News

Blink Qt: Blink Qt 0.6.0 for Linux with MSRP chat sessions

Added by Adrian Georgescu over 8 years ago

Blink Qt version 0.6.0 is the first Linux program that features chat sessions using MSRP protocol part of SIP SIMPLE suite.

Changelog

  • Added chat sessions using MSRP protocol
  • Adjusted to the latest changes in SIP SIMPLE Client SDK
  • Fixed several memory leaks
  • Simplified processing Google contacts authorization and fixed some bugs
  • Set focus to the appropriate widgets during Google contacts authorization
  • Fixed unpickling for BonjourNeighbours
  • Modified how sipsimple is started * Improved selection of the winning presence state
  • Removed donate menu action
  • Modified how sipsimple application is started

To install or update the software go to: http://icanblink.com

SIP SIMPLE Client SDK: SIP SIMPLE client SDK 1.1.0 released

Added by Adrian Georgescu over 8 years ago

SIP SIMPLE client SDK and command line sipclients version 1.1.0 have been released with bug fixes and improvements.

Changelog

python-sipsimple (1.1.0) unstable; urgency=medium

  • Updated opus codec to version 1.1
  • Cleanup opus.c and fix compilation warnings
  • Always put useinbandfec in SDP for opus codec
  • Relax codec matching when doing SDP negotiation
  • Use single global c line when creating SDP
  • Added function to manually refresh sound devices
  • Added trace_msrp setting
  • Fixed SIP and PJSIP logging
  • Fixed not posting state change notifications for different provisional
    responses
  • Changed notification API for renegotiating streams
  • Renamed streams to proposed_streams on SIPSessionNewProposal
  • Renamed streams to proposed_streams on SIPSessionHadProposalFailure
  • Added audio.muted runtime setting to SIPSimpleSettings
  • Post SIPApplicationWillEnd even if Engine failed
  • Renamed MediaStreamRegistrar to MediaStreamType
  • Properly handle mutex creation failures
  • Added missing context attribute to MediaStreamDidFail notification
  • Fixed memory leak by initializing the handler after the stream initialized
  • Moved AudioConference to audio module
  • Added helper functions to allocate and release memory pools
  • Create null sound port only once and reuse it
  • Simplified audio device fallback code
  • Fixed crash when in-dialog request fails to be sent within a subscription
  • Properly patch dnspython to make it nonblocking
  • Added initial_delay to WavePlayer, replacing initial_play
  • Always use timezone aware timestamps in MSRP streams
  • Make sure MSRPlib always gets bytes, not unicode
  • Always return unicode as the received chat message body
  • Post SIPEngineGotException also if Engine fails to start
  • Make send_composing_indication refresh argument optional
  • Return default refresh value in ChatStreamGotComposingIndication if not
    specified
  • Don't set last active timestamp automatically
  • Always pass copies of stream lists in Session notifications
  • Don't compile WebRTC AEC if machine is not x86 or x86_64
  • Raised Cython version dependency to 0.19.0
  • Cleanup Cython imports and remove no longer needed workarounds

To install or upgrade the software go to http://sipsimpleclient.org/projects/sipsimpleclient/wiki/SipInstallation

Blink Cocoa: Blink for OSX with OTR end-to-end encryption

Added by Adrian Georgescu almost 9 years ago

Blink can now setup MSRP sessions that are encrypted end-to-end using OTR protocol.

SIP2SIP: PSTN gateway unavailable for SIP2SIP (15 comments)

Added by Adrian Georgescu about 9 years ago

It seems like our PSTN gateway provider went out of business and we were left holding the bag (GRN VoIP). We are working on finding another PSTN termination partner. The balances stored in SIP2SIP platform will remain unchanged, no money was lost.

I apologise for this and keep you posted.

News: September 4th: We have a new PSTN carrier now. Please provide feedback about quality.

