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VoIP Development · 7+ Years Telecom

Telecom grade VoIP, contact centres and AI voice bots

FreeSWITCH, Asterisk, Kamailio, OpenSIPs and WebRTC built by an India based team that has been shipping production telecom since 2018. Carrier integrations, multi tenant PBX, intelligent IVR, session border controllers and LLM powered voice bots, delivered with the same AI native engineering we use everywhere else.

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RG INSYS LLP is a specialist VoIP development partner with seven plus years of production telecom delivery. We build on FreeSWITCH, Asterisk, Kamailio, OpenSIPs and WebRTC, integrate carrier SIP trunks, design session border controllers, deliver outbound dialers, IVR systems, multi tenant PBXs, billing engines and AI voice bots powered by GPT-4, Claude and open source LLMs. Telecom operators, contact centres, healthcare providers and SaaS platforms in the UK, US, UAE and India use us when call quality, scale and uptime are non negotiable.

What we deliver
FreeSWITCH/Asterisk dialplans and modules, Kamailio/OpenSIPs SBC and routing, WebRTC softphones, intelligent IVR, AI voice bots, call recording and transcription, real time CRM integration, billing and rating engines, NMS dashboards.
Typical timeline
4 to 6 weeks for a PoC, 12 to 20 weeks for a production multi tenant platform, ongoing operations on a monthly retainer.
Pricing from
$20,000 fixed price PoC. $6,000/month dedicated VoIP engineer with AI tooling. Custom enterprise engagements quoted from a written scope.
Stack
FreeSWITCH, Asterisk, Kamailio, OpenSIPs, WebRTC (JsSIP, SIP.js, Verto), Node.js, Lua, C, PostgreSQL, Redis, RTPproxy, Janus, Deepgram, Whisper, ElevenLabs, OpenAI, Anthropic.
Compliance-ready for
HIPAA voice handling, GDPR, PCI DSS for payment IVR, TRAI/DLT/DND for India outbound, MiFID II call recording where applicable.
What's included

Everything a real telecom platform needs

🛰️

FreeSWITCH and Asterisk core

Dialplan design in XML and Lua, custom C modules where performance matters, ESL based event handling in Node.js. Multi tenant configurations, conference bridges, music on hold, recording, queues and routing logic built to survive production load.

🛡️

Kamailio / OpenSIPs SBC

Session border controllers for SIP trunk termination, registration, NAT traversal, encryption (TLS/SRTP), DDoS protection and topology hiding. Load balancing in front of FreeSWITCH/Asterisk clusters with intelligent routing and failover.

🌐

WebRTC apps and softphones

Browser based calling using JsSIP, SIP.js and Verto. STUN/TURN setup with coturn, codec negotiation, ICE traversal, audio quality tuning, in app video and screen share. Embedded inside SaaS dashboards or delivered as standalone agent consoles.

🤖

AI voice bots and intelligent IVR

Speech to text (Deepgram, Whisper, Google), LLM intent and dialogue (GPT-4, Claude, Llama 3), text to speech (ElevenLabs, Azure, Polly). Real time barge in, sub second latency, CRM lookups mid call, escalation to human agents with full context handover.

📞

Contact centre and outbound dialer

Inbound queues with skills based routing, predictive and progressive outbound dialers, agent desktops, supervisor dashboards, call recording, transcription and sentiment scoring. CRM popups and click to dial across Salesforce, HubSpot, Zoho and custom backends.

💵

Billing, rating and CDR analytics

Real time call detail record processing, rating engines for prepaid and postpaid, fraud detection, partner reconciliation and operator settlement. Reporting dashboards built on PostgreSQL with timeseries extensions or ClickHouse for high volume CDR.

Our method

How a VoIP build actually unfolds

01
Discovery, week 1

Workshop on call flows, traffic projections, carrier setup, regulatory requirements and integration points. Output: a written architecture document, codec and protocol decisions and a fixed price PoC scope.

02
PoC, weeks 2 to 5

A working prototype on FreeSWITCH or Asterisk with one or two real call flows, a CRM or web integration, recording and a basic admin UI. Stress tested with SIPp and benchmarked for concurrent channels and CPS.

03
Production build, weeks 6 to N

Full Kamailio/OpenSIPs SBC, FreeSWITCH cluster, WebRTC clients, AI voice bots, billing, dashboards. Two week sprints, demo every Friday, AI agents handling boilerplate while senior engineers own routing logic and security.

04
Go live and operate

Carrier cutover, load testing in production, NOC runbooks, monitoring on Prometheus and Grafana with VoIP specific metrics (ASR, ACD, MOS). Optional 24/7 ongoing support on retainer.

