Struggling with WebRTC after installing Lets Encrypt certificate

[moe]

Dear All,

I have successfully implemented certificate on FusionPBX and I can see the WebRTC module in the Applications menu but when I try to make a call.nothing happens! I can see the camera turns on but no window, no pop no nothing is happening.

I can't tell where to start and what to look for, The logs show nothing about WebRTC .

Any help would be much appreciate.
Thank you

 

[Adrian Fretwell]

Have you configured your SIP profile with a wss binding? Here is a screen shot of the relevant part of my Internal SIP profile:


Also make sure you have a valid certificate file at the location pointed to by tls-cert-dir.

It's a while since I set this up, but I seem to remember that the fullchain.pem and the privkey needed to be combined into one pem file. This may have changed in later versions of FreeSwitch so please check for yourself. For completeness, the part of my certificate script that puts the certificates into FreeSwitch is shown below. I set this up over two years ago, so there may be better ways of doing it now.

    #setup freeswitch tls

    #make sure the freeswitch directory exists
    #mkdir -p /etc/freeswitch/tls

    #make sure the freeswitch certificate directory is empty
    rm /etc/freeswitch/tls/*

    #combine the certs into all.pem
    cat mydomain.uk-fullchain.pem > /etc/freeswitch/tls/all.pem
    cat mydomain.uk-privkey.pem >> /etc/freeswitch/tls/all.pem
    #cat mydomain.uk-chain.pem >> /etc/freeswitch/tls/all.pem

    #copy the certificates
    cp mydomain.uk-cert.pem /etc/freeswitch/tls/cert.pem
    cp mydomain.uk-chain.pem /etc/freeswitch/tls/chain.pem
    cp mydomain.uk-fullchain.pem /etc/freeswitch/tls/fullchain.pem
    cp mydomain.uk-privkey.pem /etc/freeswitch/tls/privkey.pem

    #add symbolic links
    ln -s /etc/freeswitch/tls/all.pem /etc/freeswitch/tls/agent.pem
    ln -s /etc/freeswitch/tls/all.pem /etc/freeswitch/tls/tls.pem
    ln -s /etc/freeswitch/tls/all.pem /etc/freeswitch/tls/wss.pem
    ln -s /etc/freeswitch/tls/all.pem /etc/freeswitch/tls/dtls-srtp.pem

    #set the permissions
    chown -R www-data:www-data /etc/freeswitch/tls

 

[moe]

Hi Adrian,
I have got all of this done already, the wss config are showing and under profile I have it there also enabled.
When I go to SIPjs I can see in registration that user is register and agent is SIP.js/0.7.8 but when trying to make a call nothing happens at all.

I have created wss for both internal and external profiles, and changed of the ext-rtp-ip and ext-sip-ip in both profiles to my public IP but I am still facing the same issue.

I ran out of ideas of where to look anymore. I'd appreciate your help

 

[moe]

When I try to make a call with another extension, It gets stuck at this

 

[DigitalDaz]

Has ANYONE got this thing working? I gave up on it.

 

[moe]

From the looks of it, Adrian has got it working I suppose? I would appreciate if anyone whose got it working would share their profile settings with me so I can configure it the same. Maybe it would work.

 

[Adrian Fretwell]

@moe I may have misunderstood your original post. Were you referring to fusionpbx/fusionpbx-apps/webrtc ? I somehow don't think you were. The webrtc app I mention above is a whole different thing. This connects with mod_verto on port 8082. It is not a SIP endpoint, so I would not expect to see a SIP registration. @DigitalDaz were you referring to this app?

What webrtc app are you using?

We have had working connections with secure web sockets (wss) on port 7443 mainly using sip.js to implement a web browser based soft phone. Here is a good example: https://github.com/Collecttix/ctxSip We never put it in to production though.

 

[ad5ou]

I have had same results. WSS is working well but clients are a pain.
The link Adrian posted uses same SIP.js library as the screen shot but isn’t integrated to Fusion.

I had good results with ctxSip but only in testing. With a few modifications it could be integrated with Fusion login info but I haven’t had time to work on that part

 

[DigitalDaz]

I think it was the sip.js app, I thought that was the one in the image posted above.

 

[moe]

@moe I may have misunderstood your original post. Were you referring to fusionpbx/fusionpbx-apps/webrtc ? I somehow don't think you were. The webrtc app I mention above is a whole different thing. This connects with mod_verto on port 8082. It is not a SIP endpoint, so I would not expect to see a SIP registration. @DigitalDaz were you referring to this app?

What webrtc app are you using?

We have had working connections with secure web sockets (wss) on port 7443 mainly using sip.js to implement a web browser based soft phone. Here is a good example: https://github.com/Collecttix/ctxSip We never put it in to production though.

How is it possible to implement it ? I not an expert on Java at all.

Thank you