PCMU:0 is normal. 0 is the payload type for PCMU.
Perhaps you have the firewall enabled on the machines themselves ?  Flush your iptables rules, just to see if it works.



On Thu, Jul 9, 2020 at 8:17 PM Muhammad Naseer Bhatti <nbhatti@gmail.com> wrote:

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 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.

Asterisk SIP Trace is here https://pastebin.freeswitch.org/view/f38abc97 

Appreciate some input to figure out this problem.


Thanks,
Naseer
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