Tutorial for Antisip services! We have published today our first video tutorial. It will provide first time users with precise guidelines to create an account on sip.antisip.com services. The service we provide will let you call each other on internet … Continue reading →
People from sipnet service has made a simple tutorial to describe how to configure and use Voip By Antisip with sipnet.ru SIP services! Check the tutorial here!
2N has published a tutorial to configure Voip By Antisip for they 2N® Helios IP intercom systems. The tutorial is published there!. 2N provides a list of similar third party applications, but guess what, only 2 of them provides full … Continue reading →
Last year, we already achieved sip vs webrtc audio and video calls and announced it, but we didn’t stopped there and have completed internal features to better support RTCP feedback (NACK, PLI, SLI) and by adding the mandatory DTLS-SRTP encryption support. … Continue reading →
After 116 updates in more than 4 years on Google Play, Voip By Antisip reached 4.0 rating today! As shown in the history graph, Voip By Antisip started with a 2.5 rating. The initial versions wasn’t the best ones for … Continue reading →
Firewall Voip Checker is an app do SIP test on accounts and network access. It’s available on google play! Download here A default account can be used in order to test. Then, you can try (and compare) with your own … Continue reading →
osip 4.1.0 and eXosip 4.1.0 were officially publised on January, 6th 2014. As always, osip and eXosip are already complete and stable. The git repository is showing regular activity but no major change: osip 4.1.0 mainly brings built-in support for … Continue reading →
As always, we are keeping our Voip By Antisip app on google play store in sync with our latest developments. Current vbyantisip 2.0.8 has been released this week and it’s time to describe a few hints about new features coming … Continue reading →
We 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, … Continue reading →
Audio quality and latency after a dozen years of developments is always the primary concern. amsip 5.0.0 is now integrating new PLC technology. We believe we made a nice step again towards a better quality and latency experience. Under normal … Continue reading →