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  1. 1. Workshop schedule, 3rd of August 2015

OpenSIPS Workshop , 3rd of August 2015 , InterContinental Chicago, Chicago, IL, USA.

1. Workshop schedule, 3rd of August 2015

09:00 - 10:00 Registration and Coffee

10:00 - 10:40 Răzvan Crainea - OpenSIPS project - Introducing OpenSIPS 2.1
» 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.
» The "high-level" features become richer with the 2.1 release - the ability to do Fraud Detection, SIP Compression, Advanced Topology Hiding and Emergency Call comes as to complement the "low-level" features as WebSocket support or Asynchronous DB queries.

10:40 - 11:20 Dan Christian Bogos - ITsysCOM - Real-time charging with LCR for OpenSIPS Async (workshop)
» In an era when customers are increasingly demanding quality from their VoIP networks, it is vital for the operators to have the ability of monitoring their traffic statistics and being able to react on network issues in real-time. This becomes even more of a challenge when dealing with large call volume peaks.
» In this talk, Dan will guide the audience through advanced traffic stats and costs monitoring facilities built within CGRateS charging system, touching concepts such as filtered monitor instances with granular time and/or item count windows and configurable reactions.

11:20 - 12:00 Vlad Paiu - OpenSIPS Project - Async Operations with OpenSIPS 2.1 (workshop)
» 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. The REST client in OpenSIPS can also operate in asynchronous mode, allowing an efficient way to query external services like LRN.

12:00 - 13:30 Lunch break

13:30 - 14:10 Razvan Crainea - OpenSIPS Project - OpenSIPS 2.1 as edge proxy (I) - Compression and TH (workshop)
» OpenSIPS 2.1 is a powerful, but flexible Edge Proxy. It can front any kind of SIP core (any as service and any as implementation) in order to secure or enhance the core service.
» This first part of the workshop focuses on configuring OpenSIPS as front SIP Filter and Load Balancer. With 2.1 release, OpenSIPS can take care of more security and call handling aspects as ever. The workshop will show you how to do transparent but efficient fronting and how to do traffic balancing in a SIP wise manner (in conjunction with advanced services as Attended Call Transfer or Conferencing).
» Further we'll look into more complex capabilities of an Edge proxy. Mainly about how do handle the SIP traffic in a different way, on the inbound and outbound direction. The workshop shows how to take advantage of the compression ability of OpenSIPS 2.1 in order to minimize the SIP traffic and avoid network issues. From topology perspective, OpenSIPS 2.1 can do hiding on both levels, network and SIP - minimizing the information about the SIP core network is the best way to prevent the external attacks or inter-operability issues.

14:10 - 14:50 Razvan Crainea - OpenSIPS Project - OpenSIPS 2.1 as edge proxy (II) - WebSockets and TLS (workshop)
» The second part of the workshop focuses on SIP transport protocols - how to address related aspects with OpenSIPS 2.1. Starting from how to achieve bridging between different types of networks, the workshop will guide you all the way to doing SIP protocol conversion - providing TCP, TLS and WebSockets to end users, while converting everything to UDP in the core network.

14:50 - 15:30 Vlad Paiu - OpenSIPS Project - Fraud Detection with OpenSIPS 2.1 (workshop)
» Fraud is a major problem nowadays and it is more complex as it depends on the end-user/end-device security level. And all the VoIP providers are looking forward for ways to protect their users and avoid the blame.
» This tutorial uses OpenSIPS 2.1 to detect potential fraud attempts by inspecting the traffic in realtime and correlating it with the calling patterns of the end-user. Several calling parameters are monitored and inspected in order to identify (in a per-user basis) users who were potentially hijacked and scammed.

15:30 - 16:00 Coffee break

16:00 - 16:40 Alex Goulis - RateTel - Using OpenSIPS as your registrar to deliver load balanced class 5 features with Freeswitch (workshop)
» 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.

16:40 - 17:20 Eric Tamme - OnSIP - Federated SIP - Building SIP servers the right way
» You can send email between domains, but you can't make a SIP call to a different domain? That is just broken, and not how SIP was intended to be implemented. Learn how to build a federated SIP service, the key components of SIP federation, and some advanced uses of OpenSIPS and RTPEngine such as WebSockets, DTLS-SRTP <-> RTPbridging along the way.

17:20 - 18:00 Pete Kelly - Sourcevox - OpenSIPS and FreeSwitch integration - how to use OpenSIPS b2bua and FreeSWITCH to build a calling card solution (workshop)
» How you can use FreeSWITCH to complement OpenSIPS in terms of transcoding audio, dealing with Voicemail, playing prompts, playing an IVR, transferring a call or dispatching.

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