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__

Use set literals instead of set()

Fix passing formatting arguments to string

Remove mismatched HTML closing tag

Log errors when setting up streams in new_from_sdp

Fix potential access to uninitialized variable

Fixed statements with no effect

Make the Jingle MediaStreamRegistry analogous to the SIP one

Catch exceptions when accepting incoming subscriptions

Fix access to MediaStreamRegistry after SDK upgrade

Set version to 3.2.0-dev

Updated version in Debian changelog

Disable i/o buffering when running with systemd

Rename variable for clarity

Un-vendor Klein

Renamed variable

Update copyright years

webrtcgateway: enable WebSocket pinging

Fix method name

Fix sending XMPP messages after API changes

Update todo

Update references, some of the drafts are now RFCs

Fix overriding local_uri for MSRP streams

Fixup leftover old streams API usage

webrtcgateway: skip 'detached' event

Fix per-room pstn_access_numbers setting

Raised version and updated changelog

webrtc: add display name support for incoming and missed calls

Fix webrtc_gateway_url setting

Updated Janus version dependency

Adapt to changes in SIP SIMPLE SDK

Raised python-sipsimple dependency

Updated Janus config to match new version

Added webrtc_gateway_url settings for conference rooms

Added 'missed_session' event