webrtcgw: improved logging of incoming connections

Adjusted sample config with latest default values

Changed codec order

Changed sample rate to favor opus

Fixed typo

Simplified code

Added clarifying comment

Fixed chatroom capabilities

Fix settings the recording folder for video conference recordings

Added per video room access control and recording options

Log remote peer address

Change log line

Log account registration failure reason

Log user agent

Improve logging of WebRTC gateway and video conference applications

Mention implemented standards for WebRTC video conferencing

Use proper capitalization

Update readme

Updated version

Updated changelog

webrtc: fix closing WS connections with recent AutoBahn versions

Don't use self-signed certificate for the web server by default

Update docs

Remove deprecated kwarg

Update webrtc readme

webrtc: update builtin HTML description page

Update remaining description texts

webrtc: reorganize API module

Remove unused imports

Updated README

Add log line when WS connection is opened

Set version to 4.0.0

Fix spacing

Remove no longer used code

Fix typo

Add support for TLS certificate chains for the builtin web server

Update readme

Updated TODO

webrtc: add ability to invite participants to a conference

conference: update webrtc_gateway_url example

Update TODO

webrtc: ignore 'textroom' Janus plugin

Update TODO

webrtc: Ignore 'proceeding' event from the SIP plugin

webrtc: add initial support for multi-party video conferencing

Refactor JanusSessionInfo

Separate Janus event handlers per plugin

Updated version

Advertise other means to join a room also when in bonjour mode

Fix advertising WebRTC URLs

Don't advertise XMPP support by default

Refactor echo application

Advertise call-by-uri WebRTC URIs in conference application

Raised version and updated changelog

Update default Janus WebSocket API URL

Raised Janus version dependency

Update Janus configuration

webrtcgateway: add ability to customize User Agent

Remove obsolete sylkrtc test application

Added TODO note

Remove unused variable

Update version to 3.3.0-dev

Add support for setting account display name

webrtcgateway: refactor API message handling

Updated version number

Reorganize models

xmppgateway: fix unicode error when sending MSRP chunks

Forcefully exit if we fail to start TLS

Updated changelog

Join the Engine thread just for 5 seconds

Use new notification to listen for Engine failures

Adapt to transpoert API change in Jingle streams

Added command line option for memory debugging

Rewrite set computation for clarity

Simplified logic for starting server

Move code out of try-except block

Use collections.Counter instead of a custom defaultdict

Update INSTALL

webrtcgateway: enable optional SRTP-SDES for outgoing calls

Avoid creating unnecessary list

Adapted to changes in SIPSIMPLE SDK 3.0.0

Fixed compatibility with AutoBahn >= 0.12

Adapt to API changes in SIPThor

Don't set GnuTLS compression parameters

Reject incoming sessions with a Replaces header

Use set literals

Avoid creating unnecessary lists

Simplified code

Handle parsing errors for is-composing payload

Style

Simplified exception handlers

Read code and reason from notification when posting SIPSessionDidFail

Style

Fix potential use of unitialized variable

Remove unused variable

Style

Compact code

Improved logger buffer construction

Fix passing incorrect argument to function

Initialize instance attribute in __init__