BYO SIP Trunk: Keep Your Existing Phone Numbers

Published: January 19, 2026 Updated: January 19, 2026 Reading time: 12 minutes

Add AI to your phones without changing your carrier.

That sentence might sound too good to be true if you have spent any time evaluating voice AI platforms. Most vendors want you to port your numbers to their infrastructure. They want you to abandon your existing carrier relationships, migrate your phone system, and rebuild from scratch.

For IT and telecom managers, this is a non-starter. You have phone numbers your customers know. You have carrier contracts with negotiated rates. You have compliance requirements tied to specific providers. You have routing configurations that took months to optimize. The last thing you need is a voice AI project that forces you to tear all of that down.

BYO SIP Trunk changes the equation. You keep your existing phone numbers, your existing carrier, and your existing infrastructure. You add Burki's AI layer on top. Calls flow through your SIP trunk, get enhanced with AI capabilities, and your callers never know anything changed except that a remarkably intelligent assistant now answers.

This guide explains how BYO SIP works, why it matters for enterprise deployments, and how to connect your existing telephony infrastructure to AI-powered voice assistants.

The Number Porting Problem

If you have evaluated voice AI platforms before, you have likely encountered the porting conversation. The vendor explains that you need to port your numbers to Twilio, Telnyx, or their proprietary carrier. They make it sound simple. It is not.

Porting Takes Weeks

Number porting in the United States typically takes 7-14 business days. International numbers can take 30-60 days or longer depending on the regulatory environment. For businesses with hundreds of phone numbers across multiple regions, a porting project can stretch into months.

During this time, you are in limbo. The numbers are transitioning between carriers. Your ability to make changes is limited. Any issues require coordination between old carrier, new carrier, and the voice AI vendor.

Risk of Downtime

Porting failures happen. A rejected port request means your number stays with the old carrier while your voice AI system expects it on the new one. Calls drop. Customers cannot reach you. The business impact is immediate and painful.

Even successful ports carry risk. The cutover moment when the number switches carriers can result in minutes or hours of downtime. For call centers handling thousands of calls daily, those minutes translate directly to lost revenue and damaged customer relationships.

Contractual Complications

Enterprise telecom contracts do not disappear because you want to add AI. You may have:

Multi-year agreements with volume commitments
Bundled services where voice is part of a larger package
Negotiated rates that took months to finalize
Termination fees that make early exit prohibitively expensive

Porting numbers often means breaking these contracts, paying penalties, and losing the favorable rates you worked hard to secure.

Compliance Complexity

Certain industries require specific carrier relationships. Healthcare organizations may need carriers with Business Associate Agreements. Financial services may have regulatory requirements about call recording jurisdiction. Government contractors may have FedRAMP or other certification requirements.

Switching carriers can jeopardize compliance status. The carrier you are forced to port to may not have the certifications you need. Suddenly your voice AI project has become a compliance crisis.

The BYO SIP Solution

BYO SIP (Bring Your Own SIP Trunk) eliminates these problems by working with your existing telephony infrastructure instead of replacing it.

What Is a SIP Trunk?

SIP (Session Initiation Protocol) is the standard protocol for voice over IP communications. A SIP trunk is your connection between your phone system and the public telephone network. Think of it as the digital equivalent of the copper lines that used to connect PBX systems to the phone company.

Your current carrier provides a SIP trunk that:

Delivers inbound calls to your phone system
Routes outbound calls to the PSTN
Handles caller ID, number formatting, and signaling
Manages the relationship with the physical telephone network

How BYO SIP Works with Burki

Instead of replacing your SIP trunk, Burki integrates with it. The architecture looks like this:

Inbound Call Flow:
Caller → PSTN → Your Carrier → Your SIP Trunk → Burki → AI Response

Outbound Call Flow:
Burki (AI) → Your SIP Trunk → Your Carrier → PSTN → Destination

Your carrier relationship stays intact. Your phone numbers stay where they are. Burki handles the AI processing while your existing infrastructure handles the telephony.

Keep Your Current Numbers

Every phone number you have today continues working. The numbers do not move. They do not port. Your carrier continues to own the relationship with those numbers. You simply configure your SIP trunk to route calls through Burki for AI enhancement.

