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
- Add local_net entries to PJSIP transports in LAN/VPN mode when
VLAN/VPN subnets are configured, so Asterisk recognizes VPN traffic
as local and mobile devices no longer appear offline on VPN
- Auto-detect VPN interfaces (Tailscale, WireGuard, OpenVPN) and
suggest their subnets in the VLAN/VPN configuration menu
- Add rtp_keepalive=15, rtp_timeout=120, and rtp_timeout_hold=120
to mobile device endpoints to maintain NAT mappings during
WiFi-to-cellular transitions (fixes one-way audio)
- Reduce qualify_frequency from 60s to 30s for faster re-registration
detection when mobile devices change networks
- Update README with VPN troubleshooting and one-way audio guidance
including client-side fixes for Sipnetic and Linphone
https://claude.ai/code/session_01Vm6NLaQuzM4VosAotqS1q8
Claude