We’ve come across an audio issue during SIP calls with Asterisk version 20.2.0, utilizing Jssip and TURN as the ICE server. Interestingly, this problem seems to be occurring primarily with specific ISPs. Despite Asterisk being behind NAT and using PJSIP, the connection is established smoothly, but audio transmission is not consistent across all ISPs.
We’re reaching out to seek insights into potential reasons behind this discrepancy. Any guidance or suggestions on how to troubleshoot and resolve this issue would be greatly appreciated. Our goal is to ensure seamless communication for all users, regardless of their ISP.