Cleanly coded website from customer selfcare interface to the sophisticated admin portal.
The following are just a few of the ways the portal helps you manage your VoIP business:
- Manage customer features, DID numbers, credit limits, rate tables and more
- Create SIP Trunking accounts
- Add lines and configure free IP to IP calling
- Assign termination credit and call restrictions
- Order DID numbers, SIP termination and all of our other services for real-time provisioning
- Customize your SIP Proxy
- Create/edit rate tables
- View CDRs
- Customize product names, voice prompts and dial plan
- Check international rates and ASR
- View your accounting statement of account
- Make manual credit card payments and set default payment methods
- View invoices
- Open help desk ticket
- Set automatic recharge thresholds
- Manage DID numbers
Custom configurations for OpenSIPS
OpenSIPS (Open SIP Server) is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. OpenSIPS, as a SIP server, is the core component of any SIP-based VoIP solution. With a very flexible and customizable routing engine, OpenSIPS unifies voice, video, IM and presence services in a highly efficient way, thanks to its scalable (modular) design.
What OpenSIPS has to offer, comes in a reliable and high-performance flavour - OpenSIPS is one of the fastest SIP servers, with a throughput that confirms it as a solution up to enterprise or carrier-grade class.
OpenSIPS has to offer many important and interesting features. To mention some of the most important ones:
- SIP registrar server
- SIP router / proxy (lcr, dynamic routing)
- SIP load-balancer or dispatcher
- SIP front end for gateways/asterisk
- SIP NAT traversal unit
- SIP application server
- SIP to SMS gateway (bidirectional)
- SIP IM server (chat and end-2-end IM)
Do it all with using WebRTC
Imagine a world where your phone, TV and computer could all communicate on a common platform. Imagine it was easy to add video chat to your web application. That's the vision of WebRTC.
Want to try it out? WebRTC is available now in the customer selfcare portal, as a module.
One of the last major challenges for the web is to enable human communication via voice and video: Real Time Communication, RTC for short. RTC should be as natural in a web application as entering text in a text input. Without it, we're limited in our ability to innovate and develop new ways for people to interact.
Historically, RTC has been corporate and complex, requiring expensive audio and video technologies to be licensed or developed in house. Integrating RTC technology with existing content, data and services has been difficult and time consuming, particularly on the web.
Gmail video chat became popular in 2008, and in 2011 Google introduced Hangouts, which use the Google Talk service (as does Gmail). Google bought GIPS, a company which had developed many components required for RTC, such as codecs and echo cancellation techniques. Google open sourced the technologies developed by GIPS and engaged with relevant standards bodies at the IETF and W3C to ensure industry consensus. In May 2011, Ericsson built the first implementation of WebRTC
WebRTC has now implemented open standards for real-time, plugin-free video, audio and data communication.