What Is SIP Trunking?
How It Works & Why Businesses Use It

SIP trunking connects a business phone system to the public telephone network over an IP connection instead of copper lines or PRI circuits. A provider, called an ITSP, supplies the trunk. Each call in progress uses one channel, and numbers, capacity and routing are managed in software rather than in hardware. Most businesses use it to replace PRI lines, or to give an on-premise PBX or Microsoft Teams a way to make and receive outside calls.

The parts that decide whether a SIP trunk works well for you are not in the definition: how many channels you buy, how much bandwidth each call takes, how your caller ID is signed, and what the provider does when a route fails. Those get their own sections below.

What is Session Initiation Protocol (SIP)?

SIP stands for Session Initiation Protocol. The IETF's RFC 3261 describes it as "an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants". In practice those sessions are 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. For a full SIP trunking vs PRI comparison including migration steps, see our dedicated guide.

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.

SIP trunk, PRI or hosted VoIP: which one do you need?

The three get confused because all of them end with a working phone number. What differs is where your phone system lives and how capacity is bought.

If you are deciding between keeping a PBX and moving to the cloud, UCaaS vs on-premise PBX covers that choice. A SIP trunk only makes sense if something on your side still needs a trunk to plug into.

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.

The split matters when something breaks. A call that rings but has no audio is almost always a media problem (RTP blocked by a firewall or NAT), not a SIP problem, because the signalling clearly got through. Our VoIP troubleshooting checklist works through those cases.

How many SIP trunk channels do you need?

A channel carries one call at a time, so the number you need is set by your busiest hour, not by headcount. For example, twenty staff who rarely overlap on the phone may need far fewer than twenty channels, while a ten-seat outbound team that is always dialling needs at least ten.

Traffic engineers size trunks with the Erlang B formula. It takes the traffic in your busiest hour, measured in erlangs (calls per hour × average call length in hours), and a target blocking rate: the share of calls that would find every channel busy. One percent is a common target.

The figures below are illustrative, worked with Erlang B at a 1% blocking target. Put in your own busy-hour numbers from your call records before buying.

Two things the formula does not know. Erlang B assumes a blocked caller gives up rather than redialling at once, so a team that retries busy calls needs some headroom above these figures. And a dialer that launches calls in parallel is a different case: size it from the dialer's own maximum of simultaneous calls, not from the busy-hour average.

How much bandwidth does a SIP call use?

Bandwidth depends on the codec. Cisco's per-call bandwidth tables give these figures for one direction of a call at the common 20 ms packet size, including IP, UDP, RTP and Ethernet headers:

• G.711 (64 kbps codec): about 87.2 kbps per call.
• G.729 (8 kbps codec): about 31.2 kbps per call.

Multiply by your channel count to get the worst case. Thirteen G.711 calls at once come to roughly 1.1 Mbps in each direction. The upload direction is usually the constraint on business broadband, and the figure should fit with room to spare after everything else on the line. If it does not, either the link needs to grow or voice needs priority (QoS) over other traffic.

G.729 saves bandwidth by compressing the audio, and compression can distort fax signals and in-band DTMF tones. If you send faxes or rely on keypad tones reaching an IVR, confirm with the provider which codec and DTMF method the trunk uses before choosing.

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.

Caller ID and attestation on SIP trunks

When a call is sent into a network that checks caller ID, the originating provider can sign it with a SHAKEN PASSporT. RFC 8588 defines three attestation levels the signature can carry:

• A, full: the provider "can fully attest to the calling identity", meaning it knows the customer and that the customer may use the number.
• B, partial: the provider originated the call but "cannot fully attest to the calling identity".
• C, gateway: the lowest level; the provider received the call from a gateway that does not support the signing.

This is worth asking about before you sign. If you present numbers that were not issued by your SIP trunk provider, such as numbers ported to another carrier or numbers belonging to your clients, calls may go out with B or C attestation. Ask the provider which level your calls will carry with the exact numbers you plan to present, and get the answer in writing.

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) between the SIP trunk and Teams. Microsoft's Direct Routing planning guide states: "Only Microsoft-certified SBCs are supported with Direct Routing." Check your SBC model against the certified list before ordering a trunk for Teams. 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?
• Which attestation level will your calls carry with the numbers you plan to present?
• Which codecs are the default, and do fax and DTMF tones pass reliably on them?

Test before you move traffic. Place calls to the destinations you care about, at your busy hour, and listen for one-way audio, clipped audio and slow call set-up. A short trial tells you more than any brochure, including this one.

MCC offers wholesale voice termination and cloud business phone services built on SIP. Contact us to discuss your requirements.

SIP trunking FAQ

Sources

• IETF — RFC 3261, SIP: Session Initiation Protocol
• IETF — RFC 8588, PASSporT extension for SHAKEN (attestation levels)
• Cisco — Bandwidth consumption calculation for voice calls (document 7934)
• Microsoft Learn — Plan Direct Routing
• Microsoft Learn — Session Border Controllers certified for Direct Routing