This month we attended the OpenSIPS Summit in sunny Amsterdam. The Summit consisted of two days of presentations, a day filled with demos and a day of training in OpenSIPS and FreeSwitch.


The Summit took place in the Radisson Blu Hotel in the heart of Amsterdam, which is unique for its combination of new architecture and old building façade. The hotel is actually made up of two buildings, connected by a tunnel which runs under a street.

The presentations took place in a nice big meeting room where the 100 or so participants could sit comfortably.

Day one featured a packed schedule with lots of interesting talks about the OpenSIPS roadmap, ways to use OpenSIPS, and updates about neighbouring projects like Asterisk and PJSIP. During day two the focus was more on how people apply OpenSIPS in their projects and the experience they had.

The first takeaway is that in open source VoIP, projects like OpenSIPS are treated as a sort of Lego blocks: everybody picks some sort of the software that’s available and puts it together in a unique way which adds value for them or their customers. Although this conference was officially about OpenSIPS, it was as much about Asterisk, FreeSwitch, PJSIP, Kamailio, rtpproxy or rtpengine.

Common topics

There were some topics that were covered in several presentations.

  • Cloud: in several presentations, we learned about how you can run OpenSIPS in the cloud. Be it Amazon, Google or another provider. Lessons learned are shared and can be applied to other platforms as well. You can’t put a price on the experience you gain from others, using what works well and preventing missteps other people before you already made.
  • WebRTC: many projects include some way to use WebRTC together with SIP. Since WebRTC is now supported on most browsers, it is a full replacement for technologies like Flash and Java that filled this space in the past. We expect to see a lot of in-app communication, but also softphones in the browser to appear in the coming years. This might even replace a lot of calls we currently still make using the Plain Old Telephone System and instantly adds a viable way to do video as well. Who knows, in the future, we might not even put our company’s phone number on our site, just a Call button might instantly get you connected to a representative using nothing more than your browser.
  • Mobile clients: the world looks very different from the view of mobile devices. Where for desk phones it’s no problem to have a connection open to the server 24/7, it really drains the battery on mobile clients. And the constant switching from the 4G mobile internet to home or company wifi makes it hard to reach a client at any of the addresses they use to register. Which one is valid? There’s no way to tell. You just can’t count on any of them to be reachable at any point in time. So multiple companies put the concept of SIP registrations on its head and only register once in 24 hours. And when a call comes in, they use push notifications to wake up the mobile device and force a valid registration, just a split second before offering the call. This way they get the best of both worlds: keep using SIP for what it does best, and use native features of the mobile platforms where needed.

OpenSIPS Summit 2019

All things considered, we had two very productive days. We learned a lot from both the presenters and attendees, and it surely helps to put a face with the names in the community. We hope to be back next year for more open source VoIP awesomeness from the community. And for sure we wouldn’t want to miss the raffle. Anybody would like to walk away from a conference with a swanky leather jacket!