Let’s face it; the success of your business relies heavily on the quality of communication. Effective communication can help build better relationships between you and your employees, customers, prospects, vendors, partners and co-workers.
The result is improved employees’ morale, fewer errors and more innovations that help take you closer to your goals.
One of the easiest ways to equip your staff to communicate more effectively is to upgrade to SIP phone systems. SIP trunking is cost effective, scalable and ensures high-definition call quality.
With ISDN lines set to be phased out by 2025, one of the common questions we get from our prospective clients here at Metrotech is: Is my phone system SIP-enabled?
If you’re planning to move your existing PBX telephony system to the cloud, then you’ve come to the right place.
In this article, you’ll learn what a SIP phone system is, how SIP calls work, what you need to upgrade to a SIP trunk and everything in between. Let’s dive in.
SIP is an acronym for Session Initiation Protocol. It is a type of VoIP (Voice Over Internet Protocol) phone call service for managing various multimedia sessions, such as voice and video calls.
SIP and VoIP are sometimes used interchangeably. However, SIP is actually a signalling protocol within a VoIP phone system.
Unlike VoIP, SIP is not limited to just voice; it can transmit all types of media and also manage all the call requirements, from initiating the call session to terminating it.
There are many other types of VoIP communication protocols besides SIP. But SIP is considered more viable because it allows for a higher call quality, better reporting and lower lifetime costs.
Deploying SIP trunks on compatible PBX telephone systems replaces the need for legacy lines or Primary Rate Interface (PRI). SIP phones open up businesses to opportunities for voice calls, video conferencing, instant messaging and more.
SIP works alongside several other protocols to carry voice data during a VoIP call. One such protocol is the session description protocol or SDP.
As SIP exchanges signalling details between IP points, SDP carries session-related information over the network. These session descriptions are conveyed as a payload of SIP messages.
Before transporting this information, SIP encodes voice using codecs, which then convert audio signals into binary data. There are many codecs used for this purpose.
But two of the most popular are:
Works with uncompressed digital voice. Offers better audio quality but consumes more bandwidth.
Works with compressed voice. Offers lower audio quality and reduced bandwidth consumption.
The real-time transport protocol (RTP) carries encoded packets of audio data for real-time audio streaming and video data. Unlike SDP, RTP is independent of SIP.
Together with the RTP control protocol (RTCP), RTP delivers information related to the service quality. This can include the number of packets exchanged and round-trip lag time. Worth mentioning is that RTCP sessions run parallel to the RTP data stream.
Finally, the transport layer protocols deliver RTP, RTCP and SIP data packets to their destinations. Two of the most commonly used transport protocols are transmission control protocol (TCP) and user datagram protocol (UDP).
UDP focuses on transmitting data packets to their destination as soon as possible. The ability to deliver data in real time makes UDP more suitable for VoIP calls than TCP.
SIP requires a phone system that establishes communications over the internet. If your PBX system was made in the 90s or later, there’s a good chance that it’s already SIP-enabled.
Phones made for SIP trunking will usually have an Ethernet jack or data jack on the back. You can also check the user manual for words like “IP Calling”, “SIP-enabled” or “SIP trunk Setup”.
You can still have SIP trunking deployed on an older phone system. But that will require a few extra steps. For example, you’ll need an Analogue Telephony Adapter (ATA), a device that converts analogue signals to digital.
Other requirements to use a SIP phone include:
SIP is an NBN-ready solution. But it’s OK to use your existing internet provider and make the switch later.
Newer systems come with licenses already. If you have an older system, consult your provider and arrange for a license.
Once everything checks out, you can have SIP trunks connected to your IP-enabled PBX hardware. Another way to enable a VoIP telephone system is through a hosted IP PBX.
Using a hosted VoIP telephone service is advantageous in many ways. First, you get to switch to cloud calling without obtaining additional IP-PBX hardware.
And secondly, you don’t have to go through the hassle of setting up the SIP trunking yourself. Your phone service provider will pre-configure the VoIP system for you.
The result? The possibility to bypass expensive traditional PRI lines and achieve a more tailored solution for your business. Additional call management features like call forwarding, auto attendance, voicemail, and much more can also be integrated.
Enabling a SIP trunk is easy if you partner with the right phone service provider like Metrotech. Here are some steps involved when transitioning to SIP trunking:
Newer PBX systems will probably be SIP-enabled and ready for SIP trunking. Older systems will need an ATA device to convert analogue signals to digital. ATA is inexpensive and readily available from companies such as Cisco and Grandstream.
How you plan to use your phones determines the number of channels you need. Having a channel for each employee doesn’t make sense unless everyone will be on the phone concurrently. For most businesses, one channel is enough for three to four employees.
SIP trunking sends calls to the Public Switched Telephone Network (PSTN) through the internet. Check to make sure that your internet bandwidth is enough to ensure quality calls. Modern business internet connections such as cable, T1 or DSL should work just fine.
QoS stands for quality of service. Your router should have QoS enabled to prioritise voice traffic over data traffic. Doing so ensures that certain activities like video streaming or downloading do not impact voice quality.
You need to complete some PBX set up to enable SIP trunking. Choosing a SIP trunking provider that configures PBX for you is a smart idea. That way you won’t need to worry about running into issues in the process.
The last step is to perform a test drive to ascertain that your new SIP solution is working and aligned with your business goals. Your SIP trunking provider should give you a free trial to make sure everything is working before you commit.
Upgrading to SIP phones comes with plenty of benefits for businesses of all sizes. Here are the advantages of using SIP phone systems:
VoIP systems offer affordable liner rentals and low monthly call rates. Switching to a VoIP phone system can save your business up to 50% on phone bills.
A SIP phone makes it possible to connect all employees together, including remote staff. You can also make phone calls anywhere in the world.
Some VoIP solutions integrate with other business applications like Outlook contacts and CRMs. Instant delivery of features allows for rich enterprise telephony capabilities.
SIP calls are encrypted for increased call security. There’s also the opportunity to separate business calls from personal calls.
As we’ve established, staying old-school with your business phone system can be a costly decision. SIP trunking is designed to replace old-fashioned PRI technology that relies on traditional copper lines.
Transitioning to SIP trunking can help your business with scaling needs. Especially if you plan on expanding offices or adding multiple business locations.
Thinking about switching to a SIP phone system but unsure where to start? Don’t know whether your existing phone system is SIP-enabled or needs ATA? Get in touch with our experts today.
With 10+ years of expertise, our UK-based experts are on standby to give you tailored answers you’re looking for. Our expert VoIP telephone solutions will meet the needs of businesses across all industries.