PortSIP VoIP SDK v15 is ready

PortSIP VoIP SDK for RCS

Jun 8, 2017 – PortSIP - A leading provider and developer of next-generation Unified Communications, released PortSIP VoIP SDK V15. This is a major update for PortSIP VoIP SDK over the last two years with fruitful new features and enhancements. We strongly recommend you to upgrade to the latest version for superior experience.

Independently developed by PortSIP, PortSIP VoIP SDK is a SIP-based client SDK designed for device manufacturers and service providers looking into accelerate and simplify development of IP-based audio, video and messaging over IP for mission-critical Apps with short time to market.

Peter Andry, CTO of PortSIP said:

After studying the feedback and suggestions arose from users and the compliance to the latest standard protocol of IETF, we have rendered a major release of PortSIP VoIP SDK V15 with improved performance and enhanced stability. Besides the enhancements, new features for supporting on iOS CallKit, compliance to 3GPP IMS standards and detection on network conditions make PortSIP VoIP SDK a top choice of framework for developing SIP-based audio and video programs.

As a leading provider of Unified Communication products and solutions, PortSIP is highly appreciated by clients for its qualified product and powerful technical support. Our major clients include Agilent, Keysight, Siemens, Qualcomm, Fujitsu, NEC etc.

Release notes:

  1.  Support callkit for iOS.
  2. Support 3GPP Call-Waiting.
  3. Support 3GPP IMS Conferencing .
  4. Support Present Agent(PUBLISH).
  5. Moved LocalIP and localPort parameters from setUser function to initialize function.
  6. New parameter "sessionId" added for setVideoBitrate and setVideoFrameRate functions, set it to specify session.
  7. Removed setDisplayName function, now use "setUser" function to set the display name.
  8. Removed detectMwi function, now use sendSubscription function to check MWI.
  9. Removed presenceOnline and presenceOffline functions, use setPresenceStatus function to instead of it.
  10. Replace createConference with createAudioConference and createVideoConference.
  11. Renamed enableCheckMwi to enableAutoCheckMwi.
  12. Renamed presenceSubscribeContact function to presenceSubscribe
  13. New parameter "sipMessage" added for callback events onInviteIncoming, onInviteSessionProgress, onInviteRinging, onInviteAnswered, onInviteUpdated, allows obtain the specify SIP header value from "sipMessage".
  14. Added roundTripTime parameter to getAudioRtcpStatistics function.
  15. Added sendSubscription and terminateSubscription functions.
  16. Added setPresenceStatus function.
  17. Added function outOfDialogRefer.
  18. Added function attendedRefer2.
  19. Added function removeUser.
  20. Added function refreshRegistration. When network changed between WIFI and LTE, should call this API to refresh registry.
  21. Added setDefaultSubscriptionTime function.
  22. Added setDefaultPublicationTime function
  23. Added setPresenceMode function.
  24. Added presenceTerminateSubscribe.
  25. Added pickupBLFCall function.
  26. iOS: New functions startAudio and stopAudio added. It will be used by callkit.
  27. iOS: callkit support added for iOSSIPSample, Added new class CallManger.
  28. iOS: Add libc++.tbd to "Link Binary With Libraries".
  29. Android: Remove API setSystemOutputMute, getSystemOutputMute, setSystemInputMute, getSystemInputMute.
  30. Fixed some other minor bugs.

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