Issues with voice quality during webRTC calls

Asterisk Asterisk WebRTC
1

Hello everyone,

I’m reaching out for help as I’ve run out of ideas and my brain is about to boil.

Context:
We have an application that includes a WebRTC softphone, connected to Asterisk 18. Everything works — calls go through, audio streams in both directions — but one of our clients is experiencing an issue with voice quality during the first 15–20 seconds of the conversation.

I’ve listened to the recordings myself:

  • The agent’s voice is affected — it sounds distant, muffled, or like they’re chewing while speaking.

  • The issue seems to happen at random times but is most concentrated in the first 20 seconds.

  • It does not sound like classic packet loss — more like microphone-related or encoding issues.

  • After clients complain mid-call, it’s like the agent gets “closer to the mic” — audio suddenly becomes crisp and clear. But this is only based on perception.

Important notes:

  • The problem is not with the SIP provider, since the client’s side hears the agent poorly, and the recording also reflects that.

  • Jitter is within normal limits — under 20ms.

  • Agents are using professional office headsets.

  • Almost all agent workstations are running Ubuntu 24.04.

What we’ve already tried:

  1. Switched codecs from opus to alaw/ulaw.

  2. Migrated the client’s server between 3 different hosting providers in 2 countries — same issue everywhere.

  3. Enabled STUN in rtp.conf using:
    stunaddr=stun.l.google.com:19302

  4. Added STUN servers to the WebRTC softphone config:
    iceServers: [{ urls: "stun:stun.l.google.com:19302" }]

  5. Tested the softphone from our side (3 locations, 2 countries, all on Windows) — no issue.

  6. Client also tested on Windows — issue persisted, according to them.

Additional information:

  • The client says they also tested their previous SIP/WebRTC setup and this issue does not happen there.

  • They are using a VPN, but it’s split-tunneled — it does not affect external traffic, and disabling it has no impact on call quality.

At this point I’m running out of ideas. If anyone has encountered similar issues, or can suggest what else to look into (hardware, WebRTC configs, ICE candidates, jitter buffer settings, browser quirks, etc.) — I’d really appreciate it!

Thanks in advance for any help!

2

Is this the same thing as:

Additionally, if using AI (even to formulate your forum post) - disclose it.

3

Ok, thank you. English is not my native language so yes, i`ve used AI to translate original text into English language.

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