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Custom VoIP Development Service

Get more from your communications with a custom VoIP development company built around your needs. We deliver secure, scalable, carrier-grade telephony. Shaped around your call patterns, integrations, and real growth.

From discovery and architecture to engineering, deployment, and support. Our VoIP software development services give you full platform ownership. A telephony system that performs, adapts, and earns its place.

2500+

Projects Delivered

600+

Global Clients

250+

Professionals

18+

Years Exp.

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    Globally Trusted Clienteles

    Committed to delivering best-in-class services with proven project management
    practices incorporating more than a decade of experience

    Custom VoIP Development: Build a Platform That Scales With Your Business

    As a custom VoIP development company with 18+ years of experience, we serve telecom carriers, ISPs, and enterprises in 50+ countries. We build production-grade platforms on FreeSWITCH, Kamailio, OpenSIPS, Asterisk, and WebRTC, covering Class 4 and Class 5 softswitches, multi-tenant IP PBX, SBCs, and contact center systems.

    Our edge sits in open-source VoIP development. We pick the right engine for your call flow, then customize it at the module level for scale, security, and uptime. Our VoIP software development services cover fresh builds, legacy migrations, and carrier-grade extensions. We handle architecture, development, deployment, and L2/L3 support in one place.

    Asterisk Development

    Asterisk is an open-source communications toolkit sponsored by Sangoma and originally built by Mark Spencer in 1999. It is not a finished PBX out of the box. It is a framework that engineers shape into PBX systems, contact centers, IVR platforms, voicemail systems, and custom call-handling applications. It supports SIP, PJSIP, IAX, and standard PSTN interfaces, which makes it a strong fit for hybrid deployments.

    • We build IP PBX systems, contact centers, IVR, and voicemail platforms on Asterisk.
    • We migrate older chan_sip deployments to PJSIP and add ARI-based call control.
    • We integrate Asterisk with CRM, ticketing, and analytics through AMI, ARI, and AGI.
    Build Your Asterisk Solution
    Hire Asterisk Developers
    Hire FreeSWITCH Developers

    FreeSWITCH Development

    FreeSWITCH is an open-source telephony engine that runs on commodity hardware. It handles voice, video, and messaging across SIP, WebRTC, and other real-time protocols. Teams use it to build softphones, IP PBX systems, Class 5 softswitches, video conferencing platforms, and Session Border Controllers. We have shipped FreeSWITCH-based platforms for 18+ years.

    • We build carrier-grade Class 5 softswitches, multi-tenant IP PBX, and SBCs on FreeSWITCH.
    • We design audio and video conferencing platforms, IVR flows, and WebRTC gateways on a single core.
    • We handle module-level customization, codec tuning, and high-concurrency deployments end to end.
    Start Your FreeSWITCH Project

    WebRTC Development

    WebRTC is the open standard that lets browsers and mobile apps run real-time voice, video, and data sessions without any plugin. It is a W3C Recommendation, supported natively in Chrome, Firefox, Safari, and Edge. The protocol handles media capture, peer connection, encryption, and NAT traversal through three core JavaScript APIs. End-to-end latency typically lands well under half a second, which is why telehealth, fintech, and contact center vendors choose it.

    • We build browser-based softphones, click-to-call widgets, and WebRTC-to-SIP gateways.
    • We design video conferencing, telehealth, and remote support apps with sub-second latency.
    • We handle STUN/TURN, signaling servers, codec selection, and media server integration.
    Launch a WebRTC Solution
    Hire WebRTC Developers
    Hire OpenSIPs Developer

    OpenSIPS Development

    OpenSIPS is an open-source SIP server that combines signaling routing with application-level logic in one process. It works as a proxy, registrar, B2BUA, presence server, load balancer, or SBC depending on how you configure it. Carriers and ITSPs use it for Class 4 wholesale switches, Class 5 residential platforms, SIP trunking, and IMS deployments. Throughput reaches tens of thousands of calls per second on production hardware.

