Daniel-Constantin Mierla

The presentation tries to reveal a list of open source small applications that are typically not in focus, but very useful for helping to build RTC applications, troubleshoot or operate SIP-based platforms. Among others, they can be used to automatize, monitor or simulate endpoints for new features or scenarios, saving time and reducing costs during the development and testing phases. The scope is also to engage the audience to share their experiences and let everyone know about other tools.

Kamailio is an open source SIP server that uses a scripting language for its configuration file to enable flexibility in deciding the routing of SIP messages. Starting with version 5.0, besides its native scripting language, Kamailio allows writing the routing logic in several other programming languages such as Lua, JavaScript, Python and Squirrel. This presentations aims to reveal the benefits and drawbacks of using any of these scripting languages for building scalable real time communication systems and services.

Real time communications evolved at a high pace in the last decade, no longer being an isolated service used primarily for voice and text sessions. Connectivity with other social media channels, integration of artificial intelligence or interaction with smart home or work environments requires lot more flexibility inside the SIP routing engine. Current version of Kamailio tries to cover the needs from fast routing needs of a performant load balancer, with a self designed scripting language, to the fast prototyping and development of innovative features, with a new scripting framework that allows leveraging the large set of extensions and libraries offered by languages such as Lua, JavaScript or Python.

This presentation focuses on the Kamailio Embedded Interface (KEMI) framework, going through the supported scripting languages, their benefits and drawbacks. Doors are now open to write Kamailio extensions also for people familiar with these programming languages, no longer only for C programmers, diversifying the sources of developers that can join the community of the project.

Kamailio is an open source SIP (RFC3261) server that can be used for building real time communications systems for IP telephony, instant messaging or presence. Seeral years ago it introduced a Lua embedded interpreter to allow more flexibility in routing calls. Recently, it was extended to allow entire RTC routing logic to be written in Lua. This talk focuses on showing the insights of implementing the LUA RTC API in Kamailio, tips and tricks discovered to make it as fast as the native pre-compiled routing language, and the benefits it bring to enhance RTC systems and integrate with third party services, such as social networking platforms (e.g., sending a Twitter DM on a missed call).

Real time communications has evolved at a fast pace in the last decade. From what it used to be direct voice calls or testing via classic or mobile telephony, nowadays everyone connects through a multitude of channels, dedicated messaging systems or other social media channels. Kamailio is an open source SIP server developed since 2001, with an initial focus on scalability for IP telephony services. To allow more flexibility and easier integration with other RTC services, an embedded LUA interpreter was added to it and with a recent development, the entire SIP routing logic can be written in Lua. The presentation will reveal insights of implementing the LUA RTC API in Kamailio, tips and tricks discovered to make it as fast as the native pre-compiled routing language, and the benefits it bring to enhance RTC systems and integrate with third party services, such as social networking platforms (e.g., sending a Twitter DM on a missed call).

The talk about software design elements that can help building high performance real time communication signaling servers, primarily based on experiences while developing Kamailio SIP server. It will look at the components in the message processing flows that can become bottlenecks and what are the options to deal properly with them, either directly in the code or via operating system tunings and configuration file parameters.

Are you one of the people who has never installed Skype (and never will)? Do you feel a sense of frustration every time someone asks to call you with Skype? Does the excruciating inefficiency of maintaining multiple profiles across Facebook, Google+ and LinkedIn give you more pain than the sound of fingernails scratching a blackboard? Have you tried free and open source alternatives, but found their usefulness limited (or completely failed to get them working)? Would you like to see a way out of this nightmare?

This talk covers the depth and breadth of VoIP and all other real-time communications, encompassing desktop VoIP, instant messaging, social networking, microblogging, server solutions and also emerging areas like WebRTC, Peer-to-peer, privacy with FreedomBox and the mobile VoIP sector.

Despite this broad agenda, the presentations from each speaker have been tightly coordinated with a coherent focus on the challenges of making it all work together comprehensively and conveniently. The speakers have come together for FOSDEM with the goal of mapping out a strategy for free and open communications standards to finally take their rightful place as the dominant paradigm.

The speakers are the leaders in the flagship open source VoIP projects and will participate in a panel discussion and question/answer session in the telephony devroom immediately after the talk.

Other people: Peter Saint-Andre Bdale Garbee Daniel Pocock Simon Tennant Evan Prodromou Emil Ivov

This talk will present a short introduction to the WebSocket protocol, how SIP/MSRP over WebSockets has updated their respective RFCs, and a live demonstration of an open-source browser based SIP user-agent built on Javascript libraries for SIP over WebSocket, MSRP over WebSocket, and JSEP, routing calls, instant messages, and file transfers through Kamailio.

Crocodile RCS has contributed MSRP and SIP over WebSocket support to Kamailio, as well as authoring the draft for MSRP over WebSocket. The talk will contain: * An short introduction to the WebSocket protocol * Brief descriptions of how draft-ietf-sipcore-sip-websocket has updated RFC 3261, and draft-pd-msrp-websocket has updated RFC 4975 and RFC 4976 * Live demonstration of an open-source browser based SIP user-agent built on Javascript libraries for SIP over WebSocket, MSRP over WebSocket, and JSEP, routing calls, instant messages, and file transfers through Kamailio * Time dependant: a look at how the use of open-source libraries and SDKs can be used by web-developers to create real-time communication applications without the need to learn or understand the details of real time session establishment and management.

Other people: Peter Dunkley

With SIP networks being used for more than just Voice/Video calls, it has become important that the SIP infrastructure provide secure signaling channels; this talk will present features of Kamailio and how they can be used to provide this security.

Besides quick update of what is new lately in Kamailio SIP Server project, the presentation will focus on showing how to use Kamailio to build a secure communication platform for SIP. Kamailio features a scalable SIP routing engine for dispatching of Voice/Video calls and Instant Messages as well as rich SIMPLE presence server for user availability states with embedded XCAP server for management of privacy rules and buddy list. With IM and Presence traffic being carried via signaling channel, encrypting SIP channel is the only way of ensuring confidentiality and protection of data exchanged. Today, Kamailio offers asynchronous TLS communication layer, tested and improved over last years for robustness and cope with the demands of telecom-grade deployments.