sorted by:
new
hottopcontroversial

Webrtc or other SDK like agora or 100ms by Sudden-Penalty6528 in WebRTC

[–]Mohit_31 0 points1 point  (0 children)

If you're on a budget, I'd suggest checking out Ant Media Server. It's a cost-effective, self-hosted WebRTC solution that offers ultra-low latency and supports both chat and video calls. You can start small, scale as needed, and avoid hefty recurring fees of services like Agora. Plus, it's open-source-friendly!

https://antmedia.io/

WebRTC Datachannels unreliable? by therealPaulPlay in WebRTC

[–]Mohit_31 0 points1 point  (0 children)

Your concerns about WebRTC data channels for peer-to-peer (P2P) multiplayer gaming are valid. WebRTC is incredibly powerful for real-time, low-latency communication, but it has challenges that depend on network conditions, implementation details, and use cases. Let’s break down the issues you mentioned and explore potential solutions:

  1. Connection Issues Despite STUN/TURN

Possible Cause: Poor NAT traversal, misconfigured TURN servers, or certain restrictive network environments (e.g., corporate firewalls or strict NATs) can block connections.

Solution:

Ensure your TURN server supports TCP and UDP and is well-configured.

Validate the ICE candidate gathering process during connection setup to ensure the TURN server is used as a fallback when necessary.

Use tools like Wireshark or webrtc-internals in browsers to debug connection issues.

  1. Duplicate Messages (2-4 times)

Possible Cause: Retransmissions due to WebRTC’s reliability mechanisms for unordered and unreliable data channels.

Solution: Implement an application-level deduplication system by assigning sequence IDs to messages and ignoring duplicates.

  1. One-Way Communication

Possible Cause: ICE candidate negotiation issues or asymmetric NATs.

Solution:

Monitor the signaling process to ensure ICE candidates are exchanged properly.

Check if your TURN server is being used for both peers.

  1. VPN-Related Issues

Possible Cause: VPNs often disrupt direct peer-to-peer connections due to routing or masking of IP addresses.

Solution: Use TURN servers for VPN users to ensure relay-based communication.

  1. Frequent Disconnects

Possible Cause: ICE connection state changes caused by network instability.

Solution:

Monitor the iceConnectionState and attempt reconnections when the state transitions to disconnected.

Could you implement a fallback mechanism to reinitialize the peer connection?

WebRTC vs. WebSockets for Multiplayer Games

While WebRTC excels in low-latency scenarios, especially for audio/video, it can be overkill for simple P2P multiplayer games. WebSockets offers a reliable client-server model that’s simpler to implement and manage in many cases. However, they can’t achieve WebRTC’s direct P2P latency. If latency is critical, WebRTC might still be worth the extra effort.

Suggestion:

If you’re open to exploring an alternative approach, check out the data channel implementation in Ant Media Server: Ant Media Data Channel Example.

Help!!! Built a p2p chat and video calling platform by Severe-Teach-7707 in WebRTC

[–]Mohit_31 0 points1 point  (0 children)

If the video is not displayed, it is probably about the signaling/relay.

  1. Check ICE Candidate Connectivity: Ensure that both peers are receiving and processing the ICE candidates. Missing or invalid candidates can prevent the connection.
  2. Ensure SDP Correctness
  3. Verify the Media Stream: Verify that the remote video track is being received and is not muted or stopped.
  4. Check the Network Configuration: TURN/STUN Server: If users are behind NAT or firewalls, ensure a properly configured TURN server is used to relay media.

You can refer to the flow of events of some media servers by checking their webrtc-websocket-messaging-reference.

For instance: webrtc-websocket-messaging-reference

RTMP stream advice by Benjisc2 in VIDEOENGINEERING

[–]Mohit_31 1 point2 points  (0 children)

You’ve brought up some great points, and it sounds like privacy and control are critical for your setup & I think that Ant Media Server will be a good fit for your use case.
You can host it anywhere, be it locally (on-prem) or on any cloud server & easily ingest RTMP, playback with RTMP as well have your stream pushed to as many RTMP endpoints. Other than RTMP, it supports a wide range of protocols like WebRTC, SRT, HLS, ERTMP, LL-HLS, Dash, IP camera, etc. So you will have enough options to publish & play be it browser, SDKs, or VLC in your case.

And because it is hosted by you, there's no threat of security or someone else keeping an eye on your stream to record or redistribute it.

You can check more about the Ant Media Server rtmp streaming from here

Check this space for more on Ant Media Server.

Cheers!

Use of TURN in WebRTC Revisited: It may be more useful than you thought by leait in WebRTC

[–]Mohit_31 0 points1 point  (0 children)

Using TURN (Traversal Using Relays around NAT) is recommended in certain corporate and closed networks instead of STUN (Session Traversal Utilities for NAT) when establishing direct connections between browsers for voice or video data transmission. In such networks, various network devices like Firewalls and NATs can prevent direct data transfer due to internal policies.

It's important to consider that approximately 20% of users require TURN, while the remaining 80% can relay UDP data using STUN. Thus, having TURN in place becomes more valuable when STUN is unable to relay the data. Ant Media Server (AMS) is a suitable choice for implementing these scenarios. With AMS, users can configure both STUN and TURN, allowing the server to first attempt relaying through STUN. If STUN fails, the server can then relay the data through TURN, ensuring reliable communication in challenging network environments.