WebRTC audio call fails with "codec collision" (payload_type 111, stereo/non-stereo Opus) — works fine in Chromium-based Edge on the same system

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- [x] I have searched the [issue tracker](https://www.github.com/electron/electron/issues) for a bug report that matches the one I want to file, without success.

### Electron Version
Reproduced on both:
- 39.8.10 (Arch/Manjaro `electron39` package)
- 43.6.0 (Arch/Manjaro `electron43` package, bundled Chromium 150.0.7871.250)

### What operating system(s) are you using?

Other Linux

### Operating System Version

Manjaro Linux, kernel 6.18.49-1-MANJARO, Wayland session (KDE Plasma), PipeWire audio.

### What arch are you using?

x64

### Last Known Working Electron version

Unknown — reproduces identically in two Electron majors 4 versions apart (39 and 43), so likely not a recent regression.

### Does the issue also appear in Chromium / Google Chrome?

I don't know how to test

### Expected Behavior

A WebRTC audio call (Google Chat / Google Meet) negotiates a valid SDP and connects, as it does in a real Chromium-based browser (Microsoft Edge stable) on the exact same machine, same OS session, same audio devices.

### Actual Behavior

The call fails to connect. Chromium's WebRTC stack logs a codec collision and the call layer reports a disconnect:

\`\`\`
[ERROR:third_party/webrtc/pc/sdp_offer_answer.cc:575] A BUNDLE group contains a codec collision between
{payload_type: 111, codec: {mime_type: audio/opus, clock_rate: 48000, num_channels: 2, parameters: {minptime: 10, stereo: 1, useinbandfec: 1}}}
and
{payload_type: 111, codec: {mime_type: audio/opus, clock_rate: 48000, num_channels: 2, parameters: {minptime: 10, useinbandfec: 1}}}.
All codecs must share the same type, encoding name, clock rate and parameters. (INVALID_PARAMETER)
\`\`\`

Two Opus codec entries for the same payload type (111) are generated — one advertising `stereo=1`, one without it — which is an invalid/self-contradictory SDP. The calling web app (Google Chat/Meet) then reports the peer connection as disconnected (`DisconnectedError`, various `EndCause`/`StartupCode` combinations depending on how far initialization got).

### Testcase Gist URL

No minimal repro gist yet — this only reproduces against a live Google Chat/Meet call (a real-world WebRTC service doing its own SDP munging), not a synthetic getUserMedia+RTCPeerConnection testcase. Happy to help build one if a maintainer can point to what would be useful (e.g. a two-peer local repro that forces a `stereo=1`/no-`stereo` transceiver pair on the same m-line).

### Additional Information

- Confirmed NOT reproducible in Microsoft Edge (Chromium-based, not Electron) on the exact same machine, same session, same audio devices, same call — rules out a system/PipeWire/audio-device-specific cause.
- Confirmed independent of which audio input device is selected (tested with two different physical microphones — same collision both times).
- Possibly related to Electron's Linux builds not shipping with the same PipeWire integration as stock Chromium/Chrome/Edge builds (see #29115) — mentioning in case that's a relevant lead for maintainers, though that issue was specifically about screen-share, not general audio/call codec negotiation.
- Audio setup: PipeWire, one active USB microphone tested at a time (Logitech A50 USB headset in `mono-fallback` PipeWire profile, and separately a Creative Live! Cam webcam mic in `analog-stereo` profile) — both trigger the identical collision.

Start by reproducing the live Google Chat or Meet call on Electron 39 or 43 and inspect the WebRTC log at third_party/webrtc/pc/sdp_offer_answer.cc:575. Compare the generated SDP and codec entries with the working Chromium-based Edge case; done means the duplicate payload type 111 Opus collision is absent and the call connects.

Written by the indexing model from the issue text.