Full changelog

Version 6.3.0

December 22nd, 2023

Updated documentation
Increase minimum version of SDK required
[conference] Handle non base64 images in conference room
[webrtcgateway] Use twisted reactor for Cassandra connection
[webrtcgateway] Catch race condition/exception when sending a message on a disconnected socket
[webrtcgateway] Added file transfers to push notification for unread messages
[webrtcgateway] Reject incoming sessions if they contain no transfer streams
[webrtcgateway] Added support for ringing and proceeding events
[webrtcgateway] Added direction to sync remove message content
[webrtcgateway] Added direction to sync remove messages

Version 6.2.2

July 7th, 2023

[webrtcgateway] Fixed pull MSRP filetransfers

Version 6.2.1

July 6th, 2023

[webrtcgateway] Added API option to get messages from timestamp
[webrtcgateway] Send filetransfer text messages only on unencrypted files
[webrtcgateway] Added OMA RCS - RCC 0.7 support
[webrtcgateway] Send OMA RCS - RCC 0.7 message to sip devices when filetransfer is received
[webrtcgateway] Fixed timestamp in filetransfer data
[webrtcgateway] Added support for receiving RCS.07 filetransfers messages
[webrtcgateway] Support 'pull' MSRP filetransfers in webrtcgateway
[webrtcgateway] Added until to filetransfer JSON and metadata
[webrtcgateway] Use MSRP transfer id as identifier for incoming filetransfer

Version 6.2.0

May 10th, 2023

[webrtcgateway] Added upload support for filetransfers from webrtc
[webrtcgateway] Added filetransfer message to send to webrtc end-points
[webrtcgateway] Refactored filetransfers internally to datatype class
[webrtcgateway] Extend filetransfer meta storage
[webrtcgateway] Reworked storage conditions for filetransfers
[webrtcgateway] Remove transferred files when originator deletes message
[webrtcgateway] Send CPIM filetransfer messages to all end-points
[webrtcgateway] Only send plain text filetransfer messages to SIP
end-points and don't store them
[webrtcgateway] Added extra field to indicate server generated messages
[webrtcgateway] Moved SIP application handlers to seperate file
[webrtcgateway] Prevent traceback by handling videoroom update event
[webrtcgateway] Fixed early media race condition
[webrtcgateway] Fixed filename in filetransfer route
[conference] Added setting to disable moh in conference rooms

Version 6.1.0

February 22nd, 2023

[webrtcgateway] Added initial support for asynchronous file transfers
[webrtcgateway] Added houskeeper method to remove expired filetransfers
[webrtcgateway] Added support for sending In-Dialog SIP messages
[webrtcgateway] Added the ability to get/set custom headers in a SIP call
[webrtcgateway] Added support for deleting messages with a custom SIP message
[webrtcgateway] Added support for PGP key lookup with a custom SIP message
[webrtcgateway] Added message storage for the originator of incoming SIP messages
[webrtcgateway] Added work around to prevent tracebacks on non UTF-8 messages
[webrtcgateway] Fixed push tokens pickle storage
[webrtcgateway] Fixed public key lookup from JSON storage backend
[webrtcgateway] Fixed removing data with maintenance script when using JSON storage backend
[webrtcgateway] Fixed removing messages in JSON storage backend
[webrtcgateway] Fixed storing of non CPIM messages
[webrtcgateway] Fixed returning error when API token is not found
[webrtcgateway] Fixed storing failed messages when the account is disconnected
[webrtcgateway] Fixed replies to refer/subscribe SIP messages
[webrtcgateway] Renamed sylk-db-maintenance script to sylk-db
[webrtcgateway] Extend sylk-db script with more functionality
[webrtcgateway] Support newer Janus versions
[webrtcgateway] Refactored message handling in webrtcgateway main app
Updated default file location for playback application
Improve default settings and include playback URI by default
Updated docs

