SIP Trunking Explained

1. What is SIP trunking?

A SIP trunk is a virtual phone line that connects your PBX or UCaaS platform to the Public Switched Telephone Network (PSTN) over an IP network. Instead of physical copper lines or ISDN circuits, SIP trunks use the Session Initiation Protocol to set up, manage, and terminate calls over your internet connection.

One SIP trunk can carry many simultaneous calls — the number is limited by your internet bandwidth and the provider's concurrent call license, not by physical circuit capacity. A typical SIP trunk supports 10-100+ simultaneous calls on a single IP connection.

SIP trunks connect to your PBX (Asterisk, FreeSWITCH, 3CX, Cisco CUCM) or directly to an SBC. The SBC or PBX handles the translation between your internal extensions and the outside world.

2. How SIP trunks work technically

A SIP trunk call flow for an outbound call:

1. Extension dials an external number → PBX matches dialplan and routes to trunk
2. PBX sends INVITE to SIP trunk provider's server
3. Provider authenticates the call (checks credentials or IP allowlist)
4. Provider routes to PSTN gateway and returns 100 Trying → 180 Ringing → 200 OK
5. PBX sends ACK → RTP media flows directly between PBX and provider
6. Call terminates with BYE from either side

Authentication methods

Digest authentication (username/password) — provider sends 407 Proxy Auth challenge, PBX responds with credentials. Most common for dynamic IP deployments.

IP allowlisting — provider only accepts calls from your registered public IP. No credentials needed. Most common for static IP enterprise deployments.

Registration vs static trunks

Registration-based: Your PBX registers to the provider like a phone. Supports dynamic IPs. The REGISTER keeps the connection alive and lets the provider route inbound calls to you.

Static/IP-authenticated: No registration. Provider routes inbound calls to your IP directly. Requires static IP and IP allowlisting.

3. Configuring a SIP trunk

Asterisk PJSIP trunk configuration

Key settings to verify

• context — inbound calls from the trunk land in this dialplan context
• direct_media=no — essential when either side is behind NAT
• dtmf_mode=rfc4733 — RFC 2833/4733 is the standard for DTMF over SIP trunks
• external_media_address — your public IP for SDP rewriting

4. Common SIP trunk issues

5. Choosing a SIP trunk provider

Key factors when evaluating SIP trunk providers:

• Pricing model — per-minute vs unlimited per channel vs concurrent call bundles
• Codec support — G.711 is universal; G.729 reduces bandwidth; Opus for quality
• Geographic coverage — does the provider have local numbers in your regions?
• Call quality SLAs — uptime guarantees, MOS score commitments
• STIR/SHAKEN support — A-attestation for your numbers reduces spam labeling
• Failover options — redundant POPs, automatic failover to backup trunks
• Support quality — SIP-knowledgeable engineers available 24/7
• Regulatory compliance — E911, CALEA, local number porting support

6. SIP trunk failover configuration

Configure multiple trunks with failover so calls continue if the primary provider has an outage:

Also configure DNS SRV records for automatic failover — most SIP providers publish SRV records that clients can use to discover backup servers without manual configuration.

Frequently asked questions

What is a SIP trunk?

A SIP trunk is a virtual phone line that connects your PBX or UCaaS platform to the PSTN over IP using the Session Initiation Protocol. Unlike physical ISDN lines, one SIP trunk can carry dozens of simultaneous calls. SIP trunks authenticate via username/password (digest auth) or IP allowlisting and connect to your PBX or SBC.

How does SIP trunking work?

SIP trunking works by sending SIP INVITE messages from your PBX to the provider's SIP server when a call is made. The provider authenticates the request, routes to the PSTN, and returns a 200 OK when the call connects. RTP media then flows directly between your PBX and the provider. Inbound calls reverse the flow — provider sends INVITE to your registered PBX.

What causes SIP trunk registration failure?

SIP trunk registration failures are caused by wrong credentials (username, password, or realm mismatch), firewall blocking port 5060, SIP ALG on the router corrupting the REGISTER packet, wrong server address in trunk configuration, or the provider requiring IP allowlisting instead of credential authentication. Check the 401 WWW-Authenticate realm matches your auth configuration.