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libwebphone issues


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Hey guys,

I see that you've moved from using the sipjs stack to using the jssip stack for the libwebphone. I cloned the repo, set up the basic wss and tried to make a call. The call is failing and FS is generating a 488 Not Acceptable Here. After some troubleshooting, i've determined the issue is with the audio candidate list having a private IP address

a=candidate:329485039 1 udp 2122260223 192.168.76.132 61644 typ host generation 0 network-id 7 network-cost 50

Reading through the documentation for 3.3.X, i don't see anywhere to define a STUN server to correct this issue. 

I then went ahead and cloned an older version of the libwebphone that used sipjs 0.7.5. I was able to define and configure a STUN server, and I was able to make and receive calls, but I had one-way audio. My computer never played audio, but my cell phone always had audio from my computer. Using wireshark, I confirmed that FS is sending the audio back to my computer, but it's like the web phone isn't getting the remote audio track or something. Have you guys run into this before? Any suggestions on resolving either issue?

I am testing with the rc3 version of kamailio and kazoo-freeswitch 4.3.-4. Kamailio is enabled for websocket role and TLS role, and i have configured certs on it and everything. 

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On 9/23/2019 at 1:27 PM, safarov said:

Error 488 Not Acceptable Here is related to used codec and encryption.
Not IP used on caller side (private or public).
Please paste INVITE here and codec related variables from "freeswitch.xml" 

The error was that there were no valid candidates in the SDP because they were all local IPs. By applying the candidate ACLs, FreeSWITCH handles the SDP differently and is able to determine the public address to send the SDP. 

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