What is SIP Trunking?
SIP trunking is a method of sending and receiving phone calls over the internet instead of through traditional copper telephone lines. SIP stands for Session Initiation Protocol, and a trunk is the connection between your phone system and the outside telephone network. Put them together, and you get SIP trunking: an internet-based phone line that connects your business PBX (Private Branch Exchange) to the public switched telephone network (PSTN).
If your company has an on-premise PBX -- whether that is an Asterisk server, a FreePBX appliance, a 3CX system, or any other IP-PBX -- a SIP trunk replaces the physical phone lines (PRI circuits or analog POTS lines) that traditionally carried your calls. Instead of a bundle of copper wires running into your building, a SIP trunk uses your existing internet connection to carry voice traffic.
Think of it this way: a traditional PRI line is like a dedicated highway toll road between your office and the phone company. A SIP trunk is like sending your calls over the existing internet highway you are already paying for. The destination is the same -- your calls still reach any phone number in the world -- but the path is digital, flexible, and significantly less expensive.
In Simple Terms: SIP trunking lets your business phone system make and receive calls over the internet. You keep your PBX, you keep your phone numbers, but you replace expensive phone lines with a more affordable internet-based connection. With providers like IPComms, setup takes minutes, not weeks.
A Brief History
SIP was first defined in 1996 and standardized by the IETF as RFC 3261 in 2002. Early adoption was slow because internet connections were unreliable and bandwidth was limited. But as broadband became ubiquitous and network quality improved, SIP trunking evolved from an experimental technology into the dominant method of business telephony.
By 2026, the transition is nearly complete. Major carriers have begun decommissioning ISDN and PRI infrastructure. The question is no longer whether to adopt SIP trunking, but which provider to choose and how to configure it properly. Businesses that have not yet migrated are paying a premium for legacy technology that offers fewer features and less reliability than modern SIP services.
How Does SIP Trunking Work?
Understanding how SIP trunking works requires understanding three key components: SIP signaling, RTP media streams, and the role of the SIP trunk provider. Here is how they fit together.
The Call Flow
When someone at your company picks up the phone and dials a number, here is what happens behind the scenes:
SIP INVITE (Signaling)
Your PBX sends a SIP INVITE message to your SIP trunk provider (e.g., IPComms) over the internet. This message contains the destination number, caller ID, codec preferences, and other call setup information. SIP handles all the signaling -- it is the protocol that sets up, manages, and tears down calls.
PSTN Routing
The SIP trunk provider receives the INVITE and routes the call to the public telephone network (PSTN). If the destination is another VoIP user, the call may stay entirely on IP networks. If it is a landline or mobile phone, the provider's gateway converts the call from VoIP to traditional telephony signals.
RTP Media Stream
Once the call is answered, the actual voice data flows as RTP (Real-time Transport Protocol) packets directly between your PBX and the provider's media servers. RTP carries the encoded audio, while RTCP (RTP Control Protocol) monitors call quality metrics like jitter, latency, and packet loss.
Call Termination
When either party hangs up, a SIP BYE message is sent to tear down the session. Both the signaling and media streams are released, freeing bandwidth for other calls.
Audio Codecs
Codecs encode and decode voice audio into digital data. The codec you choose affects both call quality and bandwidth consumption. The two most common codecs in SIP trunking are:
| Codec | Bandwidth per Call | Audio Quality | Best For |
|---|---|---|---|
| G.711 (ulaw/alaw) | ~100 Kbps | Excellent (toll quality) | LAN/high-bandwidth connections |
| G.729 | ~32 Kbps | Good | Low-bandwidth/remote connections |
| Opus | 6-510 Kbps (variable) | Excellent (HD) | WebRTC / HD voice applications |
| G.722 | ~80 Kbps | Very Good (wideband) | HD voice desk phones |
The Role of Session Border Controllers (SBCs)
In enterprise deployments, a Session Border Controller (SBC) sits between your PBX and the SIP trunk provider. The SBC acts as a specialized firewall for voice traffic, providing:
- ✓ Security: Protects against SIP-based attacks, toll fraud, and unauthorized access
- ✓ NAT Traversal: Solves network address translation issues that cause one-way audio
- ✓ Protocol Normalization: Translates between different SIP implementations
- ✓ Encryption: Terminates TLS/SRTP connections for secure voice
- ✓ Quality of Service: Prioritizes voice packets over data traffic
For Smaller Deployments: Many small and mid-size businesses connect their PBX directly to the SIP trunk provider without a dedicated SBC. Modern PBX platforms like Asterisk and FreePBX have built-in SIP security features. IPComms supports both direct PBX registration and SBC-mediated connections.
