No Audio in Chrome (WebRTC) with Asterisk version 20.16.0 (Works in Electron)

[kamalaryal123]

Environment
JsSIP version: 3.10.0
Browser: Google Chrome (latest version)
Electron: Works correctly with same codebase
Asterisk version:20.16.0
OS: (Linux / Windows / macOS)
Transport: WSS (WebRTC)
Issue description

We are experiencing a one-way audio issue (no incoming audio in browser) when using JsSIP with above mentioned Asterisk server.

Architecture

We have three Asterisk servers connected to different SIP trunks:

Server A
Server B
Server C

The dialer uses WebRTC with JsSIP and runs in:

Web browser (Chrome)
Electron desktop application
Current behavior

Server A → Works perfectly on both Web and Electron.
Server B → Works perfectly on both Web and Electron.
Server C → Electron works with proper two-way audio, but the Web browser version has no audio (cannot hear the remote party).

Additional observations
The same JsSIP dialer codebase is used for all servers.
The same SIP endpoints can successfully register and make calls.
On Server C, the call connects successfully but the browser client cannot hear audio.
Electron client works normally with Server C.
Older Chromium/Chrome versions (~144) worked correctly.
Since the same WebRTC client works with other Asterisk servers, it is difficult to conclude that the issue is caused only by Chrome version changes.
Endpoint configuration used

Basic endpoint configuration:

basic_endpoint
moh_suggest=default
context=from-extensions
inband_progress=no
rtp_timeout=10
message_context=textmessages
allow_subscribe=yes
subscribe_context=subscriptions
dtmf_mode=rfc4733
device_state_busy_at=1
disallow=all
transport=system-udp
allow=alaw,ulaw

WebRTC endpoint configuration:

webrtc_endpoint
disallow=all
transport=wss_transport
allow=ulaw,alaw,g722,opus
rtcp_mux=yes
webrtc=yes
dtls_verify=fingerprint
dtls_setup=actpass
ice_support=yes
use_avpf=yes
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
media_encryption=dtls
direct_media=no
send_rpid=yes
Issue summary
Calls connect successfully
No incoming audio in Chrome browser
Electron works correctly
Same configuration works on other Asterisk servers

We are trying to understand whether this might be related to:

ICE negotiation
DTLS / SRTP behavior changes in newer Chrome versions
Codec negotiation
WebRTC handling differences between Electron and Chrome