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:
(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!
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
anddrachtio-freeswitch-mrf
like this:Podman creates a virtual network:
and assigns IPs
10.88.0.2
todrachtio-server
and10.88.0.4
tofreeswitch
.I have added
<contact external-ip="172.21.113.226">sip:*:5062;transport=udp,tcp</contact>
todrachtio.conf.xml
.With my softphone connected to my locally running Freeswitch Windows instance, and having started
google_transcribe.js
, I can then callsip: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 tofreeswitch
.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