Don't know if it's the same problem but I incorporated FreeSWITCH code into a product a couple of years ago (master code was marked as 1.9.x) and whilst the (FreeSWITCH) server worked fine, a client built with that version of the software would only give audio in one direction.

I figured out a fix (to switch_ivr_originate.c) and raised a Jira but didn't provide the fix as I wasn’t sure the change I had made was the best way to resolve the issue. I assumed that someone with a better understanding of the code would be better placed to determine the most appropriate fix.

Interestingly if I used a much older version of the FreeSWITCH software in the client (like that shipping with the FSClient software) the problem did not occur.

Sent from my Windows 10 device

From: Muhammad Naseer Bhatti
Sent: 09 July 2020 18:44
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] Oneway Audio, can’t send RTP back to UAC

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]/[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.

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