Full changelog

Version 4.1.0

January 20th, 2017

  • Mention implemented standards for WebRTC video conferencing
  • Improve logging of WebRTC gateway and video conference applications
  • Added per video room access control and recording options
  • Fix settings the recording folder for video conference recordings
  • Fixed chatroom capabilities
  • Changed codec order
  • Changed sample rate to favor opus
  • Adjusted sample config with latest default values
  • webrtcgw: improved logging of incoming connections
  • webrtcgw: initial implementation of push notifications framework
  • webrtcgw: fix sample configuration file
  • webrtcgw: reorganized package
  • webrtcgw: fix for AutoBahn API change
  • webrtcgw: simplify ICE state flags
  • webrtcgw: uncomment log lines
  • webrtcgw: set content_available to True for FCM notifications
  • Capture validation errors when building requests
  • The new_token field is not required for the account-devicetoken request
  • Added extra logging to help debug device token handling
  • Increased debian compatibility level to 9
  • Increased debian standards version
  • Updated debian package maintainer
  • Added debian dependency on lsb-base
  • Adjusted debian package's descriptions
  • Updated boring file

Version 4.0.0

October 12th, 2016

  • webrtc: add multi-party video conferencing capability
  • webrtc: separate Janus event handlers per plugin
  • webrtc: refactor JanusSessionInfo
  • webrtc: reorganize API module
  • webrtc: update builtin HTML description page
  • core: add support for TLS certificate chains for the builtin web server
  • echo: refactor application
  • conference: advertise call-by-uri WebRTC URIs
  • conference: don't advertise XMPP support by default
  • conference: advertise other means to join a room also when in bonjour mode

Version 3.3.0

June 14th, 2016

  • webrtcgateway: refactor API message handling
  • webrtcgateway: reorganize models
  • webrtcgateway: add support for setting account display name
  • webrtcgateway: remove obsolete sylkrtc test application
  • webrtcgateway: add ability to customize User Agent
  • Raised Janus version dependency
  • Update Janus configuration

Version 3.2.0

March 8th, 2016

  • Fix per-room pstn_access_numbers setting
  • Fixup leftover old streams API usage
  • webrtcgateway: skip 'detached' event
  • Update references, some of the drafts are now RFCs
  • Fix overriding local_uri for MSRP streams
  • Fix sending XMPP messages after API changes
  • Fix method name
  • webrtcgateway: enable WebSocket pinging
  • Un-vendor Klein
  • Disable i/o buffering when running with systemd
  • Fix access to MediaStreamRegistry after SDK upgrade
  • Make the Jingle MediaStreamRegistry analogous to the SIP one
  • Catch exceptions when accepting incoming subscriptions
  • Don't set GnuTLS compression parameters
  • Adapt to API changes in SIPThor
  • Several code style improvements
  • Log errors when setting up streams in new_from_sdp
  • Remove mismatched HTML closing tag
  • Handle parsing errors for is-composing payload
  • Reject incoming sessions with a Replaces header
  • webrtcgateway: enable optional SRTP-SDES for outgoing calls
  • Update INSTALL
  • Simplified logic for starting server
  • Added command line option for memory debugging
  • Adapt to transpoert API change in Jingle streams
  • Use new notification to listen for Engine failures
  • Forcefully exit if we fail to start TLS
  • Join the Engine thread just for 5 seconds
  • xmppgateway: fix unicode error when sending MSRP chunks

Version 3.1.0

December 4th, 2015

  • Fixed default web port in sample config file
  • Terminate connections if backend goes down
  • webrtc: fix navbar rounded corners in test app
  • webrtc: show remote party in test app
  • Improve error messages for API call errors
  • Exit with a a non-zero exit code if engine failed
  • Update README with WebRTC related information
  • Added 'missed_session' event
  • Added webrtc_gateway_url settings for conference rooms
  • Adapt to changes in SIP SIMPLE SDK
  • Raised python-sipsimple dependency
  • Updated Janus config to match new version
  • Raised Janus version dependency
  • webrtc: add display name support for incoming and missed calls

Version 3.0.1

September 4th, 2015

  • webrtc: mute local video in test application
  • Adjust web port in configuration example
  • Fix installing default certificates also in /usr/share/

Version 3.0.0

August 28th, 2015

  • Added WebRTC gateway application
  • Switch to using listenSSL
  • Make main web server logging less verbose
  • Fix initializing Path datatype
  • Rework how services are published in SIPThor
  • Install all sample configuration files
  • xmppgateway: make factories not noisy
  • Add systemd unit file
  • Improved Debian package creation
  • Added build dependency on dh-python

