OpenSIPS Summit , 9th-10th of November 2015 , The Contemporary Austin Jones Center, Austin, TX, USA.
1. Conference schedule, Monday, 9th of November 2015
08:00 - 10:00 Registration and Coffee
10:00 - 11:30 Bogdan-Andrei Iancu - OpenSIPS project - OpenSIPS now and next
» OpenSIPS 2.1 is one of the most exciting release in the last years. It brings radical "under-the-hood" changes aiming to make OpenSIPS a top SIP server, in terms of scalability, reliability and performance.
» What are the plans for 2.2 and 3.0 versions, the new set of features and the new set of radical changes for OpenSIPS.
11:30 - 12:15 Jonas Borjesson - Twilio - In the Trenches of a Globally Spanning SIP Network
» Designing, implementing, deploying and running a globally spanning SIP network is both challenging and fun, but some days things just don't go as planned. This talk is about those days. The days spent fire-fighting and the lessons learned. In particular this talk will look into areas such as registration floods, when to failover, when not to failover, long-lived connections, keep-alive traffic, re-transmission issues, loops and more.
12:15 - 13:45 Lunch break
13:45 - 14:30 Trevor Francis - 46 Labs - Opensips 2.1 at Ten Thousand Calls Per Second
» Adopting new versions of any software, regardless of its level of maturity, have its benefits and potential pitfalls. Opensips 2.1 is certainly no exception. This talk is designed to cover the performance of Opensips 2.1 in production load at 10k CPS. Highlights will be made on architectural considerations, metrics that need to be monitored and key new features that extend the reliability of 2.1 at scale.
15:00 - 15:30 Pareshkumar Lukka - Crexendo - Scaling Up With OpenSIPS
» I will discuss how we use OpenSIPS at Crexendo to achieve the scalability, performance and high-availability for our communication platform by using certain things from OpenSIPS and offloading the rest to the media/app servers behind it.
15:30 - 16:00 Coffee break
16:00 - 16:45 Liviu Chircu - OpenSIPS Project - Mitigating SIP security threats with OpenSIPS (workshop)
» Enhance the security of your VoIP platform by taking some preventive measures! The workshop brings insights into common SIP attack vectors, along with mitigation solutions for each of them. Topics include: replay attacks, malicious domains, fraud detection. Practical examples will mostly consist in OpenSIPS 2.1+ scripting
16:45 - 17:30 Vlad Paiu - OpenSIPS project - Asynchronous scripting with 2.1 ( worskhop )
» Starting with the 2.1 release, OpenSIPS provides the ability to perform script I/O operations in an asynchronous way. What does this mean? It means no more blocking and waiting after queries to external entities, like databases, HTTP servers or scripts. Such a non-blocking operation mode eliminates the risks of having the service blocked due to different external queries; while waiting for responses, OpenSIPS can handle other operations or traffic, leading to a major improvement in service availability and resource usage.
» This workshop focuses on how to perform database (MySQL) queries from script by using the asynchronous support. As the DB interaction is a must for all VoIP services, we will see how to do user authentication or how to fetch user profile with no blocking/delay penalties.
17:30 - 18:00 Ovidiu Sas - VoIP Embedded - Shared Call Appearance
» OpenSIPS as a front end proxy for asterisk, handling shared call appearance in an enterprise deployment.
» This presentation will cover the architecture and OpenSIPS implementation.
2. OpenSIPS Social Event, 9th of November 2015
3. Conference schedule, Tuesday, 10th of November 2015
10:00 - 10:45 Maksym Sobolev & Jev Björsell - Sippy Soft - Scaling rtpproxy
» A look at scaling rtpproxy, past, present, future.
10:45 - 11:30 Eric Tamme - onSIP - KwikyKonf, a practical example of implementing WebRTC with OpenSIPS
» Learn about the core elements of WebRTC as it relates to OpenSIPS
* Understand the browser requrements: ICE, STUN, DTLS, websockets
* Build a sip registrar and proxy that supports WebRTC
* Integrate media relaying and legacy interop with RTPengine
* Use SIP.js to create a multi party video chat
11:30 - 12:15 Adrien Laurent / Stas Kobzar - Modulis - Building multi-tenant VoIP platform
» Using OpenSIPS as a core of multi-tenant VoIP platform with automated provisioning. Integration with other VoIP protocols like SCCP and UNIStim and seamless replacement of older Nortel or Cisco systems with a minimum of changes in infrastructure and reducing transition costs.
12:15 - 13:45 Lunch break
13:45 - 14:30 Razvan Crainea - OpenSIPS Project - Troubleshooting and Tuning your VoIP services with OpenSIPS 2.1 (workshop)
» Find out how you can offer the best quality VoIP services to your customers by using OpenSIPS 2.1. This workshop will consist of a set of techniques to troubleshoot your VoIP platform in order to determine its limitations and tune it accordingly to increase its overall performance. The entire workshop will focus on the Summit's star, OpenSIPS 2.1
14:30 - 15:15 Flavio Goncalves - SIP Pulse - Quality Routing
» One of the main tasks of a VoIP provider is to select terminations based on quality and cost. Depending on the number of vendors, this task can become difficult and burdensome. Quality routing takes out the guessing from the job of evaluating the quality of VoIP vendors. This presentation will show you how to use OpenSIPS to select the best route based on cost and quality.
15:15 - 16:00 Joseph Jackson - Homer Sipcapture project - SIP Troubleshooting with Homer & Friends (Workshop)
» This workshop focuses on tools, techniques and ideas useful when capturing, collecting and analyzing VoIP signaling and media scenarios for the purpose of troubleshooting and investigation. Leveraging experience gathered assisting operators and vendors small and BIG, the SIP Troubleshooting workshop covers and describes a range of approaches and solutions mostly driven by open-source projects such as SIPCapture's own Homer, Captagent, sipgrep, nprobe/ntopng and more.
16:00 - 16:30 Coffee break
16:30 - 17:00 Alex Goulis - Ratetel - Using OpenSIPS as your registrar to deliver load balanced class 5 features with Freeswitch
» This workshop will look at a methodology in implementing OpenSIPS as the registrar on your network to provide class 5 features to your registered users using a load balanced cluster of Freeswitch servers. Participants will examine methods to deliver basic pbx features such as voicemail , hunt groups, ivr, and conferences to registered users in the OpenSIPS server. We'll review using mysql to store our realtime configurations for both OpenSIPS and Freeswitch using Mod_xml_curl. In closing, we'll discuss some views on making the configuration multi-tenant.
17:00 - 18:00 Open Discussions
» Asking and answering tough and uncomfortable questions on OpenSIPS - a two ways dialog with the participants on why OpenSIPS, what you like and what you hate on OpenSIPS, what you expect from OpenSIPS.
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