Antisip - Desktop and Mobile Telephony

SIP, WebRTC, Audio, Video on all platforms!

SIP and WEBRTC?

Check out the amsip SDK!

WebRTC was designed with all nice features to achieve best quality and security. A state of art SIP Application will have them all! And you know what: they are 100% compatible!

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  • Latest compliance with SIP, WEBRTC, Audio and Video.
  • We offer security with TLS, SRTP, SDES, DTLS.
  • We save your infrastructure bandwidth with ICE and STUN.
  • We monitor/optimize video bitrate in real time with GOOG-REMB, TMMBR, Zig Zag monitoring.
  • We minimize bitrate and provide fast video recovery upon loss with RTCP Feedback.
  • We provide compatibility between SIP & WEBRTC.

SIP only?

Check out eXosip2 SDK!

osip and eXosip2 open source projects were started in 2001 by us. We also offer commercial license for those historical projects.

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  • Latest compliance with SIP.
  • We offer security with TLS.
  • A strong experience and stability. Project was started in 2001.
  • A flexibility for the API.
  • Our strong support & expertise with SIP and eXosip2 will be a must.

RTP and MEDIA only?

Check out the media SDK!

In amsip, we aggregates numerous famous open source projects. They can be used out of amsip too under commercial license.

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  • Latest compliance WEBRTC, Audio and Video.
  • We offer security with TLS, SRTP, SDES, DTLS.
  • We save your infrastructure bandwidth with ICE and STUN.
  • We monitor/optimize video bitrate in real time with GOOG-REMB, TMMBR, Zig Zag monitoring.
  • We minimize bitrate and provide fast video recovery upon loss with RTCP Feedback.
  • We provide compatibility between SIP & WEBRTC.
  • Our strong support & expertise with RTP and media will be a must.

All Platforms

Our toolkit are written in C and available on all desktop and mobile platforms.

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  • Android since API 14 (armv7/armv7s-neon/arm64/x86).
  • iOS since iOS6 (armv7/armv7s/arm64).
  • Windows desktop.
  • Macosx desktop.
  • Linux desktop.
  • Raspberry (every version).
  • Most probaly your arm platform is supported!

Client side: Native SDP or WebRTC SDP

Adaptative bandwidth

Adapt to congested network with TMMBR/TMMBN.

Video fast recovery

Adapt to lossy network with NACK, PLI.

No congestion

Endpoints establish direct connections with ICE, STUN, TURN.

Test with our sip service: sip.antisip.com

SIP Security

SIP over TLS with SNI, WEBSOCKET over TLS

Media Privacy

Transparent forwarding, if any. Endpoints should encrypt with DTLS-SRTP.

Free service

Service requires no bandwidth. The best comes with no extra cost.