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
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