SIP Trunking vs Traditional Phone Lines
The three main types of traditional business phone connections are PRI (Primary Rate Interface), POTS (Plain Old Telephone Service), and SIP trunking. Here is how they compare across every dimension that matters.
| Feature | PRI (T1/E1) | POTS / Analog | SIP Trunking |
|---|---|---|---|
| Channels | 23 per T1 circuit | 1 per line | Unlimited (bandwidth-dependent) |
| Monthly Cost | $300-500/month per PRI | $30-50/month per line | $0.009-0.02/min or $15-25/channel |
| Connection | Dedicated copper T1 | Copper pair | Internet (existing broadband) |
| Setup Time | 4-8 weeks | 1-3 weeks | Minutes to hours |
| Scalability | Add full PRI (23 channels) | Add one line at a time | Add channels instantly |
| Failover | Requires redundant PRI | None | Built-in (multi-site, cloud) |
| Geographic Flexibility | Tied to physical location | Tied to physical location | Numbers from any area code |
| Long Distance | Per-minute charges | Per-minute charges | Flat rate or low per-minute |
| Encryption | Not available | Not available | TLS/SRTP supported |
| Future Outlook | Being decommissioned | Being decommissioned | Industry standard |
Cost Comparison Example
Consider a business with 50 employees making an average of 2,000 minutes of outbound calls per month and needing 10 concurrent call channels:
PRI
$450/mo
1 PRI (23 channels) + long distance
POTS (10 lines)
$400/mo
10 lines x $40/ea
SIP Trunking
$20/mo
$0/channel + 2,000 min x $0.010
The PRI Sunset: Major carriers including AT&T, Verizon, and CenturyLink have announced plans to decommission legacy PSTN infrastructure. If your business still relies on PRI or POTS lines, now is the time to plan your migration to SIP trunking before those services are discontinued.
Benefits of SIP Trunking
SIP trunking has become the standard for business telephony because it delivers measurable advantages across cost, flexibility, reliability, and features. Here are the key benefits.
Cost Savings (40-60%)
The most immediate benefit is cost reduction. Businesses typically save 40-60% on their phone bills when switching from PRI to SIP trunking. With IPComms pricing, you pay $0 per channel and only $0.009/min for inbound calls and $0.010/min for outbound calls. No contracts, no minimum commitments, no hidden fees.
Instant Scalability
Need more call capacity? With PRI, you wait weeks for a new circuit and pay for 23 channels whether you use them or not. With SIP trunking, you add channels instantly through your provider's portal. Scale up for busy seasons and scale down when call volume drops. Your capacity matches your actual needs.
Business Continuity
SIP trunking enables failover capabilities impossible with traditional lines. If your primary office loses internet, calls can automatically route to a backup location, mobile phones, or a cloud-based PBX. Geo-redundant SIP providers maintain infrastructure across multiple data centers, ensuring your phones keep working even during outages.
Geographic Flexibility
With SIP trunking, you can have local phone numbers in any area code, regardless of your physical location. A company in Miami can have New York, Chicago, and Los Angeles numbers. International DID numbers let you establish a local presence in countries around the world without opening physical offices.
Additional Benefits
- ✓ Number Portability: Port your existing phone numbers to your SIP provider. Keep the numbers your customers know while gaining all the benefits of SIP trunking.
- ✓ Unified Communications: SIP trunking integrates with UC platforms, CRM systems, and collaboration tools. Connect your phone system with Microsoft Teams, Salesforce, or custom applications.
- ✓ Enhanced Call Routing: Time-based routing, skills-based routing, and IVR menus are all possible through your PBX. SIP trunking makes these features available without expensive carrier add-ons.
- ✓ Call Encryption: TLS encrypts SIP signaling and SRTP encrypts voice media, providing end-to-end call security. Essential for healthcare (HIPAA), financial services (PCI-DSS), and legal firms handling sensitive client communications.