Customers keep dialing the same numbers they always have. Internal systems keep referencing the same numbers. Marketing materials stay current. Nothing changes except the intelligence behind the calls.

Keep Your Current Carrier

Your relationship with your telephony provider remains unchanged. You continue paying them for minutes. You continue using their infrastructure. You continue benefiting from whatever rates you have negotiated.

This is particularly valuable for enterprises with:

Volume discount agreements
Bundled service packages
Specialized carrier features
Established support relationships

The carrier sees normal SIP traffic. They do not know or care that AI is involved. From their perspective, you are simply routing calls through an additional endpoint.

Add Burki's AI Layer on Top

Burki becomes your AI processing layer. When a call arrives via your SIP trunk:

Audio streams to Burki in real-time
Speech-to-text converts the caller's voice to text
The LLM generates an intelligent response
Text-to-speech converts the response to audio
Audio streams back through your SIP trunk to the caller

The entire round trip happens in under 1.2 seconds. Callers experience a natural conversation. Behind the scenes, your existing infrastructure handles all the telephony while Burki handles all the AI.

Technical Overview: How to Connect Your SIP Trunk

Setting up BYO SIP with Burki involves configuring your SIP trunk to route calls to Burki's SIP endpoint. The process varies slightly by carrier, but the core concepts are consistent.

SIP Registration and Authentication

Burki supports two authentication methods for SIP trunks:

Credentials-Based Authentication: You configure a SIP username and password. Your trunk authenticates with Burki using these credentials. This is the most common approach and works with any carrier that supports standard SIP registration.

IP-Based Authentication: You whitelist specific IP addresses that are authorized to send SIP traffic to Burki. Calls from those IPs are accepted without additional credentials. This is common for carriers that use static IP addresses for their SIP infrastructure.

Basic Configuration Steps

Step 1: Gather Your SIP Trunk Details

From your current carrier, you need:

SIP server address (e.g., sip.yourcarrier.com)
Authentication credentials or IP addresses
Supported codecs (typically G.711 u-law for US, a-law for international)
Any port or firewall requirements

Step 2: Configure Burki's SIP Settings

In your Burki dashboard:

Navigate to Settings > Provider Settings > SIP Trunk
Select "BYO SIP Trunk" as your provider
Enter your SIP domain or server address
Configure authentication (credentials or IP whitelist)
Enable inbound, outbound, or both directions
Save your configuration

Step 3: Configure Your Carrier's Routing

On your carrier's side:

Create a SIP endpoint pointing to Burki's SIP address
Configure routing rules to send specific numbers or all traffic to this endpoint
Set up failover to your existing system if needed

Step 4: Test the Connection

Make a test call to a number configured for Burki
Verify the AI assistant answers
Check call quality and latency
Monitor for any authentication or routing errors

Multiple Gateway Configuration

For redundancy and load balancing, Burki supports multiple SIP gateways. You can configure:

Primary and backup SIP endpoints
Geographic distribution for latency optimization
Separate gateways for different number ranges
Failover rules if one gateway becomes unavailable

Benefits of BYO SIP Trunk

Beyond avoiding the porting headache, BYO SIP offers tangible operational and financial benefits.

No Number Changes

Your marketing materials, business cards, websites, and customer records all reference your current phone numbers. Changing numbers means updating everything, notifying customers, and dealing with months of misdirected calls to old numbers.

With BYO SIP, nothing changes. The same numbers that worked yesterday work tomorrow. The only difference is what happens when someone calls.

Keep Negotiated Rates

Enterprise telecom contracts often include significant volume discounts. A company doing 500,000 minutes per month might have negotiated rates 30-40% below list price. These rates are tied to the carrier relationship.

BYO SIP preserves these rates. Your minutes continue flowing through your existing carrier. Your negotiated pricing applies. You do not sacrifice years of procurement work to implement AI.

Compliance with Local Regulations

Different countries have different requirements for telephone service providers. In some markets, only licensed carriers can handle voice traffic. In others, specific data residency requirements apply.

Your existing carrier has already navigated these requirements. They have the licenses, the certifications, and the compliance infrastructure. BYO SIP lets you leverage that investment while adding AI capabilities.

Faster Implementation

Number porting takes weeks. BYO SIP takes days. The difference is significant for projects with tight timelines.

