Two bugs caused ICE candidate gathering to fail and timeout (~27 seconds)
before falling back to direct media on every call:
1. TURN credential mismatch — when TURN_PASSWORD was empty in .env,
coturn defaulted to "changeme" but the entrypoint auto-generated a
different random password for Asterisk. Every TURN auth attempt failed
("check_stun_auth: user easyasterisk credentials are incorrect").
2. STUN DNS TTL=0 — using the FQDN as stunaddr caused DNS resolution
that returned TTL=0, making Asterisk cancel recurring STUN resolution
entirely. Since coturn runs on the same host (network_mode: host),
rtp.conf now uses 127.0.0.1 which needs no DNS at all.
Also documents the Android Call Integration audio issue (ConnectionService
routes audio through the native telephony path, breaking VoIP RTP).
https://claude.ai/code/session_01KWVtEt9MmZdywcu7WmgchX
Claude
de46202ab4