Sub-400ms Real-Time Voice AI Engineered for Natural Human Cadence.
Deploy enterprise-grade autonomous voice agents with native bidirectional audio streaming, instant interruption handling, and seamless Bring Your Own Carrier (BYOC) SIP Trunking.
Test Real-Time Conversational Cadence
Turn-by-Turn Audio Latency Breakdown
Compared to legacy IVR systems (3,500ms+) and cascading text bots (1,800ms+), Lokam delivers natural sub-400ms human turnaround times.
Carrier-Grade Voice AI Architecture
Engineered for high-volume telecommunication workloads with zero compromise on audio fidelity, stability, or compliance.
Instant Barge-In & Interruption
Humans interrupt and talk over each other naturally. When a user speaks, Lokam halts playback in <35ms and recalculates conversational trajectory without repetitive restarts.
BYOC SIP Trunking & Carrier Interconnect
Bring your existing numbers from Twilio, Plivo, FreePBX, or Asterisk. Route calls via sub-millisecond DID tables to specific tenant agent flows with zero phone number porting.
Natural Fillers & Conversational Breathing
Eliminate awkward dead air during heavy database queries. Pre-cached conversational fillers ("mhm", "one moment please") stream in <50ms while dynamic tools retrieve customer data.
Warm PSTN Transfer & Live Handoff
When callers request human escalation, Lokam bridges the call to staff phone lines, injecting a structured call summary and live conversation transcript directly into the CRM.
Enterprise Safety Watchdogs
Hard 15-minute call duration caps protect against runaway telecom bills. Automated silence watchdogs re-prompt after 6.5s of dead air and softly disconnect or transfer after 15s.
In-Memory PII Redaction
Real-time privacy filters redact credit cards, Social Security Numbers, and passwords in memory before call transcripts or logs are written to database storage.
Telephony Specifications
| Metric / Specification | Lokam Voice Engine | Standard Cloud IVR |
|---|---|---|
| End-to-End Turn Latency | < 380ms | 1,800ms – 3,500ms |
| Audio Quality | Ultra-Clear HD Voice | Low-bitrate compressed audio |
| Interruption Cutoff (Barge-In) | < 35ms | Disabled / Collides |
| Carrier Integration | BYOC SIP Trunking (Twilio, Plivo, PBX) | Locked to single proprietary carrier |
| Concurrent Channels | Elastic (Up to 10,000+ simultaneous calls) | Fixed port limits with queue wait |
Experience Sub-400ms Voice Telephony
Connect your SIP trunk or test live in your browser with zero telephony setup.