SIP/PSTN Telephony Tutorial Series for MediaSFUOpen Users

We’ve released comprehensive (but short) video tutorials specifically for MediaSFUOpen users wanting to integrate SIP telephony and PSTN connectivity powered by MediaSFU Cloud.

New 9-Part Tutorial Series (1+ hrs total):
Step-by-step guide showing how MediaSFUOpen (self-hosted) connects seamlessly with MediaSFU Cloud for SIP/PSTN functionality, enabling traditional phone network connectivity without local telephony infrastructure.

Architecture Covered:

  • MediaSFUOpen Community Edition handling local WebRTC infrastructure
  • MediaSFU Cloud manages SIP protocol complexity and PSTN bridging
  • Seamless integration between self-hosted and cloud components
  • No local SIP servers or telephony hardware required

Series Coverage:

  • SIP trunking configuration through the MediaSFU Cloud dashboard
  • PSTN connectivity setup for Community Edition users
  • Telephony provider integration (detailed Twilio walkthrough)
  • Voice AI agent deployment for SIP calls
  • Real-time SIP call control via HTTP APIs
  • Enterprise telephony platform access

Video Resources:

Live SIP Integration Testing:

  • Mixed Support Demo: +1 785 369 1724
  • AI Conversation: +44 7445 146575
  • Technical Support: +1 587 407 1990
  • Friendly AI Chat: +1 647 558 6650

Key Benefits for MediaSFUOpen Users:

  • Keep your self-hosted video infrastructure
  • Add telephony via cloud without complexity
  • Cost-effective at $0.10 per 1000 minutes for SIP processing
  • Maintain control over core WebRTC components

The tutorials demonstrate how Community Edition users can leverage cloud-based telephony services while preserving their self-hosted MediaSFU deployment for video functionality.

SIP/PSTN Guide: mediasfu.com/telephony

AI Voice Agents: Test configurations at agents.mediasfu.com and playground

This hybrid approach provides MediaSFUOpen users with enterprise-grade SIP/PSTN capabilities, without compromising the benefits of self-hosting their primary video infrastructure.