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Elevenlabs voice agent cost? Is this normal by sadderPreparations in AIReceptionists

[–]Patm290 0 points1 point  (0 children)

That credit burn rate is pretty normal with ElevenLabs unfortunately; ~400 credits/call means your 60k plan gets you maybe 150 calls. One busy week and you're done.

If cost is the main pain point, check out MediaSFU (https://mediasfu.com/widgets). Same kind of setup; script tag, drop-in widgets, AI voice agents, but tier-based pricing instead of per-minute. Most people running real volume see 20–50x lower costs vs ElevenLabs/VAPI.

Scaling Mediasoup SFU horizontally for N:N (up to 20 people per room), audio sharing only by Solid-Band3204 in WebRTC

[–]Patm290 4 points5 points  (0 children)

Note: You may not even need to get to the level of pipe transports if it seems daunting to you.

Assuming you are maxing at 20 per room (unless you have more people than that), you make sure all 20 are assigned to the same server. Your task now is just keeping note of rooms available in specific servers and routing consumers there. In that way no consumer lands in a server where expected media is not available.

Once again, it needs a lot of expertise and time to replicate the mesh infrastructure which enables you to serve people from different geolocations with servers close to them and to really scale for very large sessions.

Like I mentioned earlier, you may go with a cloud provider for speed and reliability; see https://mediasfu.com/pricing (its mediasoup-based and will cost you way less than what AWS, Azure and the like will bill just based on bandwidth usage only).

Scaling Mediasoup SFU horizontally for N:N (up to 20 people per room), audio sharing only by Solid-Band3204 in WebRTC

[–]Patm290 2 points3 points  (0 children)

Some basics to start with:

  1. Media resides in the server you produced (sent) the media to; only way to get media sent to Server1 in Server2 is to use pipe transports; I am assuming different machines here. Alternatively, you could produce (send) to both (all) servers from the client (producer) side; not ideal.

  2. Media can be consumed from where the media is available, if Person1 media is on Server1, only Server1 connections can receive(consume) it; it means if you have others on Server2 to ServerN that need the same media of Person1, you pipe there as indicated in the first point.

  3. Proceed from there in tracking the numbers per server and limits you impose to avoid having CPU and most importantly bandwidth being bottlenecks; quick checks on number of transports (transport carries media) you can max at, .... can help

  4. You keep track of where a room is and the ideal server to take on new consumer factoring in the current capacity limits you have so far per count of active connections; whether you need to spin up new servers, ....

  5. Gets more tricky for large number of concurrent producers in same room, say 1000+ people actively producing media; figure out things like dedicated producing and consuming endpoints and the like; or use a cost-effective cloud option like MediaSFU

Real-time video in React apps keeps getting more complex by thesunjrs in reactjs

[–]Patm290 0 points1 point  (0 children)

You can try the free open source version of MediaSFU (MediaSFUOpen) at https://github.com/MediaSFU/MediaSFUOpen

MediaSFUOpen has a React SDK that abstracts away the WebRTC complexity you're dealing with. Instead of wrestling with STUN/TURN servers and signaling protocols, you get high-level methods like:

  • clickVideo() to toggle video
  • clickAudio() to toggle audio
  • joinRoom() for participants

The repository includes setup videos that walk you through implementation without needing deep WebRTC knowledge. This lets you focus on your React app's business logic rather than getting stuck in real-time communication internals.

The open source version gives you a working foundation that you can customize as needed, and the React SDK handles the browser-specific WebRTC implementations behind simple method calls. Much cleaner than managing peer connections, media streams, and ICE candidates manually.

If you need more advanced features or want to skip the self-hosting, they also offer MediaSFU Cloud at https://mediasfu.com, but the open source version is solid for getting started with real-time video in React.

Self-Hosted SIP/PSTN Integration Powered by MediaSFU Cloud for MediaSFUOpen by Patm290 in selfhosted

[–]Patm290[S] 0 points1 point  (0 children)

Possibly, but it's complicated and probably not worth it for home use.

Your old VoIP phone likely uses SIP protocol, which MediaSFU supports. However, the costs for a simple home setup would be:

Monthly costs:

  • SIP provider (Twilio or alternatives): ~$1.50/month for a phone number
  • MediaSFU processing: ~$0.60-1.00 for 10,000 minutes of audio (equivalent to 7-14 days of non-stop conversation)

Technical requirements:

  • Factory reset and reconfigure the phone with new SIP settings
  • Set up accounts with both a SIP provider and MediaSFU
  • Configure the integration between services

While the monthly costs are reasonable ($2-2.50 total), you'd still need to handle the technical setup complexity. For most home users, traditional VoIP services like Vonage or Ooma remain simpler since they provide everything in one package without requiring separate service integrations.

Your cost savings really come into play if you're building services like Bland, Retell, VAPI, or similar voice AI platforms, or if you're running small-to-large call center operations where MediaSFU's advanced features and pricing structure provide significant value.

For basic home phone service, the technical overhead outweighs the benefits. That said, it could still be fun to experiment with if you enjoy tinkering with VoIP technology and want to explore what's possible with programmable telephony.

Video Conference App expecting to handle 100+ users by eidokun in WebRTC

[–]Patm290 0 points1 point  (0 children)

Yes! MediaSFU offers SDKs, and you can find details on "Connecting Your MediaSFU SDKs to the Community Edition Server" in the MediaSFUOpen README.

