SIP (Session Initiation Protocol) is the signaling protocol that sets up, modifies, and tears down voice and video calls over IP networks.

SIP trunking uses SIP to connect a PBX or call control platform to a service provider instead of using legacy PRI or T1 trunks. It matters because it's cheaper, far more flexible, scales without needing physical lines, and it's where the entire industry is going. AT&T, Verizon, and regional carriers are sunsetting copper and TDM as fast as they can.

SIP trunking has become the standard way businesses connect their phone systems to the public switched telephone network for inbound and outbound calling.

SIP trunking works by establishing a signaling connection between your PBX (on-premise or cloud-hosted) and a SIP trunk provider using the Session Initiation Protocol. When someone makes a call, your PBX sends a SIP INVITE message to the provider with details about the calling party, called party, and supported codecs.

The provider routes the call to its destination and sends back SIP response codes (100 Trying, 180 Ringing, 200 OK) indicating call progress. Once the call is answered, RTP (Real-time Transport Protocol) carries the actual audio between endpoints. The SIP signaling handles call setup, modification (hold, transfer), and teardown (BYE message), while RTP handles the media.

A Session Border Controller typically sits at the edge of your network to handle NAT traversal, security, and protocol normalization between your equipment and the carrier.

SIP Trunking can be used with a static IP address, and IP-address-based peering in a Network-to-Network Interface (NNI), or User-Network Interface (UNI). With the latter, a UNI connecting an enterprise or subscriber site to a carrier, SIPConnect-style registration can be used, removing the need for a static IP address. This means that the enterprise Voice platform is connected to the Voice provider and it can work regardless of the IP address in use. SIP REGISTER is used to establish the IP address of the SIP trunk.

Traditional SIP trunking sells you a fixed number of channels – say, 50 concurrent calls – and you pay for those whether you use them or not. Elastic SIP trunking, popularized by Twilio, Bandwidth Programmable Voice, and Telnyx, charges you per minute or per concurrent session as you use them, with no fixed channel count.

It's better for variable workloads like contact centers, marketing campaigns, and seasonal businesses, and worse for predictable steady traffic where a flat rate works out cheaper. The signaling and media protocols are the same SIP and RTP either way; the difference is the commercial model and the API surface for programmatic control.

The things that actually differentiate SIP trunking providers are:

• DNS capability: Do they publish proper NAPTR and SRV records or hand you static IPs?

• Transport: Do they support TLS for signaling and SRTP for media, or are they UDP-only?

• Codecs: G.711 is universal, but do they support G.722 for HD voice, G.729 for low-bandwidth, AMR-WB for mobile quality?

• E.164 expectations: Do they want +1NPANXXxxxx, 1NPANXXxxxx, or NPANXXxxxx, and how do they handle international?

• Domain names in SIP headers: Some carriers expect a specific FQDN in From and To URIs; others want the IP address.

• T.38 fax: Do they support it reliably?

• DTMF: RFC 4733 only, or do they accept SIP INFO too?

• STIR/SHAKEN: Do they deliver the Identity header unmodified and sign outbound calls with correct attestation?

Most of these you can't tell from a marketing page; you have to test.

The most common SIP authentication is digest authentication.

Your phone sends a REGISTER. The server replies with a 401 Unauthorized that includes a WWW-Authenticate header with a nonce (random string). The phone takes the SIP address of record, domain name, secret password, nonce, and other values, computes an MD5 hash of them, and sends a second REGISTER with an Authorization header containing the hash. The server computes the same hash on its side and verifies it matches.

This proves the phone has the password without sending the password over the wire. For SIP trunking specifically, you may also see IP-based authentication (no credentials, just an IP ACL) or SIP Connect-style pilot users where one REGISTER represents thousands of underlying phone numbers.

The advantages of SIP trunking include lower cost (typically 30-50% less than PRI), instant scalability by configuration rather than ordering more T1 lines, geographic flexibility with numbers from any area code without physical presence, support for modern features like HD voice and video, easier disaster recovery and business continuity, simplified multi-site deployments, and elimination of physical line maintenance.

The benefits of SIP trunking also include better integration with unified communications platforms, support for remote workers, and access to advanced features like STIR/SHAKEN caller ID verification. However, SIP trunking requires network quality of service planning, an SBC at the network edge, and more sophisticated troubleshooting when problems occur.

VoIP (voice over IP)  is the broad category, and includes everything from iPhone FaceTime Audio to home Comcast service to enterprise calls over Microsoft Teams.

