Carrier-grade voice, driven by your API.
Asterisk and Kamailio platforms, SIP trunking, contact centre and dialer systems - with a REST and webhook layer over the switch so your product team can drive calls without learning telephony internals.
What we actually deliver.
PBX & softswitch
Asterisk for call handling behind Kamailio for signalling and load distribution - the split that keeps registration load off the media layer.
- Asterisk dialplan and ARI applications
- Kamailio SIP proxy and registrar
- High availability and failover design
SIP trunking & carriers
Carrier and gateway integration including the parts that cause most incidents - NAT traversal, session border configuration and trunk failover.
- Multi-carrier trunking and least-cost routing
- NAT traversal and SBC configuration
- GSM and PSTN gateway integration
Contact centre & dialer
Queues, skills-based routing, IVR, predictive and progressive dialing built around real agent workflow rather than a vendor's default.
- Queues, skills routing and IVR
- Predictive, progressive and preview dialing
- Agent state, wrap-up and disposition
APIs over the switch
A REST and webhook layer over ARI and AMI so your application drives calls through a normal API instead of speaking SIP.
- REST call control and webhooks
- CRM screen pop and click-to-dial
- CDR and interaction logging
Recording & quality
Selective recording with payment-segment pause, encrypted at rest, plus per-leg quality monitoring rather than waiting for complaints.
- Selective recording, PCI pause
- Encryption at rest and access logging
- MOS, jitter and packet loss per leg
Security & fraud control
Toll fraud is the expensive failure mode in VoIP - controls belong in the build, not in the incident review after the bill arrives.
- Rate limiting and fraud detection
- TLS and SRTP for signalling and media
- Destination allow-lists and spend caps
A sequence that de-risks delivery.
Map the call flows
How calls actually arrive, route and end - including the paths staff use as workarounds, which is where the real requirements hide.
Design signalling and media separately
Registration load and media load scale differently. Separating them early is what keeps the platform stable as concurrency grows.
Build and integrate
Switch, trunks and the API layer, wired into your CRM or product so telephony is part of the workflow rather than beside it.
Load test, then cut over
Concurrency tested to a multiple of expected peak before any traffic moves, with the old route kept available during transition.
What clients measure afterwards.
Quality measured
MOS and jitter tracked per call leg, so degradation is caught before complaints.
Per-seat cost removed
Licensing replaced by infrastructure you control, where agent count justifies it.
Telephony in the product
Calls driven by your application through REST, not a separate console.
Fraud contained
Spend caps and destination controls before the first unexpected bill.
Work we have actually delivered.
Built self-hosted HD video calls with recording, no per-seat fees.
- 2160p
- 4K-capable (H.264 L5.1)
- ~2%
- One core per recording stream
Built a Slack-style team chat with live presence for the workspace.
- 45s
- Presence TTL · 30s heartbeat
- RLS
- Org-isolated per frame
Tools we work fluently in.
- Asterisk
- Kamailio
- FreeSWITCH
- RTPengine
- SIP / SDP
- Opus
- G.711
- G.729
- SRTP
- WebRTC
- ARI
- AMI
- REST & webhooks
- PostgreSQL
- Node.js
- Homer / SIP capture
- Prometheus
- Grafana
- Fail2ban
Senior engineers, accountable outcomes.
We run switches in production
Kamailio in front of Asterisk with GSM gateways, carrying real traffic. The traps we design around are ones we have debugged at 2am.
An API your product team can use
We put REST and webhooks over ARI and AMI so application developers drive calls without learning SIP, dialplans or telephony internals.
We fix before we replace
One-way audio and drops under load are usually NAT, SBC or trunk configuration - not the switch. We would rather fix the cause than sell a migration.
Fraud controls from day one
Toll fraud is the expensive VoIP failure. Spend caps and destination allow-lists are part of the build rather than a lesson learned later.
Questions, answered.
The questions CTOs, contact centre leads and product owners ask us most often before a voice platform engagement.
- Often you should not - hosted platforms are the right answer for standard business telephony. Self-hosting wins when per-seat pricing has become material at your agent count, when you need call flows a hosted product will not expose, or when the telephony has to be embedded in your own application rather than sit beside it. Contact centres and dialer products are the usual cases, because that is where per-seat economics and control requirements both bite.
Book a telephony call - leave with an architecture.
A 45-minute session with an engineer who operates voice platforms. We map your call flows, concurrency and integration needs - and you leave with an architecture sketch, whether you engage us or not.
Tell us about your project.
Briefs, NDAs, architecture reviews, anything goes. A senior engineer responds within 24 hours.
- Email[email protected]
- Phone+91 99228 74956
- LocationsHyderabad · Pune · Bangalore
- 1 · A senior engineer reviews your brief within 24 hours.
- 2 · We schedule a 30-minute discovery call.
- 3 · You receive a written proposal in 48-72 hours.