Call Quality Troubleshooting

This guide helps you resolve common call quality issues in CrystalQore: echo, choppy audio, one-way audio, and no audio. Solutions cover both WebRTC calls (browser-based) and PBX calls (via FusionPBX/SIP).

Common Issues

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Echo

Causes

• Acoustic echo — Your speakers are loud enough that your microphone picks up the sound
• Hardware echo — Headset or device feeding speaker output back into the mic
• Network echo — Less common in software; can occur with certain codecs or gateways

Solutions

1. Use a headset — Headphones prevent speaker sound from reaching the microphone
2. Use wired headphones — Bluetooth can introduce delay and echo
3. Lower speaker volume — Reduce the chance of mic pickup
4. Enable echo cancellation — CrystalQore and browsers use echo cancellation; ensure it's not disabled
5. Mute when not speaking — Reduces feedback if the other party has echo
6. Check your microphone — Ensure you're using the correct input device; avoid built-in mics when using external speakers

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Choppy Audio

Causes

• Packet loss — Network drops packets; audio sounds robotic or cuts out
• Jitter — Variable delay causes uneven playback
• Insufficient bandwidth — Connection can't sustain the codec bitrate
• CPU or resource limits — Device struggling to encode/decode in real time

Solutions

1. Use a wired connection — Ethernet is more stable than Wi‑Fi
2. Close bandwidth-heavy apps — Streaming, downloads, and backups compete for bandwidth
3. Switch Wi‑Fi bands — If on 2.4 GHz, try 5 GHz for less congestion
4. Check firewall — Ensure WebRTC ports (UDP/TCP) are not blocked
5. Test with a different browser — Chrome and Firefox generally handle WebRTC well
6. Reduce video quality — For video calls, lowering resolution frees bandwidth for audio
7. Restart your router — Resolves transient network issues
8. Test from another network — Rule out home or office network problems

WebRTC Ports

WebRTC uses a range of UDP and TCP ports. If you're behind a restrictive firewall, ensure:

• UDP — Typically 10000–60000 (configurable)
• STUN/TURN — If using TURN, the TURN server port must be open
• HTTPS — The CrystalQore site must be served over HTTPS for WebRTC (except localhost)

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One-Way Audio

Causes

• Microphone muted — System or app mute enabled
• Wrong input device — Browser or app using the wrong microphone
• Permissions — Microphone permission denied or revoked
• Firewall/NAT — One direction blocked (e.g., RTP from server to client)
• SIP/RTP — For PBX calls, RTP or SIP routing issue

Solutions

1. Check microphone permissions — Browser and OS must allow microphone access
2. Unmute — Ensure you're not muted in the call UI or system settings
3. Select correct microphone — In call settings or browser permissions, choose the right input device
4. Test microphone — Use the OS sound settings or an online mic test to confirm the mic works
5. Try a different browser — Chrome, Firefox, and Edge are recommended
6. Check firewall — Corporate or home firewall may block incoming RTP; allow CrystalQore and WebRTC
7. For PBX calls — Verify SIP registration, RTP ports, and codec settings in FusionPBX

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No Audio

Causes

• Permissions — Microphone (and sometimes speaker) not allowed
• Wrong device — Output set to wrong speaker or disabled device
• Muted — System or app mute
• Codec mismatch — For PBX/SIP, no common codec negotiated
• Connection failure — WebRTC or SIP failed to establish media

Solutions

1. Grant permissions — Allow microphone (and camera if video) when the browser prompts
2. Check device selection — Ensure the correct speaker/headset is selected
3. Unmute — Check system volume and app mute controls
4. Verify devices — In OS sound settings, confirm mic and speaker work
5. Test with different browser — Chrome and Firefox are best supported
6. For PBX calls — Check SIP registration status; verify codec configuration (e.g., G.711) in FusionPBX
7. Refresh and retry — Reload the page and attempt the call again

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PBX (SIP) Specific

For calls routed through FusionPBX or SIP:

SIP Registration

• Check registration — Ensure your extension is registered in FusionPBX
• Credentials — Verify username and password are correct
• Network — SIP typically uses UDP 5060; ensure it's not blocked

Codec Settings

• G.711 — Most compatible; ensure it's enabled if you have issues
• Transcoding — PBX may need to transcode; check PBX capacity and codec support
• DTMF — If dial pad doesn't work, check DTMF mode (RFC2833, SIP INFO, etc.)

RTP

• Port range — RTP needs a port range open (e.g., 16384–32768)
• NAT — If behind NAT, ensure proper STUN/RTP handling or use a SIP provider with NAT support

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Quick Checklist

Related Documentation