Schedule

The schedule may still change, check periodically the website for the latest version!

Event structure:

Thursday, May 7, 2026: 08:30 – Registration
Thursday, May 7, 2026: 09:00 – 18:00 – Presentations
Thursday, May 7, 2026: 19:00 – 21:00 – Social Networking Event
Friday, May 7, 2026: 08:30 – Registration
Friday, May 8, 2026: 09:00 – 17:00 – Presentations

Thursday, May 7, 2026

08:30 ♦ Registration

09:00-09:10 ♦ Welcome
Elena-Ramona Modroiu, Co-Founder Kamailio, Germany
Welcome notes.

09:10-09:30 ♦ Kamailio – Last Year In Review
Daniel-Constantin Mierla, Co-Founder Kamailio, Germany
A walk through the most relevant events of Kamailio project, with a special focus on the development during the last year and the plans for the future. Details about what is new in the latest stable releases, Kamailio v6.0 and v6.1, and what else has been developed meanwhile.

09:30-10:00 ♦ TBA
TBA
TBA

10:00-10:30 ♦ How To Build Your Own Open Source AI Voice Bot From Scratch
Andreas Granig, CEO, Austria
Learn how to build your own open source voice bot from scratch leveraging BareSIP and using different types of architectures such as OpenAI or Gemini speech-to-speech vs sst-llm-tts approaches, cloud vs local services and how to measure the performance and accuracy of such a bot.

10:30-11:00 ♦ Coffee Break

11:00-11:30 ♦ Scalable Push-Driven Call Delivery in Distributed Kamailio Architectures
Elena Darriba, Senior Voice Engineer CloudTalk, Portugal
Delivering mobile calls through push notifications in distributed SIP infrastructures introduces complex signaling and routing challenges, particularly when operating behind load balancers and WebSocket-based client connections. In this talk, we present a production-proven approach for delivering mobile calls using Kamailio with asynchronous push notifications and transaction suspension. In our architecture, SIP signaling must maintain WebSocket affinity to the same Kamailio instance in an auto-scaling group, even when operating behind a load balancer. To achieve this, we implemented custom internal signaling mechanisms that ensure requests are routed through the correct node while preserving session consistency.

11:00-11:30 ♦ Scalable Push-Driven Call Delivery in Distributed Kamailio Architectures

Elena Darriba, Senior Voice Engineer CloudTalk, Portugal

Delivering mobile calls through push notifications in distributed SIP infrastructures introduces complex signaling and routing challenges, particularly when operating behind load balancers and WebSocket-based client connections.

In this talk, we present a production-proven approach for delivering mobile calls using Kamailio with asynchronous push notifications and transaction suspension. In our architecture, SIP signaling must maintain WebSocket affinity to the same Kamailio instance in an auto-scaling group, even when operating behind a load balancer. To achieve this, we implemented custom internal signaling mechanisms that ensure requests are routed through the correct node while preserving session consistency.

11:30-12:00 ♦ TBA
Mathias Pasquay, CEO Pascom, Germany
TBA

12:00-12:30 ♦ Implementing Real Time Text In A VoIP Infrastructure
Henning Westerholt, CEO Gilawa, Cyprus
The addition of real-time text (RTT) functions to voice and video calls is a requirement that has recently been introduced in the European Union and other countries. This presentation describes various options for implementing RTT with open source VoIP components such as Asterisk. It provides an overview of the protocols involved, such as T140 over RTP in accordance with RFC 4103. It also offers an overview of test clients, interoperability options and potential challenges during an implementation.

12:00-12:30 ♦ Implementing Real Time Text In A VoIP Infrastructure

Henning Westerholt, CEO Gilawa, Cyprus

The addition of real-time text (RTT) functions to voice and video calls is a requirement that has recently been introduced in the European Union and other countries. This presentation describes various options for implementing RTT with open source VoIP components such as Asterisk. It provides an overview of the protocols involved, such as T140 over RTP in accordance with RFC 4103. It also offers an overview of test clients, interoperability options and potential challenges during an implementation.