    • We build Class 4 wholesale switches, trunking platforms, and SBCs on OpenSIPS.
    • We implement custom routing scripts, dialplans, and dynamic least-cost routing for carrier traffic.
    • We integrate OpenSIPS with billing, OSS/BSS, and fraud detection systems through native APIs.
    Talk to OpenSIPS Experts

    Kamailio Development

    Kamailio is a high-performance SIP server written in C for Unix and Linux systems. It routes signaling traffic without touching the media path, which is why it scales further than full PBX stacks. A single instance can handle thousands of call setups per second, and stateless load-balancer deployments push that figure past five thousand. Tier 1 carriers, CPaaS platforms, and ITSPs use it to front their media servers.

    • We deploy Kamailio as a SIP proxy, registrar, and load balancer across FreeSWITCH or Asterisk clusters.
    • We configure dispatcher-based routing, NAT traversal, TLS, and anti-fraud modules for production traffic.
    • We architect geo-redundant Kamailio clusters for high-availability SIP signaling at scale.
    Build with Kamailio
    Hire Kamailio Developers

    Process We Follow

    Our VoIP software development services follow a proven process. We adapt each stage to the specific requirements of our clients.

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    01

    Project Kickoff

    We initiate the partnership by defining project goals, identifying stakeholders, and establishing clear communication channels.

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    02

    Discovery & Analysis

    Our team conducts in-depth research to understand your technical requirements, market landscape, and user needs.

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    03

    Architecture Design

    We craft a scalable technical blueprint, selecting the right technology stack and designing the system infrastructure.

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    04

    Development

    Our engineers build your solution using agile methodologies, ensuring high-quality code and regular progress updates.

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    05

    UAT & Deployment

    We conduct rigorous user acceptance testing and final refinements to ensure the solution meets all business objectives before launch.

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    TESTIMONIALS

    Look What
    Our Customers Say!

    Ecosmob has successfully helped service providers and enterprises all across the globe to scale their businesses and create unique software-based experiences. Here is what they have to say about it.

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    FAQs

    Timelines vary by scope: an MVP can take about 3–6 months, a core platform with integrations 6–9 months, and an enterprise-grade carrier solution 9–18 months. Parallel activities (requirements, testing, interop) and regulatory approvals can affect delivery. Book a scoping call to get a realistic timeline for your project.

    Carrier‑grade VoIP meets high standards for uptime, latency, scalability, and regulatory compliance—designs include HA, geo-redundancy, SBCs for security, and traffic-engineering for quality. It matters because enterprises and carriers require predictable reliability, clear SLAs, and fraud prevention. See our case studies for proven carrier deployments.

    Yes — we design, deploy, and integrate SBCs for signalling control, security, NAT traversal, and interconnect with carriers and PSTN gateways. Our SBC work includes provisioning automation, interoperability testing, and placement for DDoS mitigation and call routing. Request an SBC architecture review.

    We apply layered security: TLS/SRTP for signalling/media encryption, SBC-based signalling control, access policies, rate-limiting, fraud detection engines, and DDoS safeguards. Compliance is handled per-region (GDPR, HIPAA, PCI where required) with secure provisioning and regular penetration tests. Ask for our security checklist.

    Yes — we connect VoIP platforms to billing, rating engines, CRM, provisioning systems, and OSS/BSS via APIs, webhooks, and adapters. Integrations include real-time CDR export, automated rating, account lifecycle management, and reconciliation workflows. Share your current stack for a compatibility review.

    We architect platforms for horizontal scaling—individual nodes can handle thousands of concurrent calls, and clusters scale to tens or hundreds of thousands depending on design, codecs, and media handling. We provide example throughput baselines and sizing guidance after a network and use-case assessment. Request a capacity planning session.

    Yes — we offer managed services, 24/7 support, proactive monitoring, patching, and tiered SLAs tailored to carrier or enterprise needs. Support models include staff augmentation, fully managed hosting, and hybrid operations with defined response and resolution times. Review our SLA matrix to pick the right support tier.

    Yes — we migrate legacy softswitch and PSTN infrastructure to cloud-native or hybrid environments, handling number portability, interop testing, and phased cutover strategies to minimize downtime. Migrations include data mapping, billing transition, and rollback plans. Start with our migration readiness assessment.

    Our free VoIP audit evaluates your architecture, call flows, SBC placement, security posture, interoperability gaps, and cost-saving opportunities—delivering a prioritized report with remediation steps and a proposed roadmap. The audit is non‑binding and tailored to carriers and enterprises. Request your audit today.

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