# SIP/PSTN Telephony Tutorial Series for MediaSFUOpen Users

**URL:** https://mediasoup.discourse.group/t/sip-pstn-telephony-tutorial-series-for-mediasfuopen-users/6855
**Category:** Off Topic
**Created:** [September 2, 2025, 10:08am UTC](https://mediasoup.discourse.group/t/sip-pstn-telephony-tutorial-series-for-mediasfuopen-users/6855 "2025-09-02T10:08:09Z")
**Posts on this page:** 1
**Page:** 1

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### Author: ![Bdayz](https://avatars.discourse-cdn.com/v4/letter/b/da6949/32.png) [@Bdayz](https://mediasoup.discourse.group/u/Bdayz)
#### Post date: [September 2, 2025, 10:08am UTC](https://mediasoup.discourse.group/t/sip-pstn-telephony-tutorial-series-for-mediasfuopen-users/6855/1 "2025-09-02T10:08:09Z")

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We’ve released comprehensive (but short) video tutorials specifically for MediaSFUOpen users wanting to integrate SIP telephony and PSTN connectivity powered by [MediaSFU](https://mediasfu.com) 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:**

- Complete Tutorial Playlist: [https://www.youtube.com/playlist?list=PL1JaYsvPqLVz-ucQ8OrT4bWVkJO3kY1kD](https://www.youtube.com/playlist?list=PL1JaYsvPqLVz-ucQ8OrT4bWVkJO3kY1kD)
- Full Combined Tutorial: [https://youtu.be/d21ppDGpR2k](https://youtu.be/d21ppDGpR2k)

**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](https://mediasfu.com/telephony)

**AI Voice Agents:** Test configurations at [agents.mediasfu.com](https://agents.mediasfu.com) and [playground](https://agents.mediasfu.com/playground)

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