Update WebRTC SIP SDK for web

Please download our latest update of WebRTC SIP SDK.

Change log:

  • outgoing call connection timeout reduced
  • custom headers for invite messages made possible
  • customizable user-agent SIP field added
  • DTMF implemented
  • possibility to perform multiple simultaneous calls added
  • video/audio recording implemented
  • application interface changes (done more consistent, obsolete methods removed, parameters changed, renamings)
  • possibility to select camera and microphone added
  • tag added to TO SIP header field
  • customizable register expiration timeout
  • some of the deprecation warnings resolved
  • hold redone to be more flexible with different servers
  • various fixes
  • new developer manual written