Version 6.0.0

October 12th, 2021

Migrated to Python3
Added log dir server settings
Fixed loading mime types in a sandboxed environment
Added trace_notification server setting
Set relative ./var path if started from local directory
Log SIP session failure_reson
Log spool directory at start
Log available codecs at start
[webrtcgateway] Added database maintenance script (init db/remove data)
[webrtcgateway] Added push requests for messages
[webrtcgateway] Added config option to make conference room persitent (don't remove uploaded files)
[webrtcgateway] Added config option to send a push notify when first participant joins
[webrtcgateway] Added config option to disable video in a conference room
[webrtcgateway] Added offline storage for text messaging
[webrtcgateway] Added real time message syncronization between end-points
[webrtcgateway] Added authorization token generator for external SIP devices
[webrtcgateway] Added sync conversations journal for external SIP devices
[webrtcgateway] Fixed webserver and screensharing resource for conferences
[webrtcgateway] Fixed sending conference info notifications
[webrtcgateway] Fixed handling screen sharing in conference
[webrtcgateway] Added message storage functions/API to websocket connection
[webrtcgateway] Refactored token storage
[webrtcgateway] Fixed Janus trace logging
[webrtcgateway] Fixed saving shared screens in conference
[webrtcgateway] Fixed dispatching messages in conference app
[webrtcgateway] Fixed conference subscription for sessions from sylkserver
[webrtcgateway] Fixed conference chat destination
[webrtcgateway] Fixed websocket for communicating with janus
[webrtcgateway] Fixed websocket to webrtc client
[webrtcgateway] Fixed encoding deviceid in base64
[webrtcgateway] Fixed method encoding for file uploads in webrtc webserver
[webrtcgateway] Fixed resource register for webrtc application webserver
[webrtcgateway] Fixed webserver root and path encoding
[webrtcgateway] Update WebRTC diagram
[webrtcgateway] Fixed settings conference room file transfer directory
[xmppgateway] Fixed encoding SDP attributes in xmpp gateway

Version 5.7.0

October 14th, 2020

Added (external) IMAP authentication for webrtcgateway
The authentication type can be configured per domain in auth.ini
Added apache-cassandra as storage backend
Added session_id based on callid to conference invite
Added supoort for audio only conferences
Added support for 'ping' in sylkrtc protocol
Added support for early media
Added user agent to token storage
Changed Firebase push notifications to Sylk push server
Exposed call-id in API for SIP calls
Fixed destroying videoroom when client is disconnected
Fixed push media types
Fixed rejecting SIP methods in webrtc app
Improved logging of push notifications
Improved webrtc video room logging
Mention how to enable debug mode
Read device token from push response on 410 code
Replaced call-id by random uuid in push notification and conference invite
Replaced session id on incoming call to uuid from callid
Store push tokens and parameters in pickle file
Use push notification token for VoIP

Version 5.6.0

April 7th, 2020

Added videoroom capability to raise/lower your hand
Added videoroom capability to send mute command to all participants

Version 5.5.0

February 14th, 2020

Fixed typing information
Converted method to static method
Silenced some IDE weak warnings
Removed commented out variable in debian rules
Explicitly use python2 in shebang lines
Do not ship a sysv init script anymore
Simplified MANIFEST.in
Increased debian standards version to 4.5.0
Use pybuild as the debian build system
Updated copyright years

Version 5.2.0

February 27th, 2019

Updated TLS certificates

Version 5.1.1

February 27th, 2019

Fixed attribute name to match latest sipsimple internals
Fixed a race condition that leaked audio resources

Version 5.1.0

February 25th, 2019

Replaced obsolete ptype value for subscribers
Avoid unnecessary list copy
Removed unnecessary boolean variable
Do not reset the proposed_streams attribute when the session is rejected
Reordered some operations for consistency
Make sure the greenlet attribute is always removed when greenlet exits
Removed code that duplicated cancel handling inside reject handler
Fixed spacing to be PEP-8 compliant
Check if attribute is present
Removed unnecessary attribute from AudioStream
Avoid using tracking variables in favor of a more pythonic code
Fixed memory leak that did not release streams when canceling proposals
Use setter syntax for defining properties
Turned unnecessary list comprehensions into iterators
Always clean up proposed streams when cancelling proposal

Version 5.0.0

October 5th, 2018

[webrtcgateway] Removed bogus attributes
[webrtcgateway] Fixed exception handling for validating json model
[webrtcgateway] Cleanup accounts when the client connection is lost
Removed data duplication and inconsistent attributes
Fixed argument types
Moved DNS lookup log message to debug level
Simplified building the result of the DNS lookup
Simplified and fixed DNS lookup when no outbound proxy is involved
Generalized sylkrtc json model mapping
Removed unnecessary default value
Avoid duplicate indexing when getting the data
Use python3 compatible form for except statements
Avoid unnecessary attach/detach in janus if DNS lookup fails
Added debug_level setting to ServerConfig
Added ability to toggle between configured log level and debug at runtime
Removed unused import and order imports alphabetically
Set videoroom bitrate to overwrite janus' low default of 256Kb/s
Added API call to allow a client to update session parameters on the fly
Register atexit handler to dump memory debug info
Simplified and enhanced application loading/accessing code
Fixed spurious SIPApplicationWillStart post and handling early stop
The __init__ method should not return any value
Made policy matching logic consistent and simplified code
Refactored logging

Updated code to use the new logging features from python-application.
Rewrote system logging to be more readable and easier to interpret.
Implemented trace loggers based on notifications to separate and
isolate them from the code that generates the trace log events.
Added a trace logger for DNS lookups.

