The integration of WebRTC with Asterisk 22.8 on a VPS involves complex interactions between SIP signaling over WSS, RTP media transport, and ICE framework connectivity. Developers often face silent failures where calls are established but audio is absent due to misconfigured ICE candidates, NAT traversal issues, or improper handling of the JsSIP library events.
Initial connection failures were linked to the browser collecting only private IP candidates. Implementing a custom STUN/TURN server using Coturn on the same VPS as Asterisk failed due to TCP TURN overhead and internal browser timeouts. Furthermore, failing to trigger the event.ready() method in JsSIP caused a 40-second delay in SDP exchange, leading to call timeouts.
Network configuration errors, such as incorrect mapping of local IP addresses to public IPs in Asterisk's rtp.conf, prevented media packets from reaching the destination. Additionally, forcing a relay-only policy created a loop where the TURN server refused to proxy traffic between two allocations on its own IP address, resulting in 403 Forbidden errors.
The final stable configuration relies on a combination of STUN for public IP discovery and TURN for relay when direct connections fail. Asterisk correctly advertises its public IP through the ice_host_candidates section in rtp.conf. This setup ensures successful Opus codec transmission with low latency and minimal packet loss, resolving the underlying connectivity issues.