Version 2.9.1

April 29th, 2015

  • Add spool_dir setting
  • Simplify SylkServer's stream subclasses
  • Stop the session manager first when shutting down
  • Adapt to API changes in MSRPlib
  • Refactor file transfers to match API changes in SIP SIMPLE SDK

Version 2.9.0

March 17th, 2015

  • Added ZRTP and opportunistic encryption support
  • Adapt to changes in SIP SIMPLE SDK
  • Add python-lxml as a direct dependency
  • * Relax XMPP - SIP URI marching
  • Accept any content type in echo application
  • Support inlined images in the conference application
  • Add setting for toggling presence activity logging (xmppgateway)
  • Refactored path handling and TLS certificate location
  • Simplify default paths for resources in /var
  • Add ability to skip the isfocus parameter when publishing a Bonjour
  • service
  • Publish echo application on Bonjour if enabled
  • Publish playback application on Bonjour if enabled
  • Change default directory for conference file transfers
  • Tag all messages sent by the room as status messages
  • Publish every Bonjour service with a different id

Version 2.8.0

December 5th, 2014

  • Add a custom Session class
  • Added setting for toggling ICE support
  • Add advertised_ip setting
  • Use the specified IP address both for signaling and media
  • Enhance playback application
  • Adapt to latest SDK API changes
  • Don't advertise the default conference on Bonjour if it's not the default
  • application
  • Add ability to find applications by name to ApplicationRegistry
  • Log default application on start
  • Refactor managing the single account SylkServer currently uses
  • Refactored WelcomeHandler in conference application
  • Use the specified IP address both for signaling and media
  • Allow user-part only matching on playback application
  • Don't manually create the Contact header when not needed
  • Fix JingleSession and adapt audio streams to API changes
  • Strip HTML in IRCconference application
  • Fix handling XMPP stanzas sent to a bare JID when the session was bound

Version 2.7.2

August 12th, 2014

  • Fix setting local IP address
  • Null doesn't need to be instantiated

Version 2.7.1

July 18th, 2014

  • Fix variable name

Version 2.7.0

July 9th, 2014

  • Added setting to set the hostname for conference room screen sharing URL
  • Fix race condition when initializing TLS transport
  • Fixed streams API usage after changes in SIPSIMPLE SDK
  • Fix handling cancelled proposals
  • Added display_name attribute to conference rooms
  • Simplify loading room configuration
  • Cleanup old room files on startup
  • Added logrotate file
  • Raised python-sipsimple version dependency

Version 2.6.2

June 12th, 2014

  • Fixed resource leak in playback application
  • Refactored welcome prompt playback for ircconference application

Version 2.6.1

May 21st, 2014

  • Adjust Session to changes in SIP SIMPLE SDK
  • Adapted server startup to changes in SIP SIMPLE SDK
  • Send REPORT chunks automatically for keep-alive chunks
  • Log SIP SIMPLE SDK version
  • Raised python-sipsimple version dependency

Version 2.6.0

February 19th, 2014

  • Fixed issues when shutting down the Engine
  • Fixed generating is-composing payload when refresh is not set
  • Accept multiple PSTN numbers for a given conference room
  • Use better API for building is-composing payload
  • Avoid unnecessary processing when dealing with CPIMIdentity objects
  • Simplified history storage in conference app
  • Simplified code for handling proposal failures
  • Simplified code for starting/stopping SylkServer
  • Renamed incoming_sip_message to incoming_message
  • Use the new NetworkConditionsDidChange notification
  • Bumped Debian Standards-Version
  • Raised python-sipsimple version dependency

Version 2.5.1

December 16th, 2013

  • Adapted to API changes in SIP SIMPLE SDK
  • Added option to dump core in case of a crash
  • Fixed dispatching messages when in bonjour mode
  • Limit session in echo application to 10 minutes
  • Skip broadcasting OTR messages
  • Reworked server stop mechanism
  • Removed obsolete sound files and fixed co_there_is prompt
  • Fixed removing observer if notification is processed too late

Version 2.5.0

August 9th, 2013

  • Adapted to changes in latest SIP SIMPLE SDK
  • Added playback application
  • Enabled Opus codec by default
  • Added setting for sample rate, defaults to 32 kHz
  • Advertise PSTN and XMPP access in conference rooms
  • Replaced prompts with higher quality ones
  • Fixed initializing PJSIP's internal resolver
  • Always disable echo canceller
  • Improved logging
  • Ignore audio device change notifications
  • Removed dependency on python-backports

