OpenSIPS Summit , 21st of October 2014 , Suncoast Hotel and Casino, Las Vegas, NV, USA.
Conference schedule for 21st of October 2014 :
08:00 - 09:00 Registration and Coffee
09:00 - 10:00
Keynotes - Bogdan-Andrei Iancu - Founder OpenSIPS project
» OpenSIPS 1.11 new LTS release - what are the new features
» OpenSIPS 2.1.0 - building up a new state-of-the-art architecture
» Reaching new scalability limits - 60K CPS with 40% CPU ??
10:00 - 11:00
Ali Pey - Sr. Software Engineer Architect - opensips.cfg
» This is to review best practices for OpenSIPS deployment, configuration and scripting. No matter if you are an OpenSIPS beginner or an expert, this will be a chance to ask questions and/or share your experiences as there are some OpenSIPS experts and core developers all in the same room.
» I will walk through opensips.cfg and basic principles of using a proxy server. Will discuss best practices for Network design, redundancy, NAT and OpenSIPS configuration.
11:00 - 12:00
Flavio Goncalves - SIPPulse - Using OpenSIPS to protect an Asterisk Farm
» IP-PBXs are great, but they were created with PBX features in mind. By just configuring a PBX it is not possible to prevent attacks such as SIP flooding, RTP flooding, RTP injection, Fuzzy and Toll fraud. OpenSIPS is an advanced SIP proxy capable to handle low-level parts of SIP. Positioned as an outbound proxy or b2bua, it can protect a farm of PBXs in an effective and affordable way. Learn how!
12:00 - 13:30 Lunch break
13:30 - 14:30
Pete Kelly - Sourcevox - Solving Business Problems with OpenSIPs: Least Cost Routing
» In addition to standard modules, OpenSIPs comes with a powerful "turing complete" scripting language which truly allows you to do some remarkable things when routing a simple SIP request.
» In this example Least Cost Routing is explored. Using the OpenSIPs modules and scripting language, this presentation explores how a business definition of Least Cost Routing can be easily achieved with a combination of OpenSIPs modules and the scripting language.
14:30 - 15:30
Vlad Paiu - OpenSIPS Project - WORKSHOP - OpenSIPS, a service enabler for Asterisk
» By using OpenSIPS as a front-end for the Asterisk-based system, additional/advanced SIP services can be enabled for the end-users. OpenSIPS can act as an enabler for SIP SIMPLE (presence and IM), XCAP, webRTC, TLS support, Parallel Registration, IRC-like chatting and other end-user oriented services.
» Aside the end-user service, OpenSIPS can address provider oriented services like LCR and Gateway failover for the outbound traffic, LNP and CNAME dipping, etc.
15:30 - 16:30
Razvan Crainea - OpenSIPS Project - WORKSHOP - Scaling Asterisk with OpenSIPS
» There is a huge number of successful Asterisk-based platforms which need to grow beyond the "one-instance" stage. To grow/scale as capacity or as geographical coverage. OpenSIPS provides the solution for such platform - the typical OpenSIPS sandwich consists of a pool of Asterisk servers between two OpenSIPS instances, on inbound and outbound.
» On the inbound side, OpenSIPS can solve the problem of doing SIP wise balancing and redundancy for the Asterisk cluster, to ensure traffic validation and shaping. Aside complex authentication mechanisms (as digest, ldap, IP based), the fronting OpenSIPS is responsible for provide fraud detection and other various security tools.
» On the outbound side, OpenSIPS takes care of complex routing logics to PSTN carriers / GW, LCR, CNAME or LNP dipping. Traffic to carriers can be CPS/CC controlled and accounted. The latest OpenSIPS version is able to provide Quality based Routing towards carriers.
16:30 - 17:00 Coffee break
17:00 - 18:00
Bogdan-Andrei Iancu - OpenSIPS Project - WORKSHOP - OpenSIPS Call Center with Asterisk
» Even without having media capabilities, OpenSIPS can provide a high performance and scalable call queuing engine - hundreds of queues, thousands of agents, thousands of queued calls.
» Asterisk is required in combination with OpenSIPS in order to provide media server oriented services - front-end IVRs for selecting queue, announcement playback and queuing playback. Considering that geo-distributed call centers (with unique queue on OpenSIPS), Asterisk can be even more used as a local (to agent) class V PBX for advanced services like call barging, call transfer, call listening and other call-center specific features.