Our tech stack for VoIP development

Telecom is a domain where you cannot bluff: the protocols are unforgiving and the production traffic exposes every shortcut. We work with the same open source telecom stack we have been running since 2018, paired with modern AI services for speech and language. Where you have a preferred vendor (Twilio, Telnyx, Vonage, Bandwidth) we integrate cleanly without asking you to leave it.

FreeSWITCH Asterisk Kamailio OpenSIPs WebRTC (JsSIP / SIP.js / Verto) RTPproxy / RTPengine coturn (STUN/TURN) Janus Gateway Node.js + ESL Lua C / C++ PostgreSQL Redis Deepgram / Whisper OpenAI / Anthropic ElevenLabs / Polly
Proof

A representative case study

Logistics · UAE UAE last mile logistics operator

Voice and chat dispatch automation for a UAE logistics fleet

A UAE last mile logistics company replaced a manual phone dispatch desk with a FreeSWITCH plus AI voice bot stack. Drivers and customers now interact with an Arabic and English voice bot for status, ETA and rebooking; only exceptions reach a human. Integrated with the existing fleet tracking system over Node.js webhooks and a Postgres event store, with WebRTC agent consoles for the dispatch team.

62%Calls handled by bot
14 wksFrom PoC to live
2 langsArabic + English
3 devsTotal team size

Read full case study →

Pricing

Transparent pricing for VoIP development

From $20,000

Fixed price 4 to 6 week PoC. Or move to a $6,000/month dedicated VoIP engineer retainer with full AI tooling and on call coverage.

  • Working FreeSWITCH or Asterisk prototype with real call flows
  • SIPp load tested and benchmarked for concurrent channels
  • Architecture document and production hardening plan
  • 30 day post launch stabilisation, P1 fixes inside 4 hours
Full pricing & engagement models →

All pricing transparent. No hidden fees. Free 48-hour written estimate.

FAQ

Common questions

It is the original speciality of the company. RG INSYS started in 2018 with telecom and VoIP work and has shipped FreeSWITCH and Asterisk based platforms in production every year since. Our engineers write custom modules in C, ESL scripts in Lua and Node.js, and Kamailio/OpenSIPs configurations for SBC and load balancing. We have handled outbound dialers, multi tenant PBXs, contact centres and call recording platforms in 24/7 production.
An IVR plays prompts and routes by DTMF or simple speech. An AI voice bot uses speech to text (Deepgram, Whisper, Google), an LLM (GPT-4, Claude, Llama 3) for intent and dialogue, and text to speech (ElevenLabs, Azure, AWS Polly) for natural responses. It can handle open ended speech, context, multi turn conversation, and integrate with your CRM. We build both, and pair them so the bot handles the volume, the human handles the exception.
Yes. We are an India based team and have built outbound dialing, voice broadcast and SMS platforms that comply with TRAI rules, DND scrubbing, DLT registered template enforcement and consent capture. We can advise on operator routing, CLI rotation policies and the practical realities of telecom regulation in India for clients targeting the local market.
Both are SIP proxies and SBCs derived from the same lineage. Kamailio has the larger community and more polished modules for the most common SBC scenarios. OpenSIPs has stronger commercial backing and certain advanced routing primitives. We choose based on the workload: Kamailio for general SBC, registration, NAT traversal and load balancing in front of FreeSWITCH/Asterisk; OpenSIPs when complex dynamic routing, B2BUA features or specific commercial modules are needed.
Yes. We build WebRTC applications using JsSIP, SIP.js and the verto protocol against FreeSWITCH, plus signalling layers via Kamailio. Use cases include browser softphones, in app voice and video calls, embedded support widgets, virtual contact centre agents and click to call features inside SaaS dashboards. STUN/TURN setup, codec negotiation, ICE traversal and audio quality tuning are part of the standard delivery.
Yes. We routinely integrate FreeSWITCH and Asterisk events with Salesforce, HubSpot, Zoho, Zendesk, Freshdesk and custom Node.js or Java backends. Real time popups on inbound calls, click to dial, automatic call logging, call recording links in CRM, post call transcription and sentiment analysis, all standard pieces of the work.
The fixed price PoC covers a 4 to 6 week engagement: requirements workshop, architecture document, a working FreeSWITCH or Asterisk based prototype with one or two real call flows, a basic CRM or web integration, call recording and a written report on production hardening, scale and cost. You leave with a runnable system you can demo to stakeholders and a defensible plan for the production rollout.
Yes. Many clients come to us with an existing FreeSWITCH, Asterisk or Kamailio deployment that needs hardening, performance tuning, version upgrades, security patches or call quality investigation. We do platform audits, Wireshark/SIP trace analysis and packet capture investigation, then write a remediation plan with effort estimates. Ongoing support is available on a $6,000/month dedicated retainer.
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