Hi All,
I wanted to share my webphone that I'm working on using a modified version of ctxSip. It includes an updated version (0.15.6) of Sipjs which required me to make some adjustments to webphone/phone/scripts/SIP.js/sip.js in order to get hold and mute working correctly. I had issues with one-way audio using the older sipjs version 0.7.8.
Before you attempt to install this make sure you have webrtc working on your system!!
Use this link to test it out: https://collecttix.github.io/ctxSip/
Brief instructions...
1. Make sure you have webrtc up and working. See above.
2. Is webrtc working? Read through this thread if you are stuck. https://www.pbxforums.com/threads/tls-and-webrtc-enablement.146/
3. Download the webphone files and stick them in your /var/www/fusionpbx/apps directory
https://github.com/konradSC/Fusionpbx-Public/tree/master/Apps/webphone
2. Change owner
chown -R www-data:www-data /var/www/fusionpbx/
3. Upgrade Permissions and Menu
A few things...
1. I'm not a JS programmer, so a lot of this was hacked together. I'm open to suggestions on code changes.
2. The wss port is hard coded to 7443. You can adjust that in webphone/phone/index.php. Update the hardcoded variable or use a session variable.
3. There seems to be a two or three second delay in call setup. It could be the ICE or STUN setup. I haven't spent too much time troubleshooting it yet. Let me know if you have a fix.
4. I'm releasing this so the community can test it out. If we get a solid version of this webphone then I'll submit it to the Fusion project.
Thanks,
Konrad
I wanted to share my webphone that I'm working on using a modified version of ctxSip. It includes an updated version (0.15.6) of Sipjs which required me to make some adjustments to webphone/phone/scripts/SIP.js/sip.js in order to get hold and mute working correctly. I had issues with one-way audio using the older sipjs version 0.7.8.
Before you attempt to install this make sure you have webrtc working on your system!!
Use this link to test it out: https://collecttix.github.io/ctxSip/
Brief instructions...
1. Make sure you have webrtc up and working. See above.
2. Is webrtc working? Read through this thread if you are stuck. https://www.pbxforums.com/threads/tls-and-webrtc-enablement.146/
3. Download the webphone files and stick them in your /var/www/fusionpbx/apps directory
https://github.com/konradSC/Fusionpbx-Public/tree/master/Apps/webphone
2. Change owner
chown -R www-data:www-data /var/www/fusionpbx/
3. Upgrade Permissions and Menu
A few things...
1. I'm not a JS programmer, so a lot of this was hacked together. I'm open to suggestions on code changes.
2. The wss port is hard coded to 7443. You can adjust that in webphone/phone/index.php. Update the hardcoded variable or use a session variable.
3. There seems to be a two or three second delay in call setup. It could be the ICE or STUN setup. I haven't spent too much time troubleshooting it yet. Let me know if you have a fix.
4. I'm releasing this so the community can test it out. If we get a solid version of this webphone then I'll submit it to the Fusion project.
Thanks,
Konrad