Why WebRTC Media Streams Fail on Cloud Servers Even When Signaling Works

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A developer deploying a real-time WebRTC streaming server to a cloud VM documented a series of infrastructure-level failures unrelated to application code. The core issue stems from WebRTC using two separate network paths: signaling travels over WebSocket on TCP port 443 through standard proxies, while actual audio and video travel via RTP over UDP directly between browser and server, bypassing all proxy infrastructure by design. Cloud VMs typically only see their private IP address, causing the server to advertise unreachable ICE candidates in its SDP, resulting in sessions where the WebSocket appears healthy but no media arrives. The reliable fix for one-to-one NAT environments is to explicitly configure the server's public IP using Pion's SetNAT1To1IPs setting, rather than relying on STUN-based address inference, which can fail with certain NAT types. After applying this configuration, the author observed a stable 30fps stream with a 7ms round-trip time, eliminating the silent connection failures entirely.

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Agentic Design System sandeep kumar patel Sep 19 Add Comment 1 min read

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