Added bitrate and video_codec configuration options for video rooms
Fully cleanup the connection handler resources when client disconnects
Simplified getting the session with the handle_id
Rewrote VideoRoomSessionContainer to not use weak references.
When sessions were removed from the container, there was no guarantee
that there was no other reference to them somewhere else in the code,
which would cause dangling id-to-session mappings to be kept around
in the container for undetermined periods of time.
Renamed VideoRoomSessionContainer to SessionContainer
Fixed order of operations during ConnectionHandler cleanup
Added PublisherFeedContainer class for holding videoroom publisher feeds
Use SessionContainer to hold SIP sessions instead of using 2 mappings
Provide more context in log messages to identify where they're generated
Changed default max_bitrate value from 4Mb/s to 2Mb/s
Automatically adjust participant bitrate based on number of participants
Do not warn about missing session (it might've been destroyed elsewhere)
Do session cleanup after all processing
Separated session hangup event handler
Simplified handling hangup events as they do not have an associated code
Eliminated delays when deleting sessions and video rooms
Avoid an unnecessary base session lookup
Verify that the detached feed belongs to the requester
Use dict constructors instead of dict literals for readability
Fixed race condition when deleting SIP session
Fixed exception while deleting subscriber sessions during hangup events
Use discard as session might not be yet added to the video room
Handle JanusError exceptions and relay errors back to client
Properly release resources when getting JanusError exceptions
Do not treat DNS lookup errors as API errors
Fixed invite to conference to not include oneself
Simplified finding the invited accounts
Made method for handling conference invites public
Made VideoRoomContainer consistent by not relying on weak references
Moved video room access validation from ConnectionHandler to VideoRoom
Simplified building request data for SIP calls and registrations
Removed unnecessary room attribute
Simplified code and increased readability
Unified handling of exceptions for the video room join operation
Send back error responses for videoroom-ctl requests with missing fields
Use modern syntax for catching exceptions
Improved PEP-8 compliance
Removed copyright notice from script
Updated license
Use the actual config type and file to find out which files are read
Improved some log messages
Do not log traceback twice
Removed duplication of startup error handling
Added run method on SylkServer
Use SylkServer.run to decouple from SylkServer's internals
Refactored startup script and configuration handling

Removed command line option to specify the pid
Removed command line option to specify the main configuration file
Added a command line option to specify the configuration directory
Added a command line option to specify the runtime directory
The main configuration file name is now immutable (config.ini)
The pid file name is immutable (sylk-server.pid)
All configuration files are now treated consistently
Properly read both local and system wide configurations
Do not initialize the runtime directory when not forking

Simplified the setup.py script and improved PEP-8 compliance
Added requirement on the jsonmodels python package
Do not split statements over multiple lines when not needed
Improve validator efficiency by using objects with a validate method
Do not use slow regular expression for validating SIP URIs
Fixed finding resources when forking and started with a relative path
Log the resources directory on startup
Better names for some of the json model classes
Added LimitedChoiceField for json models
Replaced emdash HTML code with the actual character
Use a local copy of bootstrap.min.css
Updated authors and sponsors
Split videoroom-ctl handler into subhandlers per option
Simplified getting static web resources for webrtcgateway
Make the web resource available as a property
Fixed handling ack messages
Simplified getting the result from janus messages
Added support for setting the active participants in a room
Log unhandled exception details
Only update active participants if different
Added UniqueStringListField JSON model field type
Moved normalization of the active participant list to the JSON model
Use the UniqueStringListField type for the invited participants
Split operation handlers into request and event handlers
Moved boilerplate code out of the request handlers
Moved general exception handling into operation handlers
Wait for the connection to be ready once before processing operations
Added generic sylkrtc request to model mapper
Simplified instantiating sylkrtc models from requests
Removed code that was never reached
Restructured code to avoid unnecessary extra return statements
Removed unnecessary additional private method
Fixed race condition when a connection handler is stopped while starting
Fixed race condition when cleaning up a connection handler while stopping
Removed unnecessary partial function
Simplified keepalive callback
Simplified webrtcgateway package structure
Removed unnecessary references to the janus backend everywhere
Renamed attribute to better reflect its function
Renamed some arguments and variables for clarity
Refactored sending keepalives to be internal to the protocol
Simplified stopping the keepalive timer
Grouped public/private API methods together
Explicitly added protocol methods to JanusBackend
Removed redundant namespace prefix
Optimized code to only send keepalive messages when necessary
Simplified code sending janus requests
Do not allow request arguments to overwrite core attributes
Do not attempt to send messages after the connection was closed
Fixed return value from deferred callback
Fixed error with cancelling already called timer
Replaced dependency on python-jsonmodels with internal module
The jsonobjects module was written to replace the functionality
provided by jsonmodels while offering the following advantages:

