José Luis Millán
José Luis Millán
Please @ikq, Build and try JsSIP master branch. It should be fixed.
Let us know if master branch is working as expected.
The change was made to [master] branch and there is no new release yet. Have you [built] JsSIP out of master and tried? [master]: https://github.com/versatica/JsSIP/commit/e3fdf420f92f1cddb75d30f528f509b78e3e3a80 [built]: https://github.com/versatica/JsSIP/blob/master/BUILDING.md
Follow the [building](https://github.com/versatica/JsSIP/blob/master/BUILDING.md) instructions and use the resulting .js in /dist folder on your app.
@ikq, Can you please send a full JsSIP log with the latest version showing this issue happening?
In order to allow muting before connection is created we could add a check in `_toggleMuteAudio` and `_toggleMuteVideo`. ```js _toggleMuteAudio(mute) { // RTCPeerConnection is not created yet, skipping. if (!this._connection)...
Hi @ikq, JsSIP code must have the same style in order to make it readable and maintainable. Comments before making a deeper review: - Debug line as the first method...
@ikq Thanks for the cleanup, :+1: We'll review the PR as we can.
@ikq, Cosmetic comment, we use camelCase in API method arguments (not in 100% of the cases, I know, but we will get there). Could you please make such change?
Hi, Super busy lately. Will come back to this when we have some time.