PORTSIP vOip sdk

Accelerate and Simplify the Development of Voice, Video and IM

Develop audio, video and IM Apps

The PortSIP VoIP SDK is a SIP SDK with intuitive and flexible APIs that developers can utilize to quickly implement IP-based voice and video applications and meet the requirements of Unified Communications, IoT/M2M and public-Safety critical communications without any need for complicated integrations or VoIP know-how.


Customize Apps for Video Communication

PortSIP VoIP SDK is a state-of-the-art standard based client that is able to add HD video and HD audio communication to any product and service. The SIP SDK is available for all leading operating systems, enabling a fast time-to-market on all platforms, including iOS, Android, macOS and Windows.

Quickly get apps and devices to market

Many new standards are being developed to address the needs of fast-growing UC, IoT and M2M market. However, keeping up with these ever-changing standards is a challenge for developers.

PortSIP VoIP SDK comes with a sample application in source code, a detailed suite of APIs and comprehensive documentations to help with reducing development, integration, and testing efforts.


Royalty free license, bundle discount

All PortSIP VoIP SDK package licenses are royalty free. The buyer may freely distribute his applications built based on the SDK. Buyer pay a one-time license fee and get in return for the rights to develop unlimited applications and distribute his applications to as many end-users as he like without paying any royalties or license fees.

The Bundle License with DISCOUNT is available if  SIP SDK is purchased with PortSIP PBX together.

Key Highlights

  • Support iOS, Android, macOS and Windows
  • Support .NET, Java, Swift, Xamarin and Objective-C, C++, Delphi, JS
  • Support callkit for iOS
  • Support 3GPP IMS Conferencing and 3GPP Call-Waiting
  • Support Present Agent (PUBLISH)
  • Provide audio, video, presence, IM in unified API for all platforms
  • Audio codecs: G.711 aLaw/uLaw, G.722.1, G.722, SPEEX, SPEEX-WB, AMR, AMR-WB, GSM, iLBC, G.729, Opus, SILK
  • Video codecs: H.264, VP8, H.263, H.263-1998
  • AMR, AMR-WB codecs: comply with RFC 4867 support, including Bandwidth- Efficient and Octet – Aligned mode, amr-red, amr-maxptime
  • Support 3GPP IMS features: Call-Waiting and Conferencing focus
  • Allows monitoring the dialog status
  • Support add/obtain custom SIP header and modify outgoing SIP header
  • Proven operability with a wide range of SIP-based network equipment including Cisco, Qualcomm, Agilent, Keysight, Avaya, Unify, HP, Philips, NEC and others
  • Product support and sample application included to help get developers quickly up to speed in designing and deploying their application
  • Deliver superb video quality even in challenging network conditions through advanced algorithms combined with PortSIP’s extensive experience in networking