RTCKit allows you to emded telephony into your web application. It is really simple:
  • Make a test call without registration
  • Register and get one free unlimited channel for development
  • When you are ready, make a payment and launch into production.

How it works

You embed into your site a special component - Flash WebPhone, which allows your users to make calls from their browser without installing any additional software. WebPhone is contolled using our JavaScript API.

Our servers convert voice and video data from RTMFP protocol, used by WebPhone, into SIP or Jingle protocol, which can then be used to interract with any traditional and IP telephony systems.

RTMFP is the most suitable protocol for real-time communications from Adobe. Unlike RTMP, it is implemented on top of UDP, which ensures minimal latency and high tolerance to network problems. This is the main advantage of RTCKit over similar services.

Technical specifications

  • Protocols:
    • SIP
    • Jingle

  • Audio codecs:
    • speex/8000/16000/32000
    • G729
    • G711 (a-law and μ-law)
    • iLBC
    • GSM

  • Video codecs:
    • H264

  • Acoustic echo cancellation (starting from Adobe Flash Player 10.3)

  • Tone dialing (DTMF)