With BYO SIP, you are configuring routing rules, not initiating carrier transitions. There is no waiting for porting windows. No coordinating between multiple parties. No risk of porting failures delaying your launch.

Reduced Operational Risk

Every infrastructure change introduces risk. The fewer changes you make, the lower the risk.

BYO SIP minimizes changes to your existing telephony stack. Your carrier infrastructure continues operating exactly as it always has. You are adding an AI processing layer, not rebuilding your phone system.

Compatible Providers

Burki's BYO SIP Trunk feature works with any SIP-compliant carrier. If your provider supports standard SIP trunking, it will work with Burki.

Tier-1 Carriers

The major carriers all offer SIP trunking services compatible with Burki:

AT&T (AT&T IP Flexible Reach)
Verizon (Verizon VoIP)
Lumen/CenturyLink (Lumen SIP Trunking)
Comcast (Comcast Business VoiceEdge)

Cloud Communications Providers

If you already use one of these providers, you can continue using them via BYO SIP:

Twilio (Elastic SIP Trunking)
Telnyx (SIP Trunking)
Vonage (Vonage Business Communications)
Bandwidth (SIP Trunking)
Plivo (SIP Trunking)
SignalWire (SIP Trunking)

Regional and Specialty Carriers

Local and specialty carriers work equally well:

Regional telcos (Frontier, Windstream, etc.)
International carriers (BT, Orange, Deutsche Telekom, etc.)
Specialty carriers (healthcare-focused, government-certified, etc.)
VoIP-only providers (Flowroute, VoIP.ms, etc.)

Any Standard SIP Trunk

The list above is illustrative, not exhaustive. Any carrier that provides:

Standard SIP signaling (RFC 3261)
RTP for audio transport
G.711 or G.729 codecs
Either credentials or IP-based authentication

...will work with Burki's BYO SIP feature. If you are uncertain about your specific carrier, contact Burki support with your carrier's technical specifications.

Setup Guide: Connecting Your SIP Trunk

Here is a step-by-step walkthrough for connecting your existing SIP trunk to Burki.

Prerequisites

Before starting, ensure you have:

Admin access to your Burki organization
SIP trunk credentials from your carrier
Network access to configure firewall rules if needed
A test phone number to validate the integration

Step 1: Access SIP Trunk Settings

Log into your Burki dashboard at app.burki.dev
Navigate to Settings in the left sidebar
Select Provider Settings
Click on SIP Trunk Configuration

Step 2: Configure SIP Connection

For Credentials-Based Authentication:

Select "Credentials" as authentication type
Enter your SIP domain (e.g., sip.yourcarrier.com)
Enter your SIP username
Enter your SIP password
Specify the SIP port (default: 5060 for UDP, 5061 for TLS)

For IP-Based Authentication:

Select "IP Whitelist" as authentication type
Enter your carrier's SIP server IP addresses
Add Burki's IP addresses to your carrier's whitelist (provided in dashboard)
Specify the SIP port

Step 3: Configure Routing

Enable "Inbound" if you want to receive calls via this trunk
Enable "Outbound" if you want to make calls via this trunk
Configure codec preferences (G.711 u-law recommended for US)
Set up any custom SIP headers your carrier requires

Step 4: Assign Phone Numbers

Navigate to Phone Numbers in your dashboard
Add your existing phone numbers (the ones on your SIP trunk)
Assign each number to an AI assistant or assistant graph
Configure any number-specific routing rules

Step 5: Configure Your Carrier

On your carrier's side, route calls to Burki's SIP endpoint:

SIP URI: sip:[email protected]
Port: 5060 (UDP) or 5061 (TLS)

Exact configuration varies by carrier. Contact your carrier's support for help configuring outbound SIP routing or call forwarding.

Step 6: Test End-to-End

Call one of your configured numbers
Verify the AI assistant answers
Have a test conversation
Check the call appears in your Burki dashboard
Verify audio quality is acceptable

Troubleshooting Tips

Calls not connecting:

Verify SIP credentials are correct
Check firewall rules allow SIP (UDP 5060) and RTP (UDP 10000-20000)
Confirm your carrier is routing to Burki's SIP endpoint

Audio issues:

Ensure codec match between carrier and Burki
Check for NAT traversal issues
Verify RTP ports are open bidirectionally

Authentication failures:

Double-check username and password
Verify SIP domain matches carrier expectations
Check IP whitelist includes all relevant addresses

Frequently Asked Questions

Can I use BYO SIP with any phone number?

