Increase version

Migrate PGP keys to SQL storage

Migrate contacts to sqlite storage

Increase app store versions

Messaging fixes

Conversations replication fixes and improvements

Added log dir server settings

Fixed loading mime types in a sandboxed environment

Fixed saving shared screens in conference

Fixed sending conference info notifications

Fixed handling screen sharing in conference

Fixed dispatching messages in conference app

Added trace_notification server setting

Update diagram

Update diagram

Discard observer

Fixed settings conference room file transfer directory

Set relative ./var path if started from local directory

Log SIP session failure_reson

Log spool directory at start

Migrate to Python3

Added note about outbound proxy support for SIP messages in webrtc-gw app

Fixed encoding SDP attributes in xmpp gateway

Log available codecs at start

Increase python version to 5.8.0

Update WebRTC architecture diagram

Update changelog

Added initial changelog for version 5.8.0

Fixed encoding UTF-8 in xmpp gateway messaging

Fixed encoding UTF-8 in xmpp gateway messaging

Fixed decoding UTF-8 in xmpp gateway messaging

Add description to exception raised by broken Jingle payload

Improve xmppgateways logging

Fixed handling xmppgateway is_composing events

Fix accumulating duplicate Jingle candidates

Handle RTCP in Jingle media sdp

Improve XMPP/SIP Message interoperability

Less logs

Fixed tls settings for xmmp application

Added TLS support for XMPP gateway

Added subject to conference info description

Added XMPP Muc Subject stanza

Initialize PJSIP engine with a fixed name server

Update WebRTC readme

Allow janus decline code to be defined in configuration

Increase version to 5.7.0

Reorder changelog entries

Update changelog for version 5.6.0

WebRTC conference log changes

Log remote participant for media type change in videoroom conference

Improved webrtc video room logging

Fixed log line

Improved logging of push notifications

Log when chat session starts

Fixed rejecting SIP methods in webrtc app

Move incoming sip request to webrtc app logline to debug

Improved logging

Moved bitrate logs to debug

Use push notification token for VoIP

Mention how to enable debug mode

Move poor connectivity logs to debug

Increase dependency version

Update todo

Compact welcome text message in conference rooms

Change for chat room welcome message

Update README with WebRTC multi-party conference features

Update TODO

Spellcheck

Improve README

Updated README

Added documentation about messages sync with SIP devices

Update authors

Updated authors and sponsors

[webrtcgateway] Capture exception if SDP is empty

[webrtcgateway] Refactor some log lines

[webrtcgateway] Remove duplicate storage of mobile token

[webrtcgateway] Delegate account removal function to the client

[webrtcgateway] Save janus handle id in each video room

[webrtcgateway] Store end-point address for each videoroom participant

[webrtcgateway] Save session start time

[webrtcgateway] Save media type for each session

[webrtcgateway] Remove redundant storage for usage data

[webrtcgateway] Don't call remove account twice

[webrtcgateway] Always delete janus account in the same thread

[webrtcgateway] Fixed DNS lookup for outbound sessions

[webrtcgateway] Purge accounts from memory when connection closes

[webrtcgateway] Log chosen SIP route

[webrtcgateway] Keep track of accounts and devices usage

[webrtcgateway] Remove account if registration failed

[webrtcgateway] Purge unused accounts from memory

[webrtcgateway] Store mobile token in each account

[webrtcgateway] Keep track per room of each video publisher and subscriber feeds

[webrtcgateway] Added sound to mobile wakeup notification

[webrtcgateway] Capture exception

Update TODO

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