Troubleshooting Call Quality

Audio Is Choppy or Robotic

Choppy or distorted audio is almost always caused by network issues.

For Web Widget Callers

1. Check internet speed. Callers need at least 1 Mbps upload and download. Test at speedtest.net.
2. Switch to a wired connection. Wi-Fi can introduce jitter and packet loss.
3. Close bandwidth-heavy apps. Video streaming, large downloads, and video calls on other tabs compete for bandwidth.
4. Try a different browser. Chrome generally provides the best WebRTC performance.

For SIP/PBX Callers

1. Check your network for packet loss. Run a ping test to sip.aivo.bz. Packet loss above 1% will degrade audio quality.
2. Enable QoS on your router. Prioritize SIP and RTP traffic (DSCP EF for RTP, CS3 for SIP).
3. Check codec settings. G.722 provides the best quality. Fall back to G.711 if G.722 is not supported.
4. Reduce jitter buffer if too high. A jitter buffer over 200ms adds noticeable delay.

Caller Cannot Hear the AI

Web Widget

1. Check volume settings. The caller's device volume and browser tab volume must both be on.
2. Check browser permissions. The browser must have permission to use the speaker/audio output.
3. Try headphones. Some devices route WebRTC audio differently.

SIP/PBX

1. Check for one-way audio. This is the most common SIP issue, usually caused by NAT.
2. Fix NAT issues:

• Set your PBX's external/public IP in its NAT settings.
• Ensure RTP ports (10000-20000) are open and forwarded.
• Disable SIP ALG on your router.

1. Check codecs. If the PBX and AIVO cannot agree on a codec, there will be no audio. Enable G.711 as a fallback.

AI Does Not Understand the Caller

Common Causes

1. Background noise. Loud environments make speech recognition difficult. The caller should move to a quieter location or use a headset.
2. Poor microphone quality. Built-in laptop microphones often produce low-quality audio. An external mic or headset improves recognition.
3. Accent or dialect challenges. The STT engine handles most accents well, but heavy accents may reduce accuracy. Adding common local phrases to your knowledge base can help.
4. Caller speaking too fast or too quietly. The AI works best with clear, moderately-paced speech.

What You Can Do

1. Review call transcripts for misrecognized words.
2. Add common mispronunciations as alternate terms in your knowledge base.
3. If a specific topic is consistently misunderstood, simplify the language in the related knowledge base article.
4. Consider increasing the silence detection timeout so the AI waits longer before responding (Settings > Voice & AI > Advanced).

Calls Dropping Unexpectedly

Web Widget

1. Browser tab closed or minimized. Some browsers throttle background tabs, which can drop WebRTC connections.
2. Network switch. If the caller's device switches from Wi-Fi to cellular (or vice versa), the call may drop.
3. Max duration reached. Check your max call duration setting in Voice & AI > Advanced.

SIP/PBX

1. Registration expiry. If your PBX registration expires mid-call, subsequent calls fail. Set registration expiry to at least 3600 seconds.
2. Firewall timeout. Some firewalls close UDP sessions after 30-60 seconds of no traffic. Enable SIP keep-alives (OPTIONS pings) every 20 seconds.
3. ISP issues. If calls drop at the same time every day, your ISP may be doing maintenance. Check with them.

Long Pauses Before the AI Responds

Possible Causes

1. High STT latency. The speech-to-text engine may be processing a complex utterance. Pauses under 2 seconds are normal.
2. Large knowledge base search. If you have hundreds of articles, the AI may take slightly longer to find the right answer. Well-organized categories help.
3. Ultra voice processing. Ultra voices add about 200ms of latency. If speed is more important than voice quality, switch to a standard voice.
4. Network latency. If your PBX is far from AIVO's servers, consider using a closer region.

Fixes

1. Keep knowledge base articles focused and specific (shorter articles = faster lookup).
2. Use standard voices if latency is a concern.
3. Check your region setting in AIVO Connect and use the nearest one.
4. If pauses exceed 3 seconds consistently, contact support with example call IDs.