12:30-13:30 ♦ Lunch Break

13:30-14:00 ♦ APIBAN and Kamailio
Fred Posner, Founder APIBan, LOD, USA
APIBAN is a free service helping you protect your SIP and WebRTC systems from unwanted traffic. Discussion regarding methods of implementing APIBAN data into Kamailio as well as what’s new in APIBAN. From new open source clients to additional features, there is much to talk about!

14:00-14:20 ♦ Seamless Deployment And Fault Tolerance With KDMQ
Viktor Litvinov, Senior Voice Engineer Net2Phone, Spain
A presentation about Kamailio’s Distributed Message Queue (KDMQ) module and its related modules, sharing tips and tricks to help getting started or to optimise existing configurations.

14:20-14:40 ♦ Open Source 5G Mobile Networks Technologies
Elena-Ramona Modroiu, Senior Researcher Technische Universitat Berlin – AV Chair, Germany
A talk about how to build a local 5G stand alone private network using open source only software, presenting also some of the affordable hardware components known to be compatible.

14:40-15:00 ♦ Advanced Vendor Routing Selection Using CGRateS And Kamailio
Dan Bogos, CEO ITSysCom, Germany
Routing SIP calls towards different vendors is a major functionality of a SIP Proxy, hence the importance of the functionality and flexibility which the routing component brings in. In this talk Dan will walk the audience through the RouteS subsystem of CGRateS and its integration with Kamailio via its versatile Evapi module. CGRateS is a battle-tested Open-Source Enterprise Billing Suite with support for various prepaid and postpaid billing modes.

14:40-15:00 ♦ Advanced Vendor Routing Selection Using CGRateS And Kamailio

Dan Bogos, CEO ITSysCom, Germany

Routing SIP calls towards different vendors is a major functionality of a SIP Proxy, hence the importance of the functionality and flexibility which the routing component brings in. In this talk Dan will walk the audience through the RouteS subsystem of CGRateS and its integration with Kamailio via its versatile Evapi module. CGRateS is a battle-tested Open-Source Enterprise Billing Suite with support for various prepaid and postpaid billing modes.

15:00-15:30 ♦ Homer SIPCapture 11
Alexandr Dubovikov, CTO QXIP, Germany
This presentation introduces the release of the latest version of Homer 11, a powerful monitoring and troubleshooting platform for VoIP networks. The session will showcase a broad range of new features and enhancements designed to improve visibility, reliability, and control across modern communication infrastructures. Particular emphasis will be placed on major performance improvements, scalability, and high-throughput processing, enabling efficient monitoring even in large-scale VoIP environments. Attendees will learn how Homer 11 helps operators and engineers gain deeper insights into SIP signaling, call flows, and network behavior, making VoIP operations more transparent, stable, and easier to manage.

15:00-15:30 ♦ Homer SIPCapture 11

Alexandr Dubovikov, CTO QXIP, Germany

This presentation introduces the release of the latest version of Homer 11, a powerful monitoring and troubleshooting platform for VoIP networks. The session will showcase a broad range of new features and enhancements designed to improve visibility, reliability, and control across modern communication infrastructures. Particular emphasis will be placed on major performance improvements, scalability, and high-throughput processing, enabling efficient monitoring even in large-scale VoIP environments. Attendees will learn how Homer 11 helps operators and engineers gain deeper insights into SIP signaling, call flows, and network behavior, making VoIP operations more transparent, stable, and easier to manage.

