Hire a VoIP Developer
Hire expert VoIP developers with deep hands-on experience in Asterisk, FreeSWITCH, Kamailio, OpenSIPS, RTPEngine, SRTP encryption, WebRTC, and PBX platforms including FreePBX, FusionPBX, VitalPBX, Vicidial, 3CX, and Amazon Connect.
What Your VoIP Developer
Brings to the Table
Every VoIP Developer we place has been vetted for real-world production experience — not just resume keywords.
VoIP Developer Roles
You Can Hire
"VoIP Developer" covers several different jobs. Tell us which problem you are solving and we will put the right one on it.
Asterisk / FreeSWITCH Developer
Writes the call logic: dialplans, ARI and ESL applications, IVR flows, queue behaviour, recording. The engineer you need when the phone system has to do something no GUI exposes.
SIP Infrastructure Engineer
Kamailio and OpenSIPS territory - registrars, load balancers, least-cost routing, RTPEngine media relays and SRTP encryption between phones and carriers. Hired when volume outgrows a single PBX, or when two sides of a call cannot agree on encryption.
Contact Centre Engineer
Queues, ACD and skills-based routing, predictive and progressive diallers, Vicidial, and the CRM screen-pop and click-to-dial that decide whether agents are actually faster.
Telephony DevOps Engineer
Deployment, monitoring and SIP security. Firewalls and fail2ban against toll fraud, Linux tuned for concurrent channels, and Homer and sngrep captures ready before something breaks.
What Clients Hire
a VoIP Developer For
Projects we have actually delivered, not a capability list. If yours is not here, it is still worth asking.
Replacing a legacy PBX
Moving off ageing or unsupported hardware without losing the dialplan behaviour people depend on. The migration is usually harder than the build, and it runs in parallel.
Multi-tenant hosted PBX
One platform serving many customers with isolated tenants, per-tenant billing and provisioning, and a control panel their admins can use without calling you.
Outbound dialler and campaign platform
Predictive, progressive or power dialling with answering-machine detection, pacing that respects abandonment limits, and the compliance controls the target market requires.
CRM and telephony integration
Screen-pop on inbound caller ID, click-to-dial from records, automatic call logging and post-call disposition - across Salesforce, HubSpot, Zoho, GoHighLevel or a custom API.
SIP trunk migration and least-cost routing
Moving carriers without dropped calls, then routing by cost and quality per destination, with failover that actually fails over when a carrier degrades.
Rescuing a system someone else left
Undocumented, half-finished or actively broken telephony inherited from a previous team or vendor. We take these on deliberately; most firms will not.
Bridging AI voice agents to an existing PBX
Putting an AI agent on real DIDs through SIP trunking or CPaaS, so it answers and transfers alongside your current dialplan rather than replacing it.
How We Work Together
Dedicated Developer
One or more developers work exclusively on your project. Full-time, your timezone, your tools, your standups.
Best for: Ongoing product developmentSprint-Based
Structured 2-week sprints with defined deliverables, daily updates and fortnightly demos. Perfect for defined projects.
Best for: Fixed-scope projectsConsulting & Audit
Expert review of your existing system — architecture, code quality, performance or VoIP configuration — with a remediation plan.
Best for: System reviews & troubleshootingYour Developer Ships Faster
with AI Coding Tools
Every developer we place is trained and actively uses the latest AI coding assistants — delivering 30–50% faster output without sacrificing quality.
Common Questions
About Hiring
Still have questions? Chat with our team →
Our VoIP developers cover the full spectrum: open-source (Asterisk ARI/AMI/AGI, FreeSWITCH ESL, FusionPBX, Kamailio, OpenSIPS, RTPEngine, WebRTC) and commercial/cloud (PortaOne PortaSwitch, PortaBilling and PortaSIP, FreePBX, VitalPBX, Vicidial, Issabel PBX, 3CX, Amazon Connect, RingCentral, Five9). Each developer has a primary speciality and cross-stack experience.
Absolutely — troubleshooting and remediation is one of our strengths. Our developers diagnose one-way audio, registration failures, call quality issues, SIP fraud, and performance bottlenecks using sngrep, Wireshark, and platform-specific logging tools. Most critical issues are resolved within the same business day.
Yes. Our VoIP developers are full-stack telephony engineers — they write application code (ARI, ESL, AMI in Python/Node.js), configure SIP trunks and dialplans, tune Linux server performance, set up firewalls and fail2ban, and manage cloud deployments on AWS/GCP/DigitalOcean.
Yes. This sits with our OpenSIPS and RTPEngine engineers. In a recent project a telecom operator's PortaSIP phones offered AES-256 SRTP under a crypto suite name RTPEngine did not recognise, so every encrypted call was rejected. We patched RTPEngine 14.2 in C to accept it and wrote OpenSIPS 3.4 routing that keeps the phone side encrypted and the carrier side on plain RTP. All seven acceptance test calls passed, four outbound and three inbound, and the gateway is now live in production. The same engineers handle SDES key exchange in SDP and DTLS-SRTP for WebRTC.
Yes. CTI integration is a core skill — screen-pop from inbound caller ID, click-to-dial from CRM records, automatic call logging, and post-call disposition updates. We have done this with Salesforce, HubSpot, Zoho, GoHighLevel, and custom REST APIs across Asterisk, FreePBX, VitalPBX, 3CX, and Amazon Connect.
Written by the Engineers
You'll Be Hiring
Multi-Tenant Telephony: What Tenant Isolation Actually Requires
A SIP domain per customer is where isolation starts, not ends. The switch, the dialplan, carrier calls, the database and webhooks each need a boundary.
Read Article →Kamailio vs OpenSIPS in 2026: Choosing the Right SIP Proxy for High-Volume Telephony
50+ enterprise SIP deployments: Kamailio vs OpenSIPS on performance, scripting, module ecosystem, load balancing and WebRTC for 2026.
Read Article →Asterisk vs FreeSWITCH in 2026: Which VoIP Platform Should You Choose?
After 250+ VoIP projects: FreeSWITCH scales higher, Asterisk deploys faster. WebRTC, IVR, AI integration & real hosting costs benchmarked. Our 2026 verdict.
Read Article →Ready to Hire a VoIP Developer?
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