Version 2.4.0

March 22nd, 2013

  • Added VoIP translation for SIP/XMPP gateway (Jingle)
  • Added Presence to Bonjour conference rooms (XEP-0174)
  • Added support for XMPP software version (XEP-0092)
  • Added support for XMPP ping (XEP-0199)
  • Reply with service-unavailable to unsupported XMPP IQ stanzas
  • Improved XMPP service discovery support (XEP-0115)
  • Fixed a race condition related to SIP subscriptions
  • Improved description of XMPP related settings

Version 2.3.0

January 30th, 2013

  • Added SIP/XMPP gateway ability to invite participants to a multiparty chat
  • Added RTP audio and MSRP chat 'echo' application
  • Added support for XEP-0030 (service discovery)
  • Added ability to load extra applications from an external directory
  • Added timestamp to generated PIDF documents
  • Simplified mechanism required for adding new applications
  • Refactored per-application logger
  • Improved logging in XMPP gateway and conference applications
  • Removed extended-away state handling as it no longer exists in the SDK
  • Made several improvements to XMPP stanza parsing
  • Fixed detecting MSRP Nickname collision
  • Fixed handling presence stanzas without a resource part in the from
  • Fixed translating resource IDs for presence
  • Fixed leaking session objects if session fails while joining a conference
  • Fixed mapping room URI for received REFER requests

Version 2.2.1

November 9th, 2012

  • Fixed stream creation after API changes in SDK
  • Fixed accessing session objects after API changes in SDK
  • Renamed ServerSession to Session

Version 2.1.1

October 9th, 2012

  • Fixed file transfers when using Bonjour mode
  • Fixed normalizing IPAddress datatype
  • Disables private messages support when using Bonjour mode

Version 2.1.0

September 17th, 2012

  • Added ability to disable applications
  • Added ability to configure the directory for resource files
  • Added ability to listen on all interfaces
  • Refactored Bonjour support
  • Fixed starting music on hold playback
  • Fixed setting extended status for XMPP dnd state
  • Fixed API calls due to changes in SIP SIMPLE SDK
  • Delay conference database initialization until application is started
  • Don't encode and quote DeviceInfo description

Version 2.0.0

September 6th, 2012

  • Added XMPP gateway application
  • Added Bonjour support
  • Added support for MSRP NICKNAME
  • Added ability to map applications by RURI, domain or username
  • Added ability to select desired application with X-Sylk-App header
  • Added ApplicationLogger, in order to prefix each application's log lines
  • Added start/stop methods to applications
  • Added ability to specify more attributes when sending MSRP messages
  • Allow applications to handle the 'presence' event on incoming
  • subscriptions
  • Modified ChatStream to send MSRP REPORT chunks manually
  • Use received reason when notifyig about REFER request progress

Version 1.3.0

December 22nd, 2011

  • Added multiparty comference screen sharing capability
  • by accepting jpeg images over an established MSRP chat stream
  • Added web-server to serve shared screens
  • Made configuration file optional by using defaults settings
  • Initialize applications after loading them
  • Fixed parsing Refer-To URI
  • Pass-through additional headers when dispatching chat
  • messages
  • Adjusted to the latest changes in XML payloads from
  • sipsimple
  • Reject incoming call transfer requests to conference
  • rooms

Version 1.2.3

September 20th, 2011

  • Adapted to API changes in SIPSIMPLE SDK

Version 1.2.2

July 20th, 2011

  • Enabled TLS by default
  • Fixed regression when sending private messages
  • Fixed renaming file when it already exists
  • Listen by default on port 5061 for SIP TLS transport
  • Fixed exception when proposal is rejected but no timer was added
  • Adapted to accounts handling changes in the middleware

Version 1.1.0

March 18th, 2011

  • Added incoming REFER support
  • Added outgoing INVITE support
  • Added SIP outbound proxy support
  • Added Trusted Peers based on source IP address
  • Added Access Control Lists support to conference application
  • Added basic multi-application support
  • Added IRC conference application
  • Added SIPThor integration
  • Fixed initialization of TLS settings
  • Made session connect method receive the contact header
  • Catch exception if outgoing NOTIFY could not be sent
  • Fixed exception when sending private message to a participant without chat
  • Refactored exception handling when sending chat messages
  • Refactored application finding mechanism
  • Reject incoming requests with 404 if application is not found
  • Removed SIP MESSAGE support in conference application

Version 1.0.1

February 17th, 2011

  • Added unicode support
  • Fixed building CPIMIdentity object
  • Use request URI to match rooms instead of the To header

Version 1.0.0

January 27th, 2011

  • Initial release