![\"Fix](\"https:\/\/www.capanicus.com\/blog\/wp-content\/uploads\/2026\/05\/Fix-One-Way-Audio-in-SIP-Calls.webp\")
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You are on an important call, and suddenly, you can hear them, but they can\u2019t hear you. Or maybe it’s the other way around. If this sounds relatable, you are in the same frustrating room of SIP (Session Initiation Protocol) communications as millions of others. But the best part is, one-way audio SIP calls are almost solvable.\u00a0<\/span><\/p>\n We have come up with this step-by-step technical guide that\u2019ll help you fix one-way audio effectively and easily. Let\u2019s understand.\u00a0<\/span><\/p>\n In SIP-WebRTC scenarios, the signalling (SIP over WSS) can advance while the media (SRTP\/RTP) fails to flow in one direction. In layman’s language, it means the call is connected, but the RTP packets (the tiny bits of data that carry your voice) are getting lost while returning. The main culprits are:<\/span><\/p>\n As an experienced SIP development company<\/a>, we\u2019ve seen that if signalling is fine, but audio is missing one way, it\u2019s recommended to check ICE, SDP, and RTP paths first.\u00a0<\/span><\/p>\n Follow these important checks to quickly identify and resolve common audio, connectivity, and call quality problems.<\/span><\/p>\n Validate mic permission\/level and run a loopback test. In simple language, \u201cIs your mic working correctly?\u201d<\/span><\/b><\/p>\n Replace the call or trigger an ICE start. Make sure at least one public STUN server is configured. Consider it like reconnecting the call path cleanly.\u00a0<\/span><\/p>\n In Asterisk\/FreePBX, keep RTP anchored to the PBX\/proxy to skip leaking private IPs in SDP. In simple terms, let the server manage the call routing to avoid connection problems.\u00a0<\/span><\/p>\n Preferably choose Opus for audio. Ensure SRTP is required on the WebRTC leg and secured on the PBX\/trunk side. Basically, allow the call data to pass through the network security.\u00a0<\/span><\/p>\n Enable the PBX\/proxy\u2019s UDP RTP range bidirectionally or disable SIP ALG on edge routers. Primarily, it allows the call data to pass through the network security.\u00a0<\/span><\/p>\n For symmetric NAT\/CGNAT users, TURN is crucial. Validate credentials and reachability. Imagine it like a middleman server when a direct connection fails.\u00a0<\/span><\/p>\n Export browser getStats() and PBX SIP\/SDP traces. Confirm ICE \u201cconnected\u201d, valid selected candidates, and increasing SRTP packets both ways. In simple terms, test again and check the reports to make sure the issue is resolved.\u00a0<\/span><\/p>\n ICE, STUN, and TURN help WebRTC and VoIP systems<\/a> establish stable audio calls between users across different networks. One-way audio issues often appear because of the following scenario:<\/span><\/p>\n So, providing trusted ICE servers and a reachable TURN significantly minimizes one-way scenarios, especially for remote agents and mobile users.\u00a0<\/span><\/p>\nWhat \u201cOne-way audio\u201d Actually Means<\/b><\/h2>\n
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10-minute Triage Checklist for SIP Audio Troubleshooting<\/b><\/h2>\n
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Verify Browser Capture + Device\u00a0<\/b><\/h4>\n<\/li>\n<\/ul>\n
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Establish a Clean ICE Restart\u00a0<\/b><\/h4>\n<\/li>\n<\/ul>\n
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Temporarily Disable \u201cDirect Media\u201d at the PBX<\/b><\/h4>\n<\/li>\n<\/ul>\n
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Lock Codec to a Safe Baseline\u00a0<\/b><\/h4>\n<\/li>\n<\/ul>\n
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Open the Firewall for SRTP\/DTLS<\/b><\/h4>\n<\/li>\n<\/ul>\n
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Add or Enforce a TURN Relay\u00a0<\/b><\/h4>\n<\/li>\n<\/ul>\n
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Re-test and Record Diagnostics\u00a0<\/b><\/h4>\n<\/li>\n<\/ul>\n
Why ICE\/STUN\/TRUN Dictate Success in Internet Calls\u00a0<\/b><\/h2>\n
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Common PBX-side Challenges (Asterisk\/FreePBX Examples)<\/b><\/h2>\n
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