15:30-16:00 ♦ Coffee Break

16:00-16:30 ♦ The Evolution of SER-Kamailio Project
Open Discussion Panel
An open discussion with the people involved in the planning and development of the project back in early 2000’s, joined by those that witnessed its evolution along 25 years of development. With: Dorgham Sisalem (CEO Frafos, Germany), Jiri Kuthan (iptel.org, Czech Republic), Jan Janak (Columbia University, USA), Dragos Vingarzan (Neat Path Networks, Germany), James Body (Telet Research, UK)

16:30-17:00 ♦ Six Strange Things You Can Do With Realtime Media In Chromium
Tim Panton, \|pipe\|, UK
The web platform keeps expanding its realtime capabilities. This talk is a swift lighthearted jaunt around some of the new and strange things you can do with a chromium based browser. Some of them have practical uses (like replaying the last minute of live video) others are just amusingly odd. Some are at the bleeding edge behind flags – others have been there for a while but not (mis)used correctly yet. Demos will be done for most of them and there will be open source code available.

16:30-17:00 ♦ Six Strange Things You Can Do With Realtime Media In Chromium

Tim Panton, |pipe|, UK

The web platform keeps expanding its realtime capabilities. This talk is a swift lighthearted jaunt around some of the new and strange things you can do with a chromium based browser. Some of them have practical uses (like replaying the last minute of live video) others are just amusingly odd. Some are at the bleeding edge behind flags – others have been there for a while but not (mis)used correctly yet. Demos will be done for most of them and there will be open source code available.

17:00-17:30 ♦ AI Voice Agents over SIP: Operating Pipecat.ai in Real‑Time Carrier Environments
Varum Singh, CTO Daily.co, USA
This talk walks through a concrete architecture of running SIP directly to Pipecat agents with Kamailio and doing tens of thousands of CPS, in some cases ramping up to place 1M SIP calls in an hour.

17:30-18:00 ♦ Your Deployment On Stage – 5 Minutes 5 Slides
Andreas Tarp, Sipgate, Germany
Your chance as a participant to Kamailio World Conference 2026 to show what you are doing in the RTC space, what are your services and products, where and how Kamailio is used. You get 5 minutes to speak on maximum 5 slides and then let the discussions to continue during the breaks and social networking events. Submit proposal via: – https://www.kamailioworld.com/k2026/deployments-on-stage/

18:00 ♦ End Of Day – Closing Remarks

19:00-21:00 ♦ Social Networking Event – Cocktail Party

Friday, May 8, 2026

08:30 ♦ Registration

09:00-09:05 ♦ Welcome

09:05-09:30 ♦ The Future Of Kamailio – Ask Me Anything
Open Discussion Panel – Kamailio Developers
An interactive session allowing the audience to ask any question about using or developing Kamailio. Prepare your questions about scalability, security or anything else you need to build RTC systems with Kamailio. The panelists will be several prominent Kamailio developers and community members, among them Daniel-Constantin Mierla, Victor Seva, Federico Cabiddu, Andreas Granig, Alexandr Dubovikov, Fred Posner, Henning Westerholt.

09:05-09:30 ♦ The Future Of Kamailio – Ask Me Anything

Open Discussion Panel – Kamailio Developers

An interactive session allowing the audience to ask any question about using or developing Kamailio. Prepare your questions about scalability, security or anything else you need to build RTC systems with Kamailio.
The panelists will be several prominent Kamailio developers and community members, among them Daniel-Constantin Mierla, Victor Seva, Federico Cabiddu, Andreas Granig, Alexandr Dubovikov, Fred Posner, Henning Westerholt.

09:30-09:50 ♦ Decentralized Notifications Using Kamailio
Jonathan Kandel, Cellact, Netherlands
Push notifications are the backbone of modern VoIP, without them, incoming calls never ring. But today’s approach funnels every notification through centralized servers that hold your Firebase and APNs credentials, creating single points of failure and custody risk. This talk presents a working architecture that eliminates the centralized notification server entirely. By storing signing credentials inside Oasis Sapphire – a confidential smart contract environment, we move credential custody on-chain where private keys are encrypted at the hardware level and never exposed to any operator. Kamailio triggers the notification flow, but the actual JWT signing happens on-chain, receives a signed JWT assertion, exchanges it for an access token, and delivers the push notification to FCM or APNs, all without a middleman server touching the credentials, thus allowing for multiple service providers with different clients to work with one/many Kamailio servers with/out trust.

