- Ya
- 26 Aprili 2026, 23:18 UTC
- Mwandishi
- Kamo
- Ahadi ya
- af6f181
Two-party browser-to-browser calls connected at the SIP layer but had no audio either direction. Voicemail (one-way Asterisk → browser) worked because Asterisk's symmetric-RTP fallback could learn the browser's address from any inbound packet. Two-party bridging needs both sides' RTP to actually reach Asterisk, which means ICE has to converge on a candidate pair the browser can route to. Asterisk had ice_support=yes per-endpoint but `STUN address: 0.0.0.0:0` at the rtp engine level, so its only offered candidate was its host (LAN/pod-network) IP — unreachable from public internet browsers. Configure **************** in rtp.conf so Asterisk can also advertise its srflx (public) candidate. STUN address now resolves to 74.125.250.129:19302 after a fresh module load. Inlined rtp.conf rather than relying on rtp_custom.conf because the rtp engine snapshots its [general] config at module-load time and never re-reads it from include files (the same gotcha that bit the earlier http.conf change). The auto-generated rtp_additional.conf and any future rtp_custom.conf are still re-included after the inline keys for forward compatibility.