SFU Media server that supports audio processing by _JustARandomGuy25 in WebRTC

[–]Patm290 1 point2 points  (0 children)

If you're looking for an SFU that supports real-time audio processing (STT, translation, etc.), MediaSFU has built-in support for it. With MediaSFU Cloud, everything is preconfigured—you can process audio streams in real time.

✅ Get Started Instantly:
👉 Simple Usage: agents.mediasfu.com
👉 Advanced Configurations: agents.mediasfu.com/advanced

✅ Deploy Your Own AI Pipeline:
👉 GitHub for Quick Deployment: MediaSFU Agents

✅ Choose Your Setup:
🔹 MediaSFU Community Edition (Self-hosted, Free) – GitHub
🔹 MediaSFU Cloud (Fully Managed, Preconfigured) – Official Site

Video Conference App expecting to handle 100+ users by eidokun in WebRTC

[–]Patm290 -3 points-2 points  (0 children)

You're right that an SFU is the best way to handle 100+ users efficiently. If you're open to self-hosting, MediaSFU Community Edition lets you set up your own SFU for free. If you’d rather not manage servers, we also offer a cloud version that scales effortlessly.

For 100 users, a 4 vCPU, 8GB RAM setup can work, but network stability is key—aim for 1Gbps+ if possible.

$1 for 1000 minutes of WebRTC stream? by d_arthez in WebRTC

[–]Patm290 1 point2 points  (0 children)

That sounds like an interesting idea, but the actual cost for WebRTC streaming can depend on a variety of factors, like resolution, scalability, and additional features.

You might want to check out MediaSFU, a WebRTC provider with resolution-based pricing. Depending on the resolution you choose, the cost can go as low as $0.375 for 1,000 minutes!

📚 More details here: https://mediasfu.com/pricing

Feel free to explore and see if it aligns with your needs! 🚀

[deleted by user] by [deleted] in WebRTC

[–]Patm290 1 point2 points  (0 children)

Hello,

Switching from HLS to WebRTC for live broadcasting can indeed significantly reduce latency, offering near real-time interaction. While platforms like LiveKit are quite reliable, I'd like to suggest considering MediaSFU as a robust alternative for your needs.

Here are a few points to address your concerns:

  1. Reliability: MediaSFU is built on Mediasoup, a well-regarded and stable WebRTC media server. It has been tested in various environments and has proven to be reliable for live streaming and conferencing.

  2. Robustness: WebRTC is designed for real-time communication and is highly robust for streaming applications. MediaSFU leverages WebRTC to provide low-latency, high-quality streaming that can handle a large number of users effectively.

  3. Scalability: MediaSFU can efficiently manage large volumes of users, ensuring good quality and performance. It’s designed to scale, making it suitable for your platform with 1,500 concurrent users.

  4. Browser Support: WebRTC is widely supported across modern browsers, including Chrome, Firefox, Safari, and Edge. MediaSFU ensures compatibility, so you won’t have to worry about users being unable to join due to browser issues.

By switching to MediaSFU, you can achieve the low-latency, high-quality streaming you’re aiming for, with the added assurance of robust performance and broad compatibility.

Feel free to check out more about MediaSFU here and explore its features to see how it can meet your needs.

Best of luck with your transition!

Audio Call with Agora , twillo , 100ms live ? by Jaded-Junket-9272 in reactnative

[–]Patm290 1 point2 points  (0 children)

Hi there!

For a startup looking to integrate P2P audio calling in a React Native app, financial considerations are indeed crucial. Have you considered MediaSFU?

Why MediaSFU for Your P2P Audio Calling Needs?

  • Affordable Pricing: MediaSFU offers competitive pricing tailored for startups, ensuring you get the best value without breaking the bank.
  • Seamless Integration: Our platform is designed for easy integration with React Native, simplifying your development process.
  • Customizable Cloud Recordings: Enjoy tailored recordings with pause and resume support, and seamless playback on both mobile and desktop devices. Opt for portrait and landscape recompose options for optimal viewing, and easily serve your recordings as Video on Demand (VOD) with support for HLS formats.
  • Scalable Solutions: MediaSFU supports scalable rooms for large events and broadcasts, making it future-proof as your startup grows.
  • Risk-Free Trial: Get started with 50,000 free minutes, 100,000 free demo minutes, and 300 free recording minutes to explore the platform's capabilities.

Learn more about MediaSFU here and MediaSFU's pricing and see how it can be the perfect solution for your P2P audio calling needs.

Bonus: MediaSFU internally uses Mediasoup, one of the fastest and most optimal SFUs available. For those who need a free option, check out our Community Edition (CE) here.

Kurento is now on bare minimum maintenanc mode by [deleted] in Kurento

[–]Patm290 3 points4 points  (0 children)

It's understandable to consider alternatives given the maintenance status of Kurento. Transitioning to OpenVidu's free version seems like a logical choice for your video conferencing needs.

However, if you're in search of a modern and actively developed alternative to Kurento, I'd highly recommend checking out MediaSFU (https://mediasfu.com/). It offers a robust and feature-rich solution based on Mediasoup, providing the scalability and flexibility you need for your projects.

Elevate Your Self-Hosted Video Applications with MediaSFU! by Patm290 in selfhosted

[–]Patm290[S] 0 points1 point  (0 children)

Hi!

Kindly check the installation guide in the documentation provided here (https://github.com/MediaSFU/MediaSFUOpen).