09:30-09:50 ♦ Decentralized Notifications Using Kamailio

Jonathan Kandel, Cellact, Netherlands

Push notifications are the backbone of modern VoIP, without them, incoming calls never ring. But today’s approach funnels every notification through centralized servers that hold your Firebase and APNs credentials, creating single points of failure and custody risk. This talk presents a working architecture that eliminates the centralized notification server entirely. By storing signing credentials inside Oasis Sapphire – a confidential smart contract environment, we move credential custody on-chain where private keys are encrypted at the hardware level and never exposed to any operator. Kamailio triggers the notification flow, but the actual JWT signing happens on-chain, receives a signed JWT assertion, exchanges it for an access token, and delivers the push notification to FCM or APNs, all without a middleman server touching the credentials, thus allowing for multiple service providers with different clients to work with one/many Kamailio servers with/out trust.

09:50-10:10 ♦ Damn Vulnerable RTC: A Kamailio, Asterisk And RTPEngine Lab For Security Testing
Sandro Gauci, Founder Enable Security, Germany
Where do you safely practice attacking real VoIP vulnerabilities without breaking production systems? After years of needing a reliable target for testing our security tools, we built the most insecure VoIP platform we could, using the same components this community works with daily. This talk introduces Damn Vulnerable Real-Time Communications (DVRTC), an intentionally vulnerable platform built on Kamailio, Asterisk, rtpengine and other open-source components. Originally developed as our internal demo server for testing security tools, we’re now releasing it to the community. It will cover why we built it and walk through key vulnerabilities we’ve implemented: SIP registration hijacking, authentication bypasses, and RTP injection. I’ll also explain how we configured the stack to be deliberately insecure while remaining realistic. Finally, I’ll demonstrate how the Homer integration lets you visualize attacks in real-time, and discuss how we’re using DVRTC as a VoIP honeypot to research real-world attack patterns.

09:50-10:10 ♦ Damn Vulnerable RTC: A Kamailio, Asterisk And RTPEngine Lab For Security Testing

Sandro Gauci, Founder Enable Security, Germany

Where do you safely practice attacking real VoIP vulnerabilities without breaking production systems? After years of needing a reliable target for testing our security tools, we built the most insecure VoIP platform we could, using the same components this community works with daily.

This talk introduces Damn Vulnerable Real-Time Communications (DVRTC), an intentionally vulnerable platform built on Kamailio, Asterisk, rtpengine and other open-source components. Originally developed as our internal demo server for testing security tools, we’re now releasing it to the community.

It will cover why we built it and walk through key vulnerabilities we’ve implemented: SIP registration hijacking, authentication bypasses, and RTP injection. I’ll also explain how we configured the stack to be deliberately insecure while remaining realistic. Finally, I’ll demonstrate how the Homer integration lets you visualize attacks in real-time, and discuss how we’re using DVRTC as a VoIP honeypot to research real-world attack patterns.

10:10-10:30 ♦ Adding Real-Time AI Pipelines To VoIP/WebRTC Calls With Juturna
Lorenzo Miniero, Founder Janus Gateway, Meetecho, Italy
AI has taken multiple industries by storm, and that has certainly been true for VoIP and WebRTC applications as well. Whether it is for transcriptions, agenting, or any generic AI-based processing, it’s become important to have ways to get access to the real-time streams used in VoIP and WebRTC conversations, and process them in a timely and structured fashion. While there are many ways to use AI with media, many of them work from the assumption that the media to process is already available, with less of a focus on the real-time aspect of it. This is what we tried to address with Juturna, an open source modular framework for building dynamic and real-time pipelines, with different nodes taking care of different aspects of typical AI workflows, all taking into account the real-time nature of the media flowing through the pipeline itself. In this presentation we’ll give an introduction to the Juturna architecture, a few typical workflows (including how we use it in production ourselves for real-time transcriptions of IETF meetings), and explain the changes we made to the Janus WebRTC Server to also allow the SIP plugin to leverage the Juturna functionality and potential for a plethora of interesting use cases.

