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Fix 27s call ringing delay: TURN credential mismatch and STUN DNS TTL=0

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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
This commit is contained in:
Claude
2026-02-24 22:07:33 +00:00
parent cbccaa4218
commit de46202ab4
4 changed files with 36 additions and 16 deletions
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README.md
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@@ -663,6 +663,22 @@ For **any SIP app**:
- FQDN mode enables ICE (Interactive Connectivity Establishment) which handles network changes better
- Alternatively, keep your phone on one network type (WiFi or mobile data) during calls
### "No audio when Call Integration is enabled" (Android)
Some Android SIP apps (Sipnetic, Olinuxino, etc.) offer a "Call Integration" feature that routes calls through Android's native dialer (ConnectionService API). When enabled:
- Incoming calls appear in the native phone app
- The headset media button works for answering
- **But the caller may get no audio**
**Why this happens:**
Android's ConnectionService routes audio through the native telephony audio path, which doesn't always relay VoIP RTP audio correctly. This is a client-side Android issue, not an Asterisk problem. It affects both stock Android and LineageOS.
**Fix:**
- Disable "Call Integration" in your SIP app's settings
- Use the SIP app's own answer button instead of the headset media button
- If you need headset button support, check if your SIP app supports media button handling without Call Integration
### "My IP changed and FQDN stopped working"
See [Dynamic IP Handling](#dynamic-ip-handling) section. You need to set up DDNS.