Evariste Systems is a consultancy and software company specialising in open source-focused VoIP service delivery engineering for Internet Telephony Service Providers (ITSPs) and carriers.
Our solutions are oriented toward wholesale and retail termination/origination vendors, VoIP application service providers, and CLECs. We build and integrate high-performance, cost-effective software platforms for core routing, call accounting, and trunk billing & rating.
We also provide our customers with a variety of technical services related to voice backend engineering and day-to-day operational support, including IP network architecture, vendor selection, capacity planning, protocol analysis and troubleshooting. In addition, our experience in the competitive carrier space has enabled us to play an advisory role on questions of PSTN-side technology and economics.
Based in Atlanta, Georgia, USA, we are an experienced team of software developers, UNIX systems experts and network engineers. We have extensive operational experience in real-world telco environments with a range of open-source and proprietary industrial telephony technology. We look forward to putting our experience to work for you!
Learn about the Canonical SIP Routing Platform, our turn-key SIP trunking engine with advanced least-cost routing (LCR), call accounting, rating and much more!
CSRP was designed with a laser focus on the oft-neglected objectives of SIP trunking and Class 4-style routing. Its solid technical core delivers flexible, high-performance least-cost routing, high availability and security, while its sophisticated business layer has a proven track record of helping customers raise gross margins and improve cash flow.
We offer a range of professional VoIP engineering services to ITSPs, carriers and voice application providers, including:
Customisation and deployment of Kamailio-based SIP routers, SBC replacements and load balancers.
Voice application development with
CDR data interchange, conversion and processing.
VoIP security assessment and DoS prevention.
Cisco VoIP gateway deployment.
SIP interoperability troubleshooting and related expertise.
We offer on-site and remote training on a wide variety of subject matter related to IP telephony.
In particular, we frequently reprise our role as a pedagogical resource to our customers for SIP fundamentals and protocol mechanics, as well as introductory and advanced Kamailio configuration.
We do not follow bland, formulaic curricula consisting of "canned" material. Training is provided only by engineers with direct subject-matter expertise. It is important to us that you get the best value in the market for your training dollar, so we will design an approach that addresses your specific objectives.