[Freeswitch-users] Oneway Audio, can’t send RTP back to UAC

[Muhammad Naseer Bhatti]

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 at 40i@8000h at 1c
2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:3839 Set Codec
sofia/internal/04232152273 at 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 at 43.225.99.130 <04232152273 at 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
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