More than 10 times faster
Much lower memory footprint
No need for manual validation
Objects are validated on creation and every time they are modified
Guaranteed data consistency throughout the lifetime of the objects
JSON arrays are first class objects allowing them to be used standalone
JSON arrays can be embedded alowing for multi-dimensional arrays
JSON arrays can define both item and array level validators
Better semantics for optional properties and default values

Added ConnectionHandler.send method to replace notify and _send_response
Renamed VideoRoom to Videoroom in class names
Refactored client API to simplify it and make it consistent
Simplified AbstractProperty getter
Improved data extraction speed by a factor of 2
Have the 'in' operator to check if a property is defined in a JSONObject
Fixed code formatting warnings
Removed unnecessary class
Removed unnecessary util.py module
Optimized returned value
Allow LimitedChoiceProperty to be optional and have a default value
Keep all internal SIP URIs as AORs for consistency
Moved FixedValueProperty and LimitedChoiceProperty to jsonobjects.py
Moved validators to validators.py
Added DisplayNameValidator that removes optional quotes
Added type hinting for sylkrtc models
Fixed DisplayNameValidator to work with empty names
Use JSON models to encode/decode messages to/from Janus

Encapsulated Janus session and plugin handle functionality to classes
Simplified generating Janus requests and processing Janus replies
Correctly handle Janus errors sent as error events in success replies
Split Janus event handlers to improve readability and maintability
Moved data validation to the JSON model validators

Use UTC timestamps in push notifications.
The local timezone of sylkserver is both irrelevant for the client that
receives the push notification and it's also 20 times slower to generate
a server local timestamp than a UTC timestamp (115us vs 6us)
Use JSON models for Firebase push notifications
Added method to send conference invite push notifications
Renamed push notification methods from xyz_session to xyz_call
Simplified signature for the push notification functions
Fixed race condition with cleaning subscriber sessions in conferences
Split debian dependencies one per line
Removed dependency on dummy package
Removed unnecessary version dependencies
Updated debian package description
Improve performance for push notifications by reusing the request
Log push notifications at DEBUG level
Avoid composing log message unless needed
Use log.warning instead of its alias log.warn
Updated python-application version dependency
Updated dependency list in the INSTALL file
Use default sound for push notifications
Send push notifications for conference invite events
Use relative imports for webrtcgateway
Adjusted the WebRTC gateway URL in the sample configuration
Fixed URL class to not mangle template variable
Adjusted conference welcome message
Fixed sending ACK to the wrong address by SofiaSIP
Ignore the Janus SIP plugin ringing event for now
Ignore the Janus SIP plugin progress event for now
Fixed matching empty file:// origins with the latest autobahn
Moved debian dependency to the proper package
Drop support for autobahn older than 0.12
Removed deprecated pycompat/pyversions files
Removed unnecessary .PHONY target
Removed code that prevented sylkserver from restarting on upgrades
Do not install janus configuration files with sylkserver-webrtc-gateway
Removed defaults file
Allow sylkserver to leave a core file when run by systemd
Removed build dependency on obsolete package
Increased debian compatibility to 11
Synced MANIFEST.in file with latest file changes
Rename .ini.sample files to .ini in the debian package
Fixed corrupted sound file names in the deb if name contains spaces

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
Fixed compatibility with AutoBahn >= 0.12
webrtcgateway: enable optional SRTP-SDES for outgoing calls
Adapted to changes in SIPSIMPLE SDK 3.0.0
Update INSTALL
Use collections.Counter instead of a custom defaultdict
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
Update bundled sylkrtc.js library
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 2.10.1

June 16th, 2015

Added missing dependency on python-werkzeug

Version 2.10.0

June 15th, 2015

Add global web server that applications can tap into
Refactor web capabilities of conference application
Make loading and starting applications more resilient
Log exceptions when initializing/starting/stopping applications
Log application map on startup
Reorganized some code in the echo application
Fix building FileSelector objects out of RoomFile objects
Fixed using undefined variable

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
Use 127.0.0.1 if the local address could not be determined
Refactor managing the single account SylkServer currently uses
Rename sylk.extensions to sylk.streams
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
Use '.log' as the extension for log files
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
Don't use signal.pause to pause the main thread
Always disable echo canceller
Improved logging
Ignore audio device change notifications
Removed dependency on python-backports
Dropped Python 2.6 support

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
Patch sipsimple.session to use ServerSession objects
Modified ChatStream to send MSRP REPORT chunks manually
Made contact_header optional in ServerSession.connect
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
Dropped support for Python 2.5

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 compatibility with Python 2.5
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