Amsip SDK – webrtc interop

amsip and webrtcWe are happy to announce the ability to interoperate with sip and webrtc projects.

This is still in beta and additional development will come to complete the feature set in 2014.

Today, you can start testing incoming call from jssip, a JavaScript sip/webrtc tool, against VoIP by Antisip.

We have added ICE with TURN support, ability to mix RTP and RTCP, support for SAVPF profile. Other optional features may come in 2014.

Interoperability with webrtc is a great challenge, but in the long term, the test showed lots of minor issues that will not make it plug and play easily in existing infrastructure.

The main issue that I see is the usage of SAVPF profile. If you make a call using the usual AVP profile, webrtc sip app will reject it. If you use SAVPF, traditional sip app will reject it… So? Let’s propose both profile? And what about interoperability with lightweight sip app not supporting this?

As a conclusion, webrtc interoperability is great and will open perspective. It will at least introduce new usage for sip and deployments!

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