FreeSWITCH - Developer-Grade Telecom Engine

For Voice, Video & Real-Time Communication

FreeSWITCH is a powerful, scalable open-source platform for building voice, video, and messaging applications. Used by telecom startups, SaaS platforms, and CPaaS providers worldwide. Build custom communication solutions with unmatched flexibility and performance.

Core Features

🔄 SIP Routing Engine

High-performance SIP proxy and B2BUA for call routing and manipulation

🌐 WebRTC Support

Native WebRTC gateway for browser-based calling and video conferencing

📹 Video Calls

HD video calling with multiple codec support including VP8, VP9, H.264

🎤 Conferencing

Multi-party audio/video conferences with recording and streaming

🤖 IVR Builder

Flexible IVR creation with speech recognition and TTS integration

⚙️ Custom Dialplan

Powerful XML dialplan for complete call flow customization

📊 Carrier-Grade Performance

Handle thousands of concurrent calls with minimal latency

🏢 Multi-Tenant

Build SaaS platforms with isolated tenant environments

Why Choose FreeSWITCH?

🚀 Unmatched Scalability

Scale from 10 to 10,000+ concurrent calls with horizontal clustering

🔧 Complete Flexibility

Build any communication solution with extensive API and module system

💻 Developer Friendly

Comprehensive documentation, active community, and ESL for easy integration

🌍 Protocol Support

SIP, WebRTC, H.323, RTMP, and 30+ audio/video codecs supported

💰 Cost Effective

Open source with no licensing fees, only infrastructure costs

🔐 Enterprise Security

TLS/SRTP encryption, secure WebRTC, and advanced authentication

Perfect For

📱 OTT Calling Apps

Build WhatsApp-like voice/video calling apps with WebRTC

☁️ CPaaS Providers

Launch communication APIs like Twilio or Vonage

📞 SIP Providers

Offer business VoIP services with advanced PBX features

🎮 Gaming Voice Chat

Low-latency voice communication for multiplayer games

📺 Live Streaming

Interactive live streams with audio/video mixing and recording

🏢 Enterprise PBX

Custom PBX solutions with advanced call routing and features

🎓 E-Learning Platforms

Virtual classrooms with video conferencing and recording

🏥 Telemedicine

HIPAA-compliant video consultations and call recording

FreeSWITCH vs Asterisk

Feature FreeSWITCH Asterisk
Architecture Multi-threaded, Modern Single-threaded, Legacy
Performance 5000+ calls/server 500-1000 calls/server
WebRTC Native, Built-in Plugin Required
Video Support Excellent (native) Limited
Learning Curve Moderate Steep
Best For SaaS, WebRTC, Video Traditional PBX

Detailed Comparison Guide →

Technical Specifications

OS: Linux, Windows, macOS, BSD
Protocols: SIP, WebRTC, H.323, RTMP
Integration: ESL, REST API, XML-RPC
Deployment: Dedicated / Cloud / Container
Scaling: Horizontal Clustering
Security: TLS, SRTP, ZRTP Encryption
Codecs: G.711, G.729, Opus, VP8, H.264
Capacity: 10,000+ calls per cluster

Key Modules & Capabilities

mod_sofia

SIP stack for signaling and call control

mod_verto

WebRTC endpoint for browser calling

mod_conference

Multi-party audio/video conferencing

mod_event_socket

External control via ESL protocol

mod_lua/python

Scripting for custom call logic

mod_av

Video transcoding and manipulation

Our FreeSWITCH Services

Installation

Get Started Fast

  • Server setup & optimization
  • Basic dialplan configuration
  • SIP trunk integration
  • Documentation package
  • Email support (30 days)
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Custom Development

Full Service

  • Custom module development
  • API integration
  • Dialplan programming
  • Performance optimization
  • Architecture consulting
  • Dedicated engineer support
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Integration Capabilities

🔗 REST APIs

Control calls, conferences, and system via HTTP/HTTPS APIs

📡 Event Socket

Real-time event streaming for external applications

💾 Database

Direct integration with PostgreSQL, MySQL, MongoDB

📨 Message Queue

RabbitMQ, Kafka integration for event processing

☁️ Cloud Services

AWS, Azure, GCP native deployment support

🎯 CDR Export

Call detail records to custom databases or APIs

Frequently Asked Questions

What are FreeSWITCH server requirements?

Minimum 4GB RAM, 4 CPU cores for 100-200 calls. For high-volume deployments (1000+ calls), use 16GB+ RAM with SSD storage and dedicated network interface.

Is FreeSWITCH better than Asterisk?

For modern applications requiring WebRTC, video, and high scalability, FreeSWITCH is superior. Asterisk is better for traditional PBX deployments with legacy hardware.

Can FreeSWITCH handle WebRTC?

Yes, FreeSWITCH has native WebRTC support through mod_verto and mod_sofia. No additional gateways or plugins required for browser-based calling.

How do I build a CPaaS platform with FreeSWITCH?

We provide complete CPaaS development services including multi-tenant architecture, API design, billing integration, and white-label customization.

What programming languages work with FreeSWITCH?

FreeSWITCH supports Lua, Python, JavaScript, Perl, and Java. ESL (Event Socket Library) works with any language for external control.

Do you provide FreeSWITCH training?

Yes, we offer developer training covering installation, dialplan programming, ESL integration, and advanced features. Available on-site or remote.

Build Your Communication Platform

Start building with FreeSWITCH today. Our experts will help you architect and deploy a scalable telecom solution.