- ✓ Detailed Analytics: SIP trunking provides comprehensive call detail records (CDRs) including call duration, destination, codec used, and quality metrics. Use this data to optimize staffing, track marketing campaigns, and identify calling patterns.
- ✓ STIR/SHAKEN Compliance: Modern SIP providers implement STIR/SHAKEN caller ID authentication, which helps your outbound calls display verified caller ID information. This reduces the chance of your calls being marked as spam.
What You Need for SIP Trunking
Getting started with SIP trunking requires a few key components. The good news: if you already have an internet connection and a PBX, you are most of the way there.
A Reliable Internet Connection
SIP trunking runs over your internet connection, so quality matters. While SIP calls work over any broadband connection, a dedicated internet connection or MPLS circuit provides the best quality for businesses with high call volumes.
Bandwidth Calculator: Each concurrent call using G.711 requires approximately 100 Kbps of bandwidth (both upload and download). For 10 simultaneous calls, you need at least 1 Mbps of dedicated bandwidth. For 50 concurrent calls, plan for at least 5 Mbps. Most modern business internet connections can handle SIP trunking with bandwidth to spare.
A SIP-Compatible PBX
Your PBX must support SIP trunking. Nearly all modern IP-PBX systems do, including:
Asterisk - Open source, highly customizable
FreePBX - GUI for Asterisk
3CX - Windows/Linux-based
FusionPBX - FreeSWITCH-based
Cisco UCM - Enterprise-grade
Avaya - Legacy + modern systems
Do not have a PBX? A hosted PBX service provides all the features of a PBX without on-site hardware. You can also connect SIP trunks directly to SIP desk phones or softphones for simple deployments.
A SIP Trunk Provider
Your SIP trunk provider is the bridge between your PBX and the telephone network. The provider supplies your phone numbers (DIDs), routes your calls to the PSTN, and handles number porting. IPComms has been providing enterprise SIP trunking since 2002, with geo-redundant infrastructure, $0 per channel pricing, and 24/7 support.
Proper Network Configuration
For optimal call quality, configure your network to prioritize voice traffic:
- ✓ Enable Quality of Service (QoS) on your router to prioritize SIP/RTP traffic
- ✓ Open firewall ports: UDP 5060 (SIP), UDP 10000-20000 (RTP), TCP 5061 (TLS)
- ✓ Consider a VLAN to separate voice and data traffic
- ✓ Ensure latency is below 150ms and jitter below 30ms for clear calls
SIP Trunking Pricing Models
SIP trunk pricing varies significantly between providers. Understanding the two main pricing models helps you choose the most cost-effective option for your call volume. For a deeper dive, see our SIP Trunking Pricing Guide.
Per-Minute (Pay As You Go)
You pay only for the minutes you use, with no charges for channels (concurrent call paths). This model is ideal for businesses with variable call volumes or lower overall usage.
IPComms pricing. No contracts or minimums.
Per-Channel (Flat Rate)
You pay a fixed monthly fee per concurrent call path (channel). Some per-channel plans include unlimited calling, while others combine a channel fee with reduced per-minute rates.
Typical industry pricing from various providers.
Which Model Saves You More?
The right pricing model depends on your calling patterns. Here is a quick comparison for different usage levels:
| Monthly Usage | Per-Minute ($0/ch + $0.010/min) | Per-Channel (10 ch x $20/ch) | Better Value |
|---|---|---|---|
| 1,000 minutes | $10.00 | $200.00 | Per-Minute |
| 5,000 minutes | $50.00 | $200.00 | Per-Minute |
| 10,000 minutes | $100.00 | $200.00 | Per-Minute |
| 50,000 minutes | $500.00 | $200.00 | Per-Channel |
IPComms Advantage: With $0 per channel pricing, IPComms eliminates the channel cost entirely. You only pay for minutes used plus your DID number fees. For most businesses, this results in significantly lower costs compared to per-channel providers. See our full pricing breakdown.
How to Choose a SIP Trunk Provider
Not all SIP trunk providers are created equal. The cheapest per-minute rate means nothing if calls drop, quality is poor, or support is unreachable. Here are the key criteria to evaluate when choosing a provider.
Network Reliability & Uptime SLA
Look for providers with a 99.99% uptime SLA backed by service credits. Ask about their infrastructure: How many data centers do they operate? Do they have geographic redundancy? What happens if one data center goes down? IPComms operates geo-redundant infrastructure across multiple carrier-neutral data centers.
