SIP SIMPLE client SDK 1.0.0 released

Added by Adrian Georgescu over 10 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


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