<html><head><meta http-equiv="Content-Type" content="text/html charset=utf-8"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class="">Hello list,<div class="">I thought I had solved this issue by reducing my codec list to a minimum, but it still persists, unfortunately. This was to reduce the TLS packet size.</div><div class="">Anytime I enable SRTP on my phones, outgoing calls will randomly fail. The problem goes away when I disable SRTP. I only work over TLS.</div><div class="">Incoming calls work reliably with or without SRTP.</div><div class=""><br class=""></div><div class="">How do you suggest debugging this?</div><div class="">I tried setting up a fresh instance of FS but the issue persists.</div><div class="">Now. It should be noted that calls fail sensibly more often when I have more than one account registered on the same server with the same device.</div><div class=""><br class=""></div><div class="">Any suggestion is welcome. Have you experienced this? </div><div class=""><br class=""></div><div class="">I’m running <span style="font-family: Menlo; font-size: 11px;" class="">FreeSWITCH Version 1.6.5 on Debian. My phone in this case is a Yealink SIP-T46G running firmware </span><font face="Menlo" class=""><span style="font-size: 11px;" class="">28.80.0.95.</span></font></div><div class=""><font face="Menlo" class=""><span style="font-size: 11px;" class=""><br class=""></span></font></div><div class=""><font face="Menlo" class=""><span style="font-size: 11px;" class="">E</span></font></div></body></html>