10:10-10:30 ♦ Adding Real-Time AI Pipelines To VoIP/WebRTC Calls With Juturna

Lorenzo Miniero, Founder Janus Gateway, Meetecho, Italy

AI has taken multiple industries by storm, and that has certainly been true for VoIP and WebRTC applications as well. Whether it is for transcriptions, agenting, or any generic AI-based processing, it’s become important to have ways to get access to the real-time streams used in VoIP and WebRTC conversations, and process them in a timely and structured fashion.

While there are many ways to use AI with media, many of them work from the assumption that the media to process is already available, with less of a focus on the real-time aspect of it. This is what we tried to address with Juturna, an open source modular framework for building dynamic and real-time pipelines, with different nodes taking care of different aspects of typical AI workflows, all taking into account the real-time nature of the media flowing through the pipeline itself.

In this presentation we’ll give an introduction to the Juturna architecture, a few typical workflows (including how we use it in production ourselves for real-time transcriptions of IETF meetings), and explain the changes we made to the Janus WebRTC Server to also allow the SIP plugin to leverage the Juturna functionality and potential for a plethora of interesting use cases.

10:30-11:00 ♦ Coffee Break

11:00-11:30 ♦ VCONs: Turning Your SIP Calls Into Structured, Auditable Conversation Records
Dan Jenkins, CEO Nimble Ape, UK
Every call that flows through your Kamailio infrastructure generates valuable data — but right now, that data is scattered across CDRs, recording files, transcripts, and proprietary analytics platforms that don’t talk to each other. What if every conversation produced a single, structured, signed, portable record that captured everything: who was in the call, what was said, what happened, and what analysis was performed on it? That’s what VCONs (Virtual Conversations) deliver. It’s an IETF standard (now a full working group with draft RFCs published in 2025) that defines a JSON-based container for conversation data — think of it as a vCard, but for conversations. In this talk, I’ll demonstrate integration modules that generate VCONs directly from SIP call data flowing through Kamailio. When a call completes, the module captures the participants, dialog metadata, timestamps, and media references and assembles them into a standards-compliant vCon. That vCon can then be signed for integrity, shipped to a conserver for storage and indexing, and fed into AI analysis pipelines for transcription, sentiment analysis, compliance monitoring, or whatever your business needs.

11:00-11:30 ♦ VCONs: Turning Your SIP Calls Into Structured, Auditable Conversation Records

Dan Jenkins, CEO Nimble Ape, UK

Every call that flows through your Kamailio infrastructure generates valuable data — but right now, that data is scattered across CDRs, recording files, transcripts, and proprietary analytics platforms that don’t talk to each other. What if every conversation produced a single, structured, signed, portable record that captured everything: who was in the call, what was said, what happened, and what analysis was performed on it?

That’s what VCONs (Virtual Conversations) deliver. It’s an IETF standard (now a full working group with draft RFCs published in 2025) that defines a JSON-based container for conversation data — think of it as a vCard, but for conversations.

In this talk, I’ll demonstrate integration modules that generate VCONs directly from SIP call data flowing through Kamailio. When a call completes, the module captures the participants, dialog metadata, timestamps, and media references and assembles them into a standards-compliant vCon. That vCon can then be signed for integrity, shipped to a conserver for storage and indexing, and fed into AI analysis pipelines for transcription, sentiment analysis, compliance monitoring, or whatever your business needs.

11:30-12:00 ♦ FreeSwitch Project Updates
Giacomo Vacca, Signalwire, USA
A presentation about what is new around FreeSwitch project and the Signalwire API features, how AI can be used with them to offer modern real time communication services.

