What Is SIP Trunking?
SIP trunking is a method of delivering telephone services over the internet using the Session Initiation Protocol (SIP). Instead of connecting your phone system to the Public Switched Telephone Network (PSTN) through physical copper or fiber lines, a SIP trunk uses your existing internet connection to make and receive calls.
Think of a SIP trunk as a virtual phone line. Where a traditional PRI circuit provides a fixed number of channels over a dedicated T1 line, a SIP trunk provides voice channels over your IP network. Each concurrent call uses one channel, and you can scale the number of channels up or down without any hardware changes.
Key Term: SIP stands for Session Initiation Protocol. It is the signaling protocol used to set up, manage, and tear down voice (and video) sessions over IP networks. SIP handles call setup; the actual voice audio travels via RTP (Real-time Transport Protocol).
How SIP Trunking Works
The process of making a call over a SIP trunk follows these steps:
- Call initiation: When a user dials a number, the PBX sends a SIP INVITE message to the SIP trunk provider (like IPComms) over your internet connection.
- Call routing: The SIP trunk provider receives the INVITE, authenticates it, and routes the call to the destination - either to another VoIP endpoint or through a gateway to the PSTN.
- Media exchange: Once the call is established, voice audio flows directly between your PBX and the provider's media servers using RTP packets.
- Call termination: When either party hangs up, a SIP BYE message ends the session and releases the channel.
For inbound calls, the process is reversed: the SIP trunk provider receives the call from the PSTN and forwards it to your PBX as a SIP INVITE, directed to the DID (Direct Inward Dial) number that was called.
Key Benefits of SIP Trunking
Cost Savings
Businesses typically save 40-60% compared to traditional PRI circuits. No dedicated physical lines, lower per-minute rates, and no long-distance charges for many routes.
Scalability
Add or remove channels instantly without waiting for telco provisioning. Scale from 2 to 200 concurrent calls with a configuration change, not a truck roll.
Business Continuity
Easily reroute calls during outages. If your office loses power, calls can failover to mobile phones, a backup site, or voicemail automatically.
Geographic Flexibility
Get local phone numbers in any area code without a physical presence. Consolidate multi-site trunking to a single provider regardless of location.
Unified Communications
SIP integrates voice with video, messaging, and presence. Build a unified communications platform on a single protocol.
HD Voice Quality
Support for wideband codecs like G.722 and Opus delivers significantly clearer audio than traditional PSTN calls.
SIP Trunking vs PRI
PRI (Primary Rate Interface) delivers 23 voice channels over a dedicated T1 line. Here is how it compares to SIP:
| Feature | SIP Trunk | PRI |
|---|---|---|
| Channels | Flexible (1 to hundreds) | Fixed (23 per T1) |
| Monthly cost | $15-25/channel | $300-500/PRI circuit |
| Scaling time | Minutes | Weeks (new circuit) |
| Physical line | Uses existing internet | Dedicated T1 required |
| Failover | Automatic IP failover | Requires backup circuit |
| HD Voice | Yes (wideband codecs) | No (G.711 only) |
Migration Path: Most businesses migrating from PRI to SIP trunking can keep their existing phone numbers through number porting. IPComms handles the porting process to minimize downtime during the transition.
SIP Trunking vs POTS Lines
POTS (Plain Old Telephone Service) analog lines are the traditional copper phone lines that have been in use for over a century. They are being phased out by carriers across the country, and for good reason:
- Cost: A single POTS line costs $40-80/month and supports only one call at a time. A SIP channel costs a fraction of that.
- Features: POTS provides basic calling. SIP supports caller ID, call forwarding, simultaneous ring, video, and more.
- Maintenance: POTS relies on aging copper infrastructure that carriers are actively decommissioning. SIP runs on modern IP networks.
- Capacity: Need more lines? POTS requires physical installation. SIP scales instantly.
POTS Phase-Out: Major carriers including AT&T and Verizon are actively discontinuing POTS service. If you still rely on analog lines, now is the time to plan your migration to SIP trunking.
Common Use Cases
Office Phone System
Connect your on-premise PBX (Asterisk, FreePBX, 3CX) to the PSTN for making and receiving calls.
Call Centers
High-volume inbound and outbound calling with dynamic channel scaling based on call volume.
Multi-Site Business
Centralize trunking for multiple offices on a single SIP provider, with local DIDs in each market.
Remote Workforce
Connect distributed employees through softphones that register to a cloud PBX via SIP trunks.
What You Need to Get Started
To use SIP trunking, you need:
- An IP-PBX or SIP-capable phone system: Asterisk, FreePBX, 3CX, or any PBX that supports SIP trunking. Alternatively, use IPComms Hosted PBX if you do not want to manage your own system.
- Adequate internet bandwidth: Each concurrent call requires approximately 85-100 kbps with G.711. A 10 Mbps connection can handle roughly 100 simultaneous calls.
- Quality of Service (QoS): Configure your network to prioritize voice traffic over data to prevent call quality issues during peak usage.
- A SIP trunk provider: An account with a provider like IPComms that connects your PBX to the PSTN and provides phone numbers (DIDs).
Choosing a SIP Trunk Provider
When evaluating SIP trunk providers, consider these factors:
- Reliability and uptime: Look for providers with redundant infrastructure and a track record of high availability
- Call quality: The provider should have direct carrier interconnects rather than relying on least-cost routing through multiple intermediaries
- Pricing transparency: Avoid providers with hidden fees, minimum commitments, or complex pricing tiers
- Technical support: When calls are not working, you need responsive support from people who understand SIP
- PBX compatibility: Ensure the provider has tested and documented compatibility with your phone system
- Number porting: Smooth porting of your existing phone numbers is essential for business continuity
Why IPComms: IPComms has provided SIP trunking since 2002, with direct carrier interconnects, no contracts, transparent per-minute pricing, and support from VoIP engineers who understand your PBX. Try it free.
Ready to Switch to SIP Trunking?
IPComms makes SIP trunking simple. No contracts, no channel commitments, and no hidden fees. Start with a free test trunk and see the difference.