Yes, any phone number connected to your SIP trunk works with Burki. This includes local numbers, toll-free numbers, international numbers, and vanity numbers. The numbers stay with your carrier and continue working exactly as they do today.

Does BYO SIP cost extra?

No. BYO SIP is included in Burki's standard platform pricing. You pay Burki's platform fee ($0.03/minute) plus your existing carrier costs. There is no additional charge for using your own SIP trunk versus a managed telephony provider.

What happens to my existing call routing?

Your existing call routing stays in place for numbers you do not assign to Burki. You can configure specific numbers to route to Burki while others continue to your existing phone system. This allows gradual migration without disrupting current operations.

Can I use BYO SIP for outbound calls?

Yes. Burki supports both inbound and outbound calling via BYO SIP. For outbound campaigns, calls originate from Burki, route through your SIP trunk, and use your carrier's outbound termination. Caller ID can be any number on your trunk.

Is there any latency added by BYO SIP?

BYO SIP adds minimal latency compared to managed telephony. The SIP signaling adds a few milliseconds. Audio processing is the same regardless of telephony source. Total response time remains under 1.2 seconds in typical deployments.

What codecs does Burki support?

Burki supports G.711 u-law (common in North America), G.711 a-law (common internationally), and G.729 (for bandwidth-constrained links). We recommend G.711 for best audio quality.

Do I need to change anything on my PBX?

That depends on your setup. If your PBX connects directly to the carrier, you may need to add routing rules to forward specific DIDs to Burki. If your carrier handles routing upstream, you may only need carrier-side configuration. Burki support can help assess your specific architecture.

Can I fail over to my existing system if Burki is unavailable?

Yes. Configure your SIP trunk with failover rules. If Burki's endpoint does not respond, calls route to your backup destination (existing IVR, call center, or voicemail). This provides business continuity during any service interruption.

Is BYO SIP secure?

Yes. Burki supports TLS for SIP signaling encryption and SRTP for media encryption. Credentials are encrypted at rest. All connections use modern cryptographic standards.

Why IT Managers Choose BYO SIP

The IT and telecom managers we talk to share common priorities: minimize risk, control costs, and avoid vendor lock-in. BYO SIP addresses all three.

Minimize risk by keeping your existing infrastructure intact. No porting. No carrier changes. No disruption to current operations. You are adding AI capabilities, not rebuilding your phone system.

Control costs by preserving negotiated carrier rates. Your volume discounts apply. Your bundled services stay bundled. You pay Burki only for AI processing, not for telephony markup.

Avoid vendor lock-in by maintaining carrier independence. If you ever decide to change voice AI providers, your phone numbers stay where they are. You are not trapped by ported numbers.

This is not about Burki being better than other voice AI platforms. It is about Burki working with your infrastructure instead of against it. The best technology deployment is one that enhances what you have rather than forcing you to replace it.

Getting Started with BYO SIP

Ready to add AI to your existing phone numbers?

Step 1: Sign up for Burki if you do not have an account. You get 200 free minutes to test.

Step 2: Gather your SIP trunk details from your current carrier.

Step 3: Configure BYO SIP in your Burki dashboard following the setup guide above.

Step 4: Assign your phone numbers to AI assistants.

Step 5: Make test calls and verify everything works as expected.

The entire process takes a few hours, not weeks. Your numbers stay where they are. Your carrier relationship continues unchanged. And your callers experience intelligent, AI-powered conversations.

Conclusion

BYO SIP Trunk is the bridge between your existing telephony investment and modern voice AI capabilities. It eliminates the false choice between keeping your infrastructure and adding AI.

No porting delays. No carrier changes. No compliance risks. No lost negotiated rates.

Just your existing phone numbers, your existing carrier, and Burki's AI answering calls in under 1.2 seconds.

Your infrastructure. Your numbers. Your AI.

That is how enterprise voice AI should work.

Ready to connect your SIP trunk to AI-powered voice assistants? [Start your free trial](https://burki.dev/signup) and configure BYO SIP in minutes. Keep everything you have. Add everything you need.

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