12:00-12:30 ♦ Kamailio Code Challenge – Crack The Logic!
Markus Monka, Kamailio, Sipgate, Germany
You know Kamailio – but can you recognize its logic at a glance? The Kamailio Code Challenge features real-world code snippets submitted by the community. Each snippet includes the Kamailio version and, optionally, the company and product context where it’s used. How it works: * A mysterious Kamailio script appears on the big screen. * The audience has 2 minutes to figure out what it does. * If someone solves it, they win a prize! * If no one cracks the code, the submitter wins instead! Ready for the challenge? Put your Kamailio knowledge to the test and claim the prize! Config snippet submission via the web form available at: – https://www.kamailioworld.com/k2026/kamailio-code-challenge/

12:00-12:30 ♦ Kamailio Code Challenge – Crack The Logic!

Markus Monka, Kamailio, Sipgate, Germany

You know Kamailio – but can you recognize its logic at a glance?
The Kamailio Code Challenge features real-world code snippets submitted by the community. Each snippet includes the Kamailio version and, optionally, the company and product context where it’s used. How it works:
* A mysterious Kamailio script appears on the big screen.
* The audience has 2 minutes to figure out what it does.
* If someone solves it, they win a prize!
* If no one cracks the code, the submitter wins instead!
Ready for the challenge? Put your Kamailio knowledge to the test and claim the prize!
Config snippet submission via the web form available at:
– https://www.kamailioworld.com/k2026/kamailio-code-challenge/

12:30-13:30 ♦ Lunch Break

13:30-14:00 ♦ Winning Business With WhatsApp And Conversational AI
David Duffett, Simwood, UK
Companies in the business of shifting voice minutes from one place to another may have noticed two unhelpful trends: 1. Less minutes than there used to be 2. A downward pressure on the price of shifting those minutes A key reason for the decline of the number of minutes is that they have moved to Over The Top services, like WhatsApp. WhatsApp is pretty much the world’s favourite comms application (take-up varies by geography), but lacks many of the facilities/utilities needed by business – until now! Recent advances in WhatsApp connectivity mean that things like call recording, AI-driven sentiment analysis and many other functions can now be available on calls from and to WhatsApp. What’s more – is that WhatsApp calls are secure, and to or from authenticated users – removing issues around CallerID spoofing, etc. Downwards pressure on the price of voice minutes mean that the smart carrier needs to think of other ways to drive revenue – with value-add services like Conversational AI being a prime example of something that pretty much everyone wants! In this session we’ll take an overview of the situation, suggest a way forward and give examples of where Simwood has forged a path to the future.

13:30-14:00 ♦ Winning Business With WhatsApp And Conversational AI

David Duffett, Simwood, UK

Companies in the business of shifting voice minutes from one place to another may have noticed two unhelpful trends:
1. Less minutes than there used to be
2. A downward pressure on the price of shifting those minutes

A key reason for the decline of the number of minutes is that they have moved to Over The Top services, like WhatsApp.
WhatsApp is pretty much the world’s favourite comms application (take-up varies by geography), but lacks many of the facilities/utilities needed by business – until now! Recent advances in WhatsApp connectivity mean that things like call recording, AI-driven sentiment analysis and many other functions can now be available on calls from and to WhatsApp. What’s more – is that WhatsApp calls are secure, and to or from authenticated users – removing issues around CallerID spoofing, etc.

Downwards pressure on the price of voice minutes mean that the smart carrier needs to think of other ways to drive revenue – with value-add services like Conversational AI being a prime example of something that pretty much everyone wants! In this session we’ll take an overview of the situation, suggest a way forward and give examples of where Simwood has forged a path to the future.

