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Is it possible to use freeswitch (as a sip registrar and proxy) and sipjs as a user agent that receives calls (RTP)? What I am trying currently is to start with the freeswitch address, but seems impos…
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For example using: https://sipjs.com/
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Hi,
Looks like an error popped up in your latest release (0.0.6) with SIPProviderContext properties. When you build the project all the properties of `useSessionCall` object in `SIPProviderContext.…
enesr updated
13 hours ago
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I've noticed the SaraP is generating double BYE when established session is disconnected using "Hangup" button on the web. The SIP.js demo client does not have this issue, so it suggests some incorrec…
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In version 0.2.1 you integrated "popup card". Since this version, I cannot see any content in the popup. When I start a call, the transparent grey overlay is shown, but no other content, even no butto…
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# YOU MUST read first!
Please use [Community Forum](https://github.com/cordova-rtc/cordova-plugin-iosrtc/discussions) for general technical discussions and questions.
- [x] **I have used Google wi…
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First and foremost, this project is awesome. Thank you for developing and maintaining it.
I installed the Asterisk add-on, configured the SSL (whoof), and installed the card addon alongside the integ…
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Hi, I am using the WebRTC sipjs client, version is 0.7.7.
When I made a call, the network changed, e.g. switching to another Wifi. The call session was still online, but there was no voice data sen…
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Person A maked a call to Person B and created active session. If Persone A terminate call session, everything works correctly.
But if Person B terminate call session, dosnt fired event "terminated" o…
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Dear Mr. Khan,
I have implemented MultiRTC project in my system now, I want to integrate Freeswitch as a media server to procure this I have used sipJS library and i can successfully register user …