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frejun.com

Programmable voice calling infra for developers

The voice platform for business teams and developers.
Automated calling workflows for your team.
Programmable voice infrastructure for your applications.

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THE PROBLEM

Why legacy telecom infrastructure fails
Al-to-human conversations

Regional data residency

Standard PSTN and SIP routing add delay that breaks conversational flow. When round-trip time exceeds a few hundred milliseconds, the AI and user end up talking over each other.

No Raw Audio Access

Most CPaaS providers don’t offer low-level access to raw media streams. Without it, developers can’t implement custom Voice Activity Detection or reduce AI processing lag.

Human-First Architecture

Legacy stacks are built for human-to-human sessions. They lack the hooks needed for instant interruption handling, dynamic media injection, and millisecond-perfect state changes.

Infrastructure Management Overhead

Managing carriers, SIP trunks, and global telephony routing distracts product teams. Developers spend more time on telecom plumbing than on optimising the actual AI model and user experience.

PLATFORM FEATURES

Programmable voice infrastructure and
real- time audio streaming at scale

Choose the approach that fits your architecture. Both support global numbers, transparent debugging, and programmable call control.

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Voice Apps

Real-time audio streaming via WebSocket. Stream audio between phone calls and your application in real-time.

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SIP Trunking

Enterprise-grade and programmable SIP infrastructure. Connect your existing SIP-enabled platforms directly to our carrier network.

CORE INFRASTRUCTURE

Everything you need
to ship AI-enabled voice features

Real-time Media Streaming
Click to Call

Real-time, bidirectional audio streaming between phone calls and your server.

Click to Call

Power through candidate lists with progressive dialing. Move from one screening to the next, with automated voicemail drops when they don’t pick.

Click to Call
Candidates see a local number from their region and actually pick up. Get numbers in UK, UAE, US, India, Canada, Australia, and 50+ countries.
Click to Call
Send job descriptions, interview links, or “nice to meet you” notes via WhatsApp immediately after a call.
Click to Call
Send job descriptions, interview links, or “nice to meet you” notes via WhatsApp immediately after a call.
Real time Media Streaming
Global Phone Numbers(1)
SIP Trunking(2)
Programmable Call Control
Call Recording & Analytics(1)

How it Works

FreJun Teler streams live call audio to your AI backend via WebSocket, enabling real-time processing and seamless AI-generated responses during the conversation.

Connection & Webhook Trigger

When a call hits your FreJun Teler number, we instantly send a webhook to your application server with all necessary session metadata.

Bidirectional WebSocket Establishment

Your server opens a full-duplex WebSocket, and FreJun begins streaming raw audio frames (L16/8000Hz) from the caller to your endpoint with sub-250ms latency.

Real-Time AI Orchestration

Your backend pipes the audio through your AI stack (STT → LLM → TTS) and streams the synthesized response back through the open socket.

Low-Latency Media Injection 

FreJun Teler receives the AI’s audio stream and injects it into the live call. Our platform handles the complex jitter buffering and synchronization required to ensure the conversation remains fluid and natural.

USE CASES

What developers build

with FreJun Teler

Al Voice Agents

Connect phone calls to your AI models via real-time audio streaming. Build schedulers, support agents, etc. with AI-to-human handoffs.

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Intelligent IVR

Build multi-level IVR flows with keypress-based menus. Route callers to the right team based on their input for faster resolution.

Customer Support

Voice-enabled support systems that handle queries, access customer data, troubleshoot issues, and escalate to humans when needed.

Outbound Campaigns

Programmatically initiate outbound calls for lead qualification, outreach, or appointment booking. Build logic that adapts based on responses.

Voice-Enabled SaaS

Add click-to-call, call recording, transcription, or voice features directly into your product. Embed telephony without building infrastructure.

Certified Security Frameworks for Reliable Voice AI

SOC2 Type 2 Certified
Enterprise security standards
ISO 27001 Certified
Information security management
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GDPR Compliant
Data privacy compliance
WHY TELER

Beyond generic CPaaS. 
Beyond raw SIP trunking

Purpose-Built for AI Interaction

Unlike general-purpose CPaaS platforms designed for broad use cases like SMS and IVRs, Teler is optimized specifically to simplify the developer experience of real-time voice and media coordination.

Simplified Integration Logic

Teler abstracts the complexity of managing live audio streams and response coordination, allowing your team to connect phone calls directly to automation workflows without needing deep telecom expertise.

Conversation-First Engineering

By providing a bridge for real-time audio streaming between callers and your application, Teler lets you focus on conversation intelligence and model quality rather than media handling.

High-Performance Abstraction

Teler provides the performance of SIP trunking while eliminating the engineering burden of managing media negotiation, scaling, and carrier reliability through a ready-to-use application interface.

Works with your

favorite tools

FreJun connects natively with Salesforce, HubSpot, 
Zoho, Freshworks and other CRMs, ATS platforms, and helpdesks.
Two-way sync keeps your data accurate across every tool
your team already uses.

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Frequently Asked Questions

Have more questions?

Reach out to us at support@frejun.com.

What is FreJun Teler?

FreJun Teler is a programmable voice API and telephony infrastructure for developers building AI phone agents, real-time voice applications, and custom voice workflows. FreJun Teler handles carriers, audio streaming, and call control so you can focus on your product.

FreJun Teler streams audio between live phone calls and your application in real time via WebSocket, with sub-250ms latency. You receive caller audio as it happens, process it with any AI provider, and stream responses back instantly.

Voice Apps give you direct WebSocket-based calling with full audio access for custom AI voice automation. SIP trunking connects your existing voice platforms or AI providers via standard SIP protocol, FreJun Teler handles media streaming and carrier complexity on both.

FreJun Teler is model-agnostic. Connect ElevenLabs, Vapi, Retell, Livekit, OpenAI, Anthropic, Deepgram, or any custom model via real-time voice streaming infrastructure. There is no lock-in and you pay your AI provider directly.

Yes. FreJun Teler offers SDKs and a Conversations API for embedding real-time communications and telephony infrastructure directly into your application.

Channels are the number of simultaneous calls your account can handle across Voice Apps or SIP trunks. They are shared across your account and can be configured via the channel limit attribute. More channels means more concurrency.

Yes. FreJun Teler provides prototyping credits so developers can test and build without a credit card.

Contact the FreJun team for specific SLA details based on your plan.

Get started with FreJun today

Everything your team needs to call, log, and analyse every conversation, 
set up and running in hours.