14:00-14:20 ♦ Enabling Rf Billing with Kamailio: From SIP Events to CDRs
Roman Onic, Kontron, Austria
As part of the modernization of operational infrastructures, Kontron Transportation GmbH developed and introduced an MCx train system based on IMS architecture in order to replace existing old analog radio and GSM-R systems. Along with this, the Rf interface was introduced in such networks. The Rf interface allows an IMS Charging Trigger Function (CTF) to issue offline charging events to a Charging Data Function (CDF). Within Kontron Transportation GmbH, this feature implements the CTF at S-CSCF based on Kamailio. In this session an introduction of this Interface along with some use-cases and showing how from SIP Signaling via Diameter over Rf a CDR will be created. On top, for testing purposes an adjustable tool to generate mass CDRs was developed on the S-CSCF. The CDRGen tool is provided on S-CSCF using kamcmd. This will be part of the session as well.

14:00-14:20 ♦ Enabling Rf Billing with Kamailio: From SIP Events to CDRs

Roman Onic, Kontron, Austria

As part of the modernization of operational infrastructures, Kontron Transportation GmbH developed and introduced an MCx train system based on IMS architecture in order to replace existing old analog radio and GSM-R systems.
Along with this, the Rf interface was introduced in such networks. The Rf interface allows an IMS Charging Trigger Function (CTF) to issue offline charging events to a Charging Data Function (CDF).
Within Kontron Transportation GmbH, this feature implements the CTF at S-CSCF based on Kamailio.
In this session an introduction of this Interface along with some use-cases and showing how from SIP Signaling via Diameter over Rf a CDR will be created.
On top, for testing purposes an adjustable tool to generate mass CDRs was developed on the S-CSCF.
The CDRGen tool is provided on S-CSCF using kamcmd. This will be part of the session as well.

14:20-14:40 ♦ To KEMI Or Not To KEMI?
Iurii Gorlichenko, RTC Consultant, Netherlands
KEMI is a powerful tool within the Kamailio ecosystem. However, it is not always obvious when it provides real benefits and when it may instead overcomplicate development and system maintenance. With all the power it offers comes significant responsibility. It requires careful consideration and a well-defined strategy when choosing the right approach for a particular project. I have been working with KEMI since it was introduced, and I also used its predecessor for many years before that. In this session, I would like to highlight when KEMI is something you simply cannot avoid. When it will not solve the task. When it provides the most value as part of a hybrid approach with native configuration I will also discuss the consequences of using KEMI compared to native configuration, and how to effectively manage those trade-offs.

14:20-14:40 ♦ To KEMI Or Not To KEMI?

Iurii Gorlichenko, RTC Consultant, Netherlands

KEMI is a powerful tool within the Kamailio ecosystem. However, it is not always obvious when it provides real benefits and when it may instead overcomplicate development and system maintenance. With all the power it offers comes significant responsibility. It requires careful consideration and a well-defined strategy when choosing the right approach for a particular project. I have been working with KEMI since it was introduced, and I also used its predecessor for many years before that.

In this session, I would like to highlight when KEMI is something you simply cannot avoid. When it will not solve the task. When it provides the most value as part of a hybrid approach with native configuration I will also discuss the consequences of using KEMI compared to native configuration, and how to effectively manage those trade-offs.

14:40-15:00 ♦ The Point Of No Return: NG112/911 Reality Vs. The Standards In 2026
Wolfgang Kampichler, Frequentis, ETSI, Austria
By 2026, Next Generation emergency communications have moved past the theoretical phase and into a “Point of No Return.” With the European Accessibility Act deadline of 2027 fast approaching and the FCC’s Phase 2 NG9-1-1 interoperability requirements in full swing, the industry is no longer just discussing NG – it is migrating under intense pressure. But what happens when NENA i3 and ETSI TS 103 479 standards meet the reality of carrier networks and diverse vendor implementations? This session provides a status report on the global state of emergency services. We will dive into the latest standardization updates from NENA and ETSI, focusing on how Kamailio serves as the backbone for the Emergency Service Routing Proxy (ESRP). Whether you are a provider navigating compliance or a developer curious how SIP saves lives, this talk provides the architectural insights and reality check needed for the future of emergency communications.