SIP is the specific signaling protocol most VoIP services use to set up and tear down calls. SIP trunking is the use of SIP to connect a phone system (PBX or hosted call control platform) to a service provider for PSTN access.

So all SIP trunking is VoIP, but not all VoIP is SIP trunking. Microsoft Teams native calling, for example, is VoIP but uses a proprietary REST-based call control protocol internally, not SIP – SIP only shows up at the boundary with carriers through Direct Routing or Operator Connect. Apple FaceTime is VoIP but doesn't use SIP at all.

PRI (Primary Rate Interface) is a TDM (time-division multiplexing) technology running on T1 or E1 circuits with fixed 23 or 30 voice channels per line, using ISDN signaling protocols. SIP trunking vs PRI: SIP uses IP packets over ethernet or internet connections with dynamic, elastic capacity limited only by bandwidth and licensing.

PRI has predictable, guaranteed voice quality because it's circuit-switched; SIP quality depends on network conditions and requires QoS configuration. PRI setup requires physical installation of copper or fiber lines; SIP can be provisioned in minutes over existing internet connections.

PRI uses Q.931 signaling with limited feature support; SIP supports rich features like presence, video, and programmable APIs. For businesses moving from PRI to SIP trunks, the main considerations are SBC deployment, carrier selection, redundancy planning, and testing edge cases like fax and DTMF that worked reliably on PRI but require configuration on SIP.

ICE (Interactive Connectivity Establishment) is the framework tying these technologies together. Generally, ICE is one approach, and session border controllers (SBCs) make another approach. With ICE, each endpoint gathers candidate transport addresses (host, server-reflexive from STUN, relayed from TURN), exchanges them in SDP, and runs STUN connectivity checks to find a working path.STUN (Session Traversal Utilities for NAT) helps clients behind NAT discover their public IP address by querying a STUN server. TURN (RFC 5766) is when STUN isn't enough – the NAT is too restrictive, and you must relay all media through a public relay server.

You need ICE primarily when integrating with WebRTC endpoints and Microsoft Teams Direct Routing – Teams uses ICE-Lite, meaning the SBC must respond to STUN connectivity checks but never initiate them, must offer exactly one IPv4 candidate, and must handle forking where calls ring on multiple Teams endpoints simultaneously.

The minimum for business SIP trunking: G.711 µ-law (payload type 0) for North America and Japan, G.711 A-law for the rest of the world – that's your interop baseline.

From there: G.722 (payload type 9) gives HD voice at the same network bandwidth as G.711, so if both ends support it, use it.

G.729 (payload type 18) is your low-bandwidth option at 41 kbps per direction but sounds like an old cell phone.

AMR-WB/G.722.2 is what modern cell phones use for HD voice over LTE. It's patent-encumbered, so less common in enterprise SIP trunks.

For video, H.264 is dominant, while Opus is increasingly common for WebRTC. And telephone-event (typically payload type 101) is RFC 4733 DTMF – compressed codecs cannot reliably carry DTMF tones in-band, so you must negotiate telephone-event for digits to work in IVR systems.

Costs go down, flexibility goes up, and you can scale by configuration rather than ordering more T1s. However, failure modes are more numerous and subtle. PRI either works or it doesn't; SIP can work for 99% of calls and fail in weird ways for the other 1% due to codec mismatches, NAT issues, or carrier-specific E.164 quirks.

You'll need an SBC at the edge – Ribbon, AudioCodes, Oracle/Acme, Sansay, or open-source options – and you'll need to think about TLS, SRTP, STIR/SHAKEN, fax, and DTMF in ways you didn't with PRI.

You'll also need redundancy – preferably two SIP trunking service providers, not just two trunks from the same carrier, because when Bandwidth or Verizon Business has an outage, both trunks go down together. The projects that go well are the ones where planning happened upfront.

SIP trunking is used for connecting business phone systems to the public switched telephone network (PSTN) for inbound and outbound calling without legacy PRI or analog lines.

Common use cases include:

• Replacing expensive PRI circuits with cost-effective IP connectivity

• Enabling remote and mobile workers to use business phone numbers

• Supporting contact centers with elastic capacity that scales during busy periods

• Providing geographic flexibility with phone numbers from any area code

• Integrating with unified communications platforms like Microsoft Teams through Direct Routing

• Enabling disaster recovery and business continuity with multiple carrier connections

• Supporting video calling alongside voice on the same infrastructure

• Facilitating international calling with better rates than traditional carriers

What is a SIP trunk used for specifically? It's the bridge between your internal phone system (PBX) and external carriers, handling call routing, number portability, emergency calling, and PSTN interconnection.