Security & Encryption
Your provider should support TLS (Transport Layer Security) for encrypted SIP signaling and SRTP (Secure Real-time Transport Protocol) for encrypted voice media. If you operate in a regulated industry (healthcare, finance, legal), encryption is not optional -- it is a compliance requirement.
STIR/SHAKEN Compliance
STIR/SHAKEN is a caller ID authentication framework mandated by the FCC. Your provider must sign outbound calls with digital certificates to verify your caller ID is legitimate. Without STIR/SHAKEN, your outbound calls may be flagged as spam or blocked by receiving carriers.
PBX Compatibility
Ensure the provider officially supports your PBX platform and can provide configuration guides. IPComms supports all major PBX platforms including Asterisk, FreePBX, 3CX, FusionPBX, Cisco, Avaya, and more. Configuration guides and technical support are available for each platform.
Support Quality
When your phones stop working, you need real people who understand SIP, not a chatbot. Evaluate the provider's support channels (phone, email, ticket), response times, and whether their support team has actual VoIP engineering expertise. Ask about their escalation process and SLA for critical issues.
Transparent Pricing
Beware of providers that hide fees behind complex rate decks, charge "regulatory recovery" fees, or require long-term contracts. The best providers publish their rates clearly and do not lock you into multi-year commitments. IPComms pricing is fully transparent with no hidden fees, no contracts, and no minimum usage requirements.
Number Porting & E911
Verify that the provider supports number porting from your current carrier and offers E911 (Enhanced 911) service. E911 is a legal requirement for business phone systems and ensures emergency calls route to the correct dispatch center with your location information.
SIP Trunking FAQ
What is the difference between SIP trunking and VoIP?
VoIP (Voice over IP) is the broad technology of sending voice calls over the internet. SIP trunking is a specific type of VoIP service that connects your on-premise or hosted PBX to the public telephone network (PSTN) using the SIP protocol. Think of VoIP as the category and SIP trunking as the connection method. You can use VoIP without SIP trunking (e.g., Skype, WhatsApp calls), but SIP trunking always uses VoIP technology.
How many calls can a SIP trunk handle?
A SIP trunk can handle as many simultaneous calls as your internet bandwidth allows. Each concurrent call using the G.711 codec requires approximately 100 Kbps of bandwidth. With a 100 Mbps connection, you could theoretically support over 1,000 concurrent calls. With providers like IPComms that offer $0 per channel pricing, there is no artificial limit on the number of channels -- you only pay for the minutes you use.
Is SIP trunking reliable?
Yes, SIP trunking is highly reliable when deployed correctly. Enterprise-grade SIP trunk providers like IPComms offer 99.99% uptime SLAs, geo-redundant infrastructure across multiple data centers, and automatic failover. SIP trunking can actually be more reliable than traditional phone lines because it supports failover to backup destinations -- something that is not possible with PRI or POTS lines. If your primary office loses connectivity, calls can automatically route to a secondary location or mobile phones.
Can I keep my existing phone numbers?
Yes. Number porting allows you to transfer your existing phone numbers from your current carrier to your SIP trunk provider. The porting process typically takes 7-14 business days for standard numbers, though toll-free numbers may take slightly longer. During the transition, your numbers continue working on your old carrier until the port is complete, ensuring zero downtime.
Do I need a PBX for SIP trunking?
Not necessarily. While SIP trunking is traditionally used with an on-premise PBX (like Asterisk, FreePBX, or 3CX), you can also use a hosted PBX service or connect SIP trunks directly to SIP-compatible desk phones and softphones. Hosted PBX solutions eliminate the need for on-site hardware entirely, which is a popular option for small businesses and remote teams.
How long does it take to set up SIP trunking?
SIP trunk setup is nearly instant. With IPComms, you can create an account, configure your SIP trunk, and start making calls within minutes. The entire process -- from signup to your first call -- can be completed in under 30 minutes if your PBX is already configured. If you are porting existing numbers, the numbers will continue working on your old carrier until the port completes (typically 7-14 business days), and you can use temporary DID numbers in the meantime.
Ready to Switch to SIP Trunking?
IPComms offers $0 per channel SIP trunks with rates starting at $0.009/min. No contracts, no minimums, no hidden fees. Set up your SIP trunk in minutes.