SIP trunking is a method of delivering telephone calls over the internet rather than through traditional copper phone lines. Businesses use SIP trunks to connect their phone system — whether on-premise PBX or cloud-based — to the public telephone network, replacing legacy physical circuits with internet-based voice channels.
What is Session Initiation Protocol (SIP)?
SIP stands for Session Initiation Protocol. It is an application-layer signalling protocol — standardised by the IETF in RFC 3261 — that establishes, modifies, and ends communication sessions such as voice calls, video calls, and messaging over IP networks.
A "SIP trunk" is a virtual phone connection provided by an Internet Telephony Service Provider (ITSP). Unlike a physical phone line, a SIP trunk exists entirely in software and can carry multiple concurrent calls over a single broadband connection.
SIP vs. Legacy PRI Lines
Historically, a Primary Rate Interface (PRI) circuit provided 23 usable voice channels (in the US) or 30 channels (in Europe). If a business needed more concurrent calls than a single PRI could support, it had to purchase a complete second circuit — regardless of how few of those additional channels were actually needed.
SIP trunking removes this fixed-increment constraint. Wholesale voice via SIP allows organisations to provision exactly the number of concurrent call sessions they require. Capacity can typically be adjusted through a provider portal or API rather than through physical infrastructure changes.
How SIP Communications Actually Work
When a call is initiated using SIP, the originating device — called a User Agent — sends a SIP INVITE message to a proxy server or SIP registrar. The INVITE includes an SDP (Session Description Protocol) payload that describes media capabilities: supported audio codecs such as G.711, G.722, or Opus, and the network address where media should be delivered.
Once the destination accepts, a 200 OK response is returned and the call is established. The actual voice audio travels separately via RTP (Real-Time Transport Protocol) rather than through SIP signalling. This separation of signalling and media is a core characteristic of the SIP architecture.
Cost Considerations
SIP trunking can reduce telephony costs compared to legacy PRI or ISDN lines, particularly for organisations with:
- High volumes of international calls, where per-minute rates via SIP are often substantially lower than PSTN alternatives.
- Variable call volumes, because SIP capacity can scale without purchasing fixed hardware increments.
- Multiple locations, where internal calls between sites may travel over the IP network rather than incurring PSTN charges.
The actual cost saving depends on your current provider, call volumes, destination countries, and whether any hardware changes are required. Contact your ITSP or contact MCC to discuss expected costs for your specific usage pattern.
Geographic Flexibility
Because SIP trunking decouples phone numbers from physical locations, a business can provision virtual phone numbers in multiple countries and route incoming calls to any destination — a central office, a distributed team, or a cloud contact centre — without changing infrastructure in each location.
This is useful for businesses with international customers who want a local number to call, or for teams operating across multiple offices.
Security and Encryption
SIP communications can be secured using standard encryption protocols. Well-configured SIP trunks use:
- TLS (Transport Layer Security) to encrypt SIP signalling, protecting call metadata such as caller ID and destination numbers.
- SRTP (Secure Real-Time Transport Protocol) to encrypt the voice audio stream itself.
Security posture varies between providers. When evaluating an ITSP, ask specifically whether TLS and SRTP are enabled by default, or whether they require configuration.
It is worth noting that legacy PSTN copper lines also carry security risks and are not inherently more secure than properly encrypted SIP. However, the security of a SIP deployment depends on correct implementation, firewall configuration, and SIP-aware intrusion prevention to guard against toll fraud.
Microsoft Teams Direct Routing
Microsoft Teams includes Direct Routing — a feature that allows organisations to connect their own SIP trunk to Teams, enabling Teams users to make and receive external calls through their preferred carrier rather than Microsoft's native calling plans.
This requires a Session Border Controller (SBC) certified by Microsoft, which sits between the SIP trunk and the Teams environment. Direct Routing can be cost-effective for organisations with high international call volumes or complex call routing requirements that Teams' native plans do not support.
Business Continuity
SIP trunking can support business continuity planning because inbound calls are routed by software rather than delivered to a fixed physical location. If a primary office loses internet connectivity or becomes inaccessible, many SIP providers allow failover rules to redirect calls to another number, another site, or a mobile device.
The effectiveness of failover depends on the provider's configuration options and the organisation's call routing setup. This should be tested before relying on it in practice.
Choosing an ITSP
When evaluating SIP trunk providers, useful questions include:
- Where are their network points of presence, and are they geographically relevant to your call destinations?
- Do they have direct interconnects with major carriers, or do calls transit multiple intermediaries?
- What codecs do they support, and are HD voice codecs (G.722, Opus) available?
- What failover and redundancy options do they offer?
- How is capacity provisioned — fixed channels or elastic concurrent call sessions?
- What SLA do they offer, and what does it cover?
MCC offers wholesale voice termination and cloud business phone services built on SIP. Contact us to discuss your requirements.
Interested in SIP trunking or wholesale voice for your business?
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