Andreas Haardt

This is a really great feature. Can I use jsSIP with Mediasoup to send an RTP stream to a SIP server (for example to mixing audio)?

Yes, not out of the box - Maybe the wrong place, but we have developed a SFU based on Mediasoup and wanted to use jsSIP to send the RTP streams...

I have been working on this ticket. I removed the "RTCPeerConnection" and the "...getUserMedia" from the RTCSession file. My current API for Browser/Node-MediaConnectionInterface: - signalingState() (ice state?, getter) - setup(pcConfig,...

Exactly, I have replaced the 'connection'. Now the 'Interface / MediaConnection' is used everywhere. I have created a feature branch: https://github.com/Haardt/JsSIP/tree/feature/427-Decouple_signaling_and_WebRTC_stuff Maybe you can look into it - i'm on...

@jmillan i made some progress: - [x] Please keep the name _connection - [x] We are not going to do getUserMedia anymore - [x] UA does not need an instance...

@jmillan i have create a draft pull request from my branch to my master.

ping - some progress in the branch.

I have implemented your comments. What do we do with 'getSender'? This is needed for mute. I have also implemented two test projects: - TrySip with the new version -...

Thank you - removed the gradle-git-properties and it works again :)!

And so, the installation guide no longer works either. It simply takes the 'latest' version, right?!