Hi, I can’t seem to be able to figure out why I can’t send RTP to the other side (UAC) of the switch (One way audio?) . I have FreeSWITCH Version 1.10.4-dev git 00113c4 with vanilla config (for testing) on Public IP address and the other switch is also on Public IP on same subnet. I receive call from SippySoft (UAC) and playing delay_echo application. The call flow is SIP UA (NATted) -> SippySwitch (Public IP) - FreSWITCH (Public IP) FreeSWITCH after answering the call says 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [PCMU:0:8000:40:64000:1]/[PCMA:8:8000:20:64000:1] 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [PCMU:0:8000:40:64000:1]/[PCMA:8:8000:20:64000:1] 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [PCMU:0:8000:40:64000:1]/[PCMU:0:8000:20:64000:1] 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5630 Audio Codec Compare [PCMU:0:8000:20:64000:1] is saved as a near-match 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5701 Substituting codec PCMU@40i@8000h@1c 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:3839 Set Codec sofia/internal/04232152273@43.225.99.130 PCMU/8000 40 ms 320 samples 64000 bits 1 channels *2020-07-09 21:48:37.902333 [DEBUG] switch_core_codec.c:111 sofia/internal/04232152273@43.225.99.130 <04232152273@43.225.99.130> Original read codec set to PCMU:0* and then starts delay_echo() app. Isn’t Original read codec set to PCMU:0 is a bad thing? Perhaps the reason not able to send RTP back? On the other hand, I installed Asterisk 13.34.0, on the same machine, just to prove the point if there is network issue but things work just fine with Asterisk default config. Seem like there is either something not configured (default) in FreeSWITCH and I can’t seem to be able to find either. SDP in Asterisk and FreeSWITCH both seems to be the same. FreeSWITCH SIP Trace is here https://pastebin.freeswitch.org/view/0925119c and console log is here https://pastebin.freeswitch.org/view/69b5f68c Asterisk SIP Trace is here https://pastebin.freeswitch.org/view/f38abc97 Appreciate some input to figure out this problem. Thanks, Naseer