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This document describes the architecture and the operational steps to install, operate and modify the Sipwise sip:provider Community Edition (CE).
The first chapter gives a general overview of the CE. It describes what it is, what it contains and who should use it.
The second chapter guides through the installation process of the CE. It lines out the prerequisites and the steps required to install it.
The third chapter describes the steps to configure the CE in order to offer VoIP services to end users.
The fourth chapter shows the different customer self-care interfaces and describes how to configure them.
The fifth chapter describes the billing interface, so you can rate calls and export call detail records.
The sixth chapter describes in detail the steps necessary if you want to start modifying local configuration files to adapt and extend the system.
The seventh chapter guides through the provisioning interface (SOAP and XMLRCP) system.
The eighth chapter describes best practices for production systems regarding system security and backup/restore procedures.
More chapters will be added in the future to document every aspect of the system.
We have set up the spce-user mailing list, where questions are answered on a best-effort basis and discussions can be started with other community users.
If you need professional help setting up and maintaining the CE, send an email to support@sipwise.com.
Sipwise also provides training and commercial support for the platform. Additionally, we offer a migration path to the sip:provider PRO appliance, which is the commercial, carrier-grade version of the CE. If the user base grows on the CE, this will allow operators to migrate seamlessy to a highly available and scalable platform with defined service level agreements, phone support and on-call duty. Please visit www.sipwise.com for more information on commercial offerings.
The CE is a SIP based Open Source Class5 VoIP soft-switch platform providing rich telephony services. It offers a wide range of features to end users (call forwards, voicemail, conferencing, call blocking, click-to-dial, call-lists showing near-realtime accounting information etc.), which can be configured by them using the customer-self-care web interface. For operators, it offers a fully web-based administrative panel, allowing them to configure users, peerings, billing profiles etc., as well as viewing real-time statistics of the system. For thight integration into existing infrastructures, it provides SOAP and XMLRPC APIs.
The CE can be installed in a few steps within a couple of minutes and requires no knowledge about configuration files of specific software components.
Unless other free VoIP software, the CE is not a single application, but a whole software platform, the Sipwise NGCP (Sipwise Next Generation Communication Platform), which is based on Debian GNU/Linux.
Using a highly modular design approach, the NGCP leverages popular open-source software like MySQL, Apache, Catalyst, Kamailio, SEMS, Asterisk etc. as its core building blocks. These blocks are glued together using optimized and proven configurations and work-flows and are complemented by building blocks developed by Sipwise to provide fully-featured and easy to operate VoIP services.
After downloading and starting the installer, it will fetch and install all the required Debian packages from the relevant Debian repositories. The installed applications are then managed by the NGCP Configuration Framework, which allows to change system parameters in a single place, so administrators don’t need to have any knowledge of the dozens of different configuration files of the different packages. This provides a very easy and bullet-proof way of operating, changing and tweaking the otherwise quite complex system.
Once installed and configured, integrated web interfaces are provided for both end users and administrators to use the CE. By using the provided provisioning and billing APIs, it can be integrated tightly into existing OSS/BSS infrastructures to optimize work-flows.
The CE is specifically tailored to companies and engineers trying to start or experiment with a fully-featured SIP based VoIP service without having to go through the steep learning curve of SIP signalling, integrating the different building blocks to make them work together in a reasonable way and implementing the missing components to build a business on top of that.
In the past, creating a business-ready VoIP service included installation and configuration of SIP software like Asterisk, OpenSER, Kamailio etc., which can get quite difficult when it comes to implementing advanced features. It required to implement different web interfaces, billing engines and connectors to existing OSS/BSS infrastructure. These things are now obsolete due to the CE, which covers all these requirements.
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