Adrian Georgescu

SylkServer: SylkServer 2.5.0 release

Added by Adrian Georgescu about 9 years ago

Changelog

  • 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

To install or upgrade go to:

http://projects.ag-projects.com/projects/sylkserver/wiki/Installation

Blink Qt: Blink Qt 0.5.0 for Linux with Opus codec

Added by Adrian Georgescu about 9 years ago

Changelog blink (0.5.0)

  • Adapted to changes in SIP SIMPLE Client SDK
  • Enabled Opus codec
  • Refactored PresencePublicationHandler in order to simplify it
  • Set default sample rate to 32 kHz
  • Fixed exception if Google contact has no name nor company
  • Fixed handling file URLs on different platforms
  • Fixed losing contact icons
  • Fixed computing hours in history entries
  • Fixed setting display name in history entry when URI is a phone number
  • Avoid publishing presence state twice when xcap settings change
  • Allow Ctrl+Delete/Backspace to hangup sessions because KDE steals Ctrl+Esc
  • Raise and activate preferences window when triggered if already visible
  • Do not allow toolbar to be hidden
  • Removed compatibility with python 2.6

To install or update the software go to: http://icanblink.com

SIP SIMPLE Client SDK: SIP SIMPLE client SDK 1.0.0 released

Added by Adrian Georgescu about 9 years ago

SIP SIMPLE client SDK version 1.0.0 has been released. The underlying core has been updated and uses Opus codec by default. On Linux, there are major improvements related to using native ALSA audio drivers and WebRTC echo-cancellation engine. These allows for full duplex, studio quality audio without the need to use a headset.

Changelog

  • Updated core to PJSIP 2
  • Added gain control and high pass filter to audio processing
  • Added Opus codec support
  • Added support for RFC5768 (ICE option tag)
  • Added enabled setting for echo canceller and echo_canceller settings group
  • Fixed echo cancelling when using 32kHz sample rate
  • Always disable sound device when idle
  • Removed unused ignore_missing_ack feature
  • Removed engine shutdown workaround
  • Removed TLS protocol setting
  • Removed NAT detector from SIPApplication
  • Don't cap codecs based on sample rate, let PJSIP resample
  • Disabled narrowband speex
  • Fixed starting media stream if ICE fails early
  • Don't reset stream statistics, always report absolute values
  • Don't add BonjourAccount to AccountManager if there is no bonjour support
  • Set session state to terminated when ended before starting
  • Prevent PJSIP from switching transports automagically
  • Dropped support for Python 2.6

To install or upgrade the software go to http://sipsimpleclient.org/projects/sipsimpleclient/wiki/SipInstallation

SIP2SIP: Special Blink Qt Edition for SIP2SIP (2 comments)

Added by Adrian Georgescu about 9 years ago

This version of Blink Qt for Microsoft Windows is integrated with SIP2SIP and features ultra-wideband OPUS codec and acoustic echo cancelation.

SIP Client Features

  • SIP2SIP account
  • Acoustic Echo Cancellation
  • Ultra-wideband audio (OPUS codec 48kHz)
  • Wideband audio (G.722 and Speex codecs)
  • PSTN audio (G711, GSM and iLBC codecs)
  • Multiparty Conferencing
  • Google contacts integration
  • DTMF, Hold, and Recording
  • sRTP encryption
  • Contacts Management (XCAP protocol)
  • SIMPLE Presence (RLS Subscriptions)

SIP Service Features

  • TLS encryption
  • Contacts sync between multiple devices
  • Presence sync across multiple devices
  • Access to SIP server account settings web page
  • PSTN termination (paid option)
  • Interoperability with Jingle domains (jit.si)

To download go to:

http://download.sip2sip.info

SIP2SIP: SIP2SIP conference bridge now has OPUS codec

Added by Adrian Georgescu about 9 years ago

SIP2SIP conference bridge now with OPUS codec and PSTN access number

http://wiki.sip2sip.info/projects/sip2sip/wiki/SipTesting

SIP2SIP: TLS port changed to 443

Added by Adrian Georgescu about 9 years ago

To increase the chance of connecting to the SIP server in adverse conditions imposed by the presence of SIP ALG enabled routers, we switched the default TLS port to 443.

If your SIP Client uses DNS for resolving sip2sip.info server addresses, there is no need to change anything on the client side.

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