What is SIP Trunking?
SIP (Session Initiation Protocol) trunking replaces traditional phone lines (ISDN/PSTN) with internet-based voice connectivity, enabling businesses to make and receive calls over their broadband connection.
What Is SIP Trunking?
SIP trunking is a method of delivering telephone services over the internet instead of through traditional phone lines. SIP stands for Session Initiation Protocol, which is the technical standard used to set up, manage, and tear down voice and video communication sessions over the internet. A "trunk" is a term borrowed from traditional telephony, where it referred to a bundle of physical phone lines connecting your business to the telephone network.
Put simply, SIP trunking replaces those physical phone lines with a virtual connection over your internet service. Your phone system connects to a SIP trunking provider via the internet, and that provider handles the routing between your system and the wider telephone network. You keep your existing phone numbers, your calls work the same way from the user's perspective, but the underlying infrastructure is completely different.
How SIP Trunking Works
To understand SIP trunking, it helps to know what it replaces. Traditional phone systems connect to the public telephone network using physical lines -- either analogue lines (one call per line) or ISDN circuits (multiple calls per circuit). These physical connections run from the telephone exchange to your premises and cost money whether you use them or not.
The SIP Alternative
With SIP trunking, your phone system (typically an IP-PBX) connects to a SIP trunking provider over your existing internet connection. The SIP protocol handles the signalling -- setting up calls, transferring them, managing hold and conference functions -- while the actual voice audio travels as data packets alongside your email, web browsing, and other internet traffic.
Each SIP trunk can handle multiple simultaneous calls. Unlike traditional lines where you need one physical line per concurrent call, SIP trunking is more flexible. You can typically add or remove capacity on demand, paying only for what you need.
The Role of Codecs
Voice audio needs to be converted into digital data for transmission over the internet. This is done by codecs -- algorithms that compress and decompress the audio. Different codecs offer different trade-offs between audio quality and bandwidth usage. Common codecs include G.711 (high quality, higher bandwidth) and G.729 (lower quality, lower bandwidth). Most SIP trunking setups use G.711 for internal calls and G.729 for calls that traverse the wider internet.
Why Businesses Use SIP Trunking
The shift from traditional phone lines to SIP trunking has been driven by several practical advantages:
- Cost savings -- SIP trunking is typically 30 to 60 per cent cheaper than equivalent ISDN or analogue lines, especially for businesses making international calls
- Flexibility -- add or remove call capacity in minutes rather than waiting weeks for new lines to be installed
- Number portability -- keep your existing phone numbers when switching providers or moving premises
- Geographic freedom -- use local numbers from anywhere, so a business in Manchester can have a London number without a physical presence there
- Business continuity -- if your office is inaccessible, SIP trunks can be rerouted to alternative locations or mobile devices automatically
- Future-proofing -- traditional ISDN and analogue lines are being switched off across the UK and many other countries, making SIP the default going forward
In the UK specifically, the PSTN (Public Switched Telephone Network) switch-off, originally planned for 2025, means that all businesses will eventually need to move to IP-based telephony. SIP trunking is the most common migration path for organisations that want to keep their existing phone system hardware.
SIP Trunking and Telephone Payments
For businesses that take payments over the phone, SIP trunking introduces both opportunities and considerations. On the positive side, SIP-based calls are easier to integrate with cloud-based payment solutions. Because the calls are already travelling as data, routing them through a secure payment platform -- for DTMF masking or IVR-based payment capture -- is technically straightforward.
However, SIP trunking also means that voice data is travelling over the internet, which raises security considerations. Card data spoken or keyed during a call is transmitted as data packets that could theoretically be intercepted if the connection is not properly secured. Businesses should ensure their SIP trunks use TLS (Transport Layer Security) for signalling and SRTP (Secure Real-Time Protocol) for voice encryption.
Additionally, the quality of the internet connection directly affects call quality, which matters during payment transactions. If audio drops out while a customer is entering card details via their keypad, digits can be missed, leading to failed transactions and frustrated customers. Ensuring sufficient bandwidth and quality of service (QoS) settings for voice traffic is essential.
Practical Considerations
- Assess your internet bandwidth before switching. Each concurrent call requires roughly 80-100 Kbps of bandwidth. Make sure your connection can handle your peak call volume with room to spare.
- Implement quality of service (QoS) on your network to prioritise voice traffic over less time-sensitive data like email and file downloads.
- Choose a provider that offers TLS and SRTP for encrypted communications, especially if you handle payments or other sensitive information.
- Keep a backup internet connection. If your primary connection fails, your phones go down. A second connection from a different provider provides resilience.
- Plan your migration carefully. Moving from traditional lines to SIP trunking should be done in stages, with thorough testing at each step.
SIP trunking is a mature, well-understood technology that has become the standard way businesses connect to the telephone network. Whether you are replacing aging ISDN lines or building a new telephony setup from scratch, SIP trunking offers the flexibility, cost savings, and future-readiness that traditional phone lines simply cannot match.
Paytia's PCI DSS Level 1 certified platform incorporates sip trunking as part of its thorough security approach. By processing phone payments through DTMF suppression, Paytia ensures card data is protected at every stage.
Frequently Asked Questions
What is sip trunking?
SIP (Session Initiation Protocol) trunking replaces traditional phone lines (ISDN/PSTN) with internet-based voice connectivity, enabling businesses to make and receive calls over their broadband connection.
Why is sip trunking important for PCI DSS?
PCI DSS requires organisations to implement sip trunking as part of their security controls for protecting cardholder data.
How does Paytia handle sip trunking?
Paytia implements sip trunking as part of its PCI DSS Level 1 certified infrastructure, ensuring all phone payments are processed securely.
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