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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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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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See: https://sipjs.com/
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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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# 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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We are made our conference according to the page (https://groups.google.com/g/jssip/c/iT0qmLfRZhw). And its working fine based on the issue no 201.
We are joining the voice track, but the calls d…
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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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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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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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projects that include sipjs can no longer be submitted to the app store since there is no 64 bit support.