14:40-15:00 ♦ The Point Of No Return: NG112/911 Reality Vs. The Standards In 2026

Wolfgang Kampichler, Frequentis, ETSI, Austria

By 2026, Next Generation emergency communications have moved past the theoretical phase and into a “Point of No Return.” With the European Accessibility Act deadline of 2027 fast approaching and the FCC’s Phase 2 NG9-1-1 interoperability requirements in full swing, the industry is no longer just discussing NG – it is migrating under intense pressure. But what happens when NENA i3 and ETSI TS 103 479 standards meet the reality of carrier networks and diverse vendor implementations? This session provides a status report on the global state of emergency services. We will dive into the latest standardization updates from NENA and ETSI, focusing on how Kamailio serves as the backbone for the Emergency Service Routing Proxy (ESRP). Whether you are a provider navigating compliance or a developer curious how SIP saves lives, this talk provides the architectural insights and reality check needed for the future of emergency communications.

15:00-15:30 ♦ Bridging SIP Infrastructure And Conversational AI: Kamailio Telemetry Through MCP And Tooling
Mack Hendricks, Founder dOpenSource, USA
Conversational AI tools such as ChatGPT and Claude has became the new user interface for finding answers. We are seeing more people wanting to use the power of these systems to gather information from multiple external systems in efforts to receive recommendations on the next best action. Many people are looking for these system to become fully autonomous so that they actually implement the action, monitor the action and make a counter action if needed. In this presentation we will discuss how we exposed Kamailio to these tools and the lessons learned.

15:00-15:30 ♦ Bridging SIP Infrastructure And Conversational AI: Kamailio Telemetry Through MCP And Tooling

Mack Hendricks, Founder dOpenSource, USA

Conversational AI tools such as ChatGPT and Claude has became the new user interface for finding answers. We are seeing more people wanting to use the power of these systems to gather information from multiple external systems in efforts to receive recommendations on the next best action. Many people are looking for these system to become fully autonomous so that they actually implement the action, monitor the action and make a counter action if needed. In this presentation we will discuss how we exposed Kamailio to these tools and the lessons learned.

15:30-16:00 ♦ Coffee Break

16:00-16:30 ♦ SIP And RTP Latency And Jitter Optimisations For Kamailio And FreeSwitch
Julien Chavanton, Founder Raven Technologies, Canada
A talk about latency optimised SIP traffic dispatching in Kamailio to dynamically avoid using congested or unresponsive SIP or RTP endpoints, combined with elastic jitter buffer in FreeSWITCH to reduce latency and packet loss of the audio streams.

16:30-17:00 ♦ Kamailio – Transport Layer Scalability – Multi-Processing And Multi-Threading
Daniel-Constantin Mierla, Co-Founder Kamailio, Asipto, Germany
In the recent releases, Kamailio has introduced multi-threading SIP traffic processing capability for UDP and TLS transport layers, as alternatives to the existing multi-process working approach. This talk present the benefits and drawbacks of each model, helping to decide which one is better suitable to meet the particular needs for scalability of the VoIP platform and the interconnected backends or API servers.

16:30-17:00 ♦ Kamailio – Transport Layer Scalability – Multi-Processing And Multi-Threading

Daniel-Constantin Mierla, Co-Founder Kamailio, Asipto, Germany

In the recent releases, Kamailio has introduced multi-threading SIP traffic processing capability for UDP and TLS transport layers, as alternatives to the existing multi-process working approach. This talk present the benefits and drawbacks of each model, helping to decide which one is better suitable to meet the particular needs for scalability of the VoIP platform and the interconnected backends or API servers.

17:00 ♦ End Of Day – Closing Remarks

Thursday, May 7, 2026

Friday, May 8, 2026

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