RTP problem with examples/google_transcribe.js (Windows/Podman)

[afimuc]

I'm having trouble getting google_transcribe.js to work with Podman under Windows. I can hear the prompt, but nothing seems to be transcribed and none of the event listeners are being called. I suspect that the RTP packets are not being correctly received by Freeswitch, but I currently have no idea how to go about debugging this.

I'm starting drachtio-server and drachtio-freeswitch-mrf like this:

podman run --rm --name drachtio-server ^
-v config/drachtio.conf.xml:/etc/drachtio.conf.xml ^
-p 9022:9022/tcp ^
-p 5062:5062/tcp ^
-p 5062:5062/udp ^
drachtio/drachtio-server

podman run --rm --name freeswitch ^
-v config/freeswitch:/usr/local/freeswitch/conf ^
-e GOOGLE_APPLICATION_CREDENTIALS=/usr/local/freeswitch/conf/credentials/google_application_credentials.json ^
-p 8022:8022/tcp ^
-p 5038:5038/tcp ^
-p 5039:5039/tcp ^
-p 20000-21000:20000-21000/udp ^
drachtio/drachtio-freeswitch-mrf ^
freeswitch ^
--sip-port 5038 ^
--tls-port 5039 ^
--rtp-range-start 20000 ^
--rtp-range-end 21000

Podman creates a virtual network:

(inside WSL)> ip route show
default via 172.21.112.1 dev eth0
10.88.0.0/16 dev podman0 proto kernel scope link src 10.88.0.1
172.21.112.0/20 dev eth0 proto kernel scope link src 172.21.113.226

and assigns IPs 10.88.0.2 to drachtio-server and 10.88.0.4 to freeswitch.

I have added <contact external-ip="172.21.113.226">sip:*:5062;transport=udp,tcp</contact> to drachtio.conf.xml.

With my softphone connected to my locally running Freeswitch Windows instance, and having started google_transcribe.js, I can then call sip:1002@172.21.113.226:5062 and hear the audio prompt. However, I don't hear any response, probably because my RTP packets don't make it to freeswitch.

Adding a route like route ADD 10.88.0.4 MASK 255.255.255.255 172.21.112.1 doesn't help.

Being a SIP newbie, I haven't been able to make much of the packet capture yet. Any suggestions about how to debug this would be welcome!

Wireshark_capture.zip