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Select a credible ISP to supply the SIP trunks.
SIP is data-based communication, it uses internet protocols to function. So the carrier, or ISP, must be in sync with the organisation's hardware and software.
The SIP software and hardware comes with its own functions and features which must be matched with the carrier so that the organisation can get the best service. As a bare minimum, the carrier must be able to provide voice services, thereafter it's a matter of weighing up each carrier's ability to support a function or feature and compare it against competitors.
An on-premise solution that is easy to install, connect, and use is Atos Unify. Alternatively, Nashua Communications has a range of credible cloud services that provide voice, messaging, video, file sharing, and other collaboration features.
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Does the ISP provide enough bandwidth for all the required SIP functions?
With SIP trunking and its accompanying features, the organisation's telephony will now rely on internet connection and reliability. Downtime or shoddy service can severely damage communication for example with dropped or low quality calls. It is important to have a quality ISP that provides guaranteed service and uptime.
There are a lot of small and large ISPs in the market to choose from. Choosing the right ISP will also save time and money in the long run. This is because switching providers after installing SIP can be an administrative headache and a costly exercise.
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Confirm the correct number of channels you require to replace the old ISDN trunks.
ISDNs use a physical line to connect PABXs to local phone exchanges, and provide simultaneous calls. The number of calls can be two at a time, 30 calls, or even more. The replacement of these channels must be matched by the new SIP service.
Although ISDNs are also a form of digital communication, SIP provides more cost and efficiency benefits. If the existing ISDN connection has sufficient capacity and speed then it can be used for SIP trunking, otherwise a dedicated broadband connection to the office will be needed.
But with a hosted SIP PABX service, after the numbers have been ported to the service, the calls are delivered to the hosted PABX instead of the local exchange. A hosted SIP has cost and redundancy benefits, most notably the ability to connect to the hosted PABX from any location and not just through a dedicated line.
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Confirm that your LAN Switches are able to handle the bandwidth QOS.
VOIP uses very low amounts of bandwidth, especially when using a few SIP trunk lines. But should the organisation require a lot of lines for VOIP along with added features from the SIP connections, then they will need a reliable broadband connection. This can be guaranteed with a QOS from the provider, but the organisation must ensure that it has LAN Switches that will support it.
LAN Switches prioritise VOIP packets over less critical traffic, along with providing emergency power to the switch and VOIP phones should there be an interruption in power supply.
Are you looking to make the move to SIP trunk lines, click here or on the link below and we’ll be in touch to help you on this journey.
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Choose between on-premise and cloud PBX SIP connections.
Switching to SIP will give the organisation the opportunity to rethink its PABX system, to either an on-premise or cloud solution.
On-premise solutions are usually more complex and expensive than cloud solutions. Most organisations opt for cloud as it is less capital intensive, easier to operate, and far more secure.
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An on-premise solution needs a Session Border Controller.
An on-premise solution must come with a Session Border Controller (SBC) system. An SBC provides security, demarcation, session management, SIP monitoring tools, SIP manipulation, and access to resolve troubleshooting problems. It can either be centralised or distributed. Its most notable functions are routing calls, granting or denying calls (e.g. dialling international numbers), and most importantly it acts as a firewall for the phone system.
Nashua Communications offers both on-premise SBC and a cloud version. A cloud solution will have a managed SBC service and is one of the many features an organisation will not have to micromanage while on the cloud.
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Installation of software and apps to accommodate the SIP solution.
SIP turns all communications into data, and this opens up boundless capabilities. Any device that can connect t o the network and host the appropriate software becomes a "soft phone": it's not necessary to use a VOIP phone to accept calls. Users can take calls from their desktop, cellphone, tablet etc. These soft phones can also handle instant messaging, conference calls, and any other installed features. This enhances productivity and provides better customer service.
To maximise the move to SIP, create a list of critical features such as unique phone numbers for each employee, an autocall router, call rules (inbound, outbound, international), contact centre, conference calling, mobile apps, and more. This will give the organisation a better understanding of its communication needs and put it in a better position to evaluate providers. Atos Unify along with Office 365 offer unified communications as a service (UCaas) solutions to organisations from small, single location outfits to large, global operations.
SIP trunking will bring new possibilities to organisations. Switching to SIP is not only smart but necessary to stay in line with global business trends. The transition can be seamless if the organisation partners with the right provider like Nashua Communications which is the premier supplier of on-premise and cloud communications solutions in South Africa and beyond.
Our Checklist is packed full of information that is aimed at helping you with everything from setting up a sip trunk to making sure that you have the correct ISP. Click on the button below to download your copy now.
