modern infra stack
for voice AI.
A unified high-performance stack for the next generation of conversational UX. QUIC transport, bare-metal C++ compute, and sub-50ms latency.
Voice AI shouldn't feel like a Walkie-Talkie.
Legacy WebRTC infrastructure was built for human-to-human streaming, not low-latency AI inference. It carries massive architectural debt: unoptimized state machines, high compute overhead, and a total lack of developer-centric observability.
Latency & UX
WebRTC stack was built for human-to-human, not human-to-AI. 200ms of state-machine overhead kills conversational flow.
Transport Bloat
Heavy SDP/ICE handshakes and multi-layered protocol overhead add jitter. QUIC-native transport is 3x more efficient.
Infra Overhead
Running WebRTC gateways is CPU-intensive. Bare-metal C++ nodes reduce your cloud compute costs by 70%.
Maintenance Hell
Debugging WebRTC streams is a black box. Telequick provides unified observability and seamless migration paths.
The Live Showdown
Experience the difference between legacy transport and the Telequick stack. Watch real-time jitter, packet loss, and processing overhead.
Average Latency
248ms
CPU Usage
High (8-12%)
Protocol
SCTP/ICE
Jitter
> 45ms
Average Latency
38ms
CPU Usage
Low (< 1%)
Protocol
Unified QUIC
Jitter
< 3ms
Conversations
without the lag.
Voice AI fails when it's awkward. We've optimized the transport path to reach sub-50ms latency, enabling natural barge-ins and eliminating the "talking over" effect.
Reflex Barge-in
Proprietary "Halt" signal that cuts server audio on the exact millisecond of user detection.
Jitter Buffering
Self-healing audio streams that adapt to packet loss without audible artifacts.
The best of
WebRTC & QUIC.
WebRTC is the standard for browser audio, but it's brittle on mobile handovers. QUIC is robust but lacks browser support. We've bridged the two into a single, high-fidelity transport stack.
IP Mobility
Session IDs persist across WiFi-to-5G handovers. No socket drops. No reconnecting.
Stream Multiplexing
Audio, JSON, and Control signals travel on independent lanes. No Head-of-line blocking.
Bare-metal C++
Scaled to Millions.
Legacy voice gateways are often written in Ruby or Python, stalling under heavy audio concurrency. Telequick is built in pure, native C++ for elite throughput and 70% lower compute overhead.
Reduction in compute cost
Concurrent streams per node
Cost Effectiveness
Unified gateway architecture removes the need for multiple middle-tier abstractions.
Observability
& Maintenance Simplified.
Stop guessing why packets are dropping. Telequick provides deep-stream observability and one-click migration paths for legacy stacks.
Elite Observability
Real-time jitter, packet-loss, and TTFB metrics for every single data pod. Visual dashboards that make debugging a breeze.
Seamless Migration
One-click bridge for legacy WebRTC gateways and SIP trunks. No need to rewrite your entire backend logic.
Automated Maintenance
Self-healing nodes and predictive scaling. Focus on your AI model, while we handle the voice persistent connections.
Telemetry Engine v2
Actively Monitoring 1.2M Streams
Drop-in SDKs.
Bare-metal performance.
Replace hundreds of lines of WebRTC boilerplate with a single unified client. Our SDKs wrap a high-performance C++ core, exposing native-speed transport to your favorite language.
- Automatic recovery from 40% packet loss
- Built-in predictive interruption logic
- Direct-to-GPU memory mapping (Edge Compute)
- Real-time stream observability dashboard
import { Telequick } from '@telequick/sdk';
const client = new Telequick({
apiKey: process.env.TELEQUICK_KEY,
prediction: true // Enable predictive barge-in
});
client.on('interruption', (event) => {
console.log('User started speaking at:', event.offset);
stopLLMGeneration();
});
await client.connect();v1.2.4 Fully Operational
Scale from Prototype to Enterprise.
Flat $0.0005 / call and $0.0010 / minute on every cloud plan. Plans differ on support, SLA, and tenancy — never on the per-call rate.
Cloud
Multi-tenant edge, run by usFlat pricing across every cloud plan
No per-tier markup. Same rate whether you ship 1 call a day or a million.
Starter
Scale
Enterprise
Self-Hosted
Bare-metal binaries, run by youMega
Giga
Tera
Developer Ecosystem
Everything you need to build, scale, and optimize your voice applications on the Telequick network.
Developer Docs
Complete API references, SDK documentation, and architectural guides for low-latency voice.
Integrations
Connect Telequick with OpenAI, Anthropic, ElevenLabs, and your existing RTMP infrastructure.
Common Questions.
Everything you need to know about the future of Voice infrastructure.
How does Telequick handle false barge-ins (like a cough)?+
Telequick is the pipe, not the brain. Because our 0-RTT network is so fast, we give you your "latency budget" back. You can use that reclaimed 200ms to run a fast VAD (Voice Activity Detection) check. If the noise is just a cough, ignore it. If it's a real interruption, trigger the Telequick HALT stream.
Do I still need Twilio or a SIP provider?+
If you are building phone-based AI, yes. Telequick is the transport layer between the carrier and your AI. You point your SIP trunks at our gateway, and we convert the legacy audio into ultra-fast QUIC streams for your LLM.
Can I run this on my own servers?+
Yes. For enterprise customers, we provide native C++ binaries that deploy directly onto your VPC, keeping all audio data strictly within your own firewall for maximum compliance.
Why not just use WebRTC?+
WebRTC was built for P2P video conferencing, not client-to-server AI. It requires heavy browser binaries and complex ICE signaling. Telequick gives you raw, multiplexed data streams with a fraction of the overhead via our WASM payload.
Still have questions?
Our engineering team is ready to help you with your specific architecture.