Aurus now supports CX integrators and BPO contact centers locally in India

We have partnered with a software industry professional DJ Dutta to run our operations in India to support local CX integrators and BPO contact centers, so their directors and engineers can meet with him in person to evaluate, how our video chat meets their customer experience and engagement needs.

“We are quite excited about the Indian market and committed to building our business here.” said Alexander Anoshin, CEO and founder of Aurus. “Since the very first steps in India we have enjoyed a positive feedback from the local customers and partners. The market is very open and extremely interested in modern customer engagement software, so landing in Delhi was not a tough decision.”

All the sales and service activities for India & APAC markets are now headed by DJ Dutta, a known professional in the enterprise collaboration industry, based out of Delhi.

“Indian users are always open to try out new products provided such products enhances overall performance of their organizations.” said DJ. “Key aspects they consider apart from product capabilities are long term commitment of the vendor and support infrastructure before investing in any new product. Aurus has started its India operation with the key objectives of empowering partners in research & sales and building India centric products jointly with the feedback of partners. We have received excellent initial response from some of the large customers and partners and we are confident that this response will increase manifold once they use Aurus products more & more.”

Aurus sales model is totally based on a network of qualified partners, so the primary goals of the dedicated entity in India are to extend the partner network, support them in research and sales and deliver value-added apps to contact center agents and end users.

You’re welcome to contact Aurus-India and DJ at +91 98101 70552.

7/11/2018

An Introduction to WebRTC Analytics

WebRTC logo

Web Real-time Communications (WebRTC) is an open source project created by Google to enable peer-to-peer communication in web browsers and mobile applications through application programming interfaces. It empowers real-time audio, video, and data transfers without the need for plugins or native app installations. With WebRTC, you can make high-quality, real-time communications applications in Chrome, Firefox, Opera, Android, and natively on iOS and Android.

WebRTC was released in 2011, and since then has become more and more prevalent in the real-time communications space. Facebook, Google, Amazon, and many other companies use WebRTC to provide fast, reliable real-time communications.

Why do you Need Real-time Analytics?

Real-time communications is a difficult feature and service to provide. Users expect and require reliable, effective communication in their day-to-day lives. Even more than that, dependable real-time communications are crucial for businesses, especially ones with remote workers.

It is crucial that real-time systems are sufficiently fast and predictably share resources. This extends to real-time communications – it is very important that events and messages happen in real time. This leaves little room for error, especially in the moment. Thus, the need for real-time analytics.

What is WebRTC getStats?

WebRTC traffic is transported over the IP network, which is susceptible to network congestion. This can increase latency and packet loss, since routers may need to drop packets to mitigate congestion. Losing packets can result in poor video and audio quality, which may lower the user experience. To ensure the highest possible quality, WebRTC includes a real-time statistics API: WebRTC getStats.

The real-time statistics API is accessible though the webrtc-internals page directly in the browser, or by using the getStats() API call.

What Does the getStats() API Include?

The getStats() API is structured into four separate components.

  1. Sender Media Capture Statistics: Sender media capture statistics are related to media generation. This includes frame rate, frame size, clock rate of the media source, the name of the codec, and more.
  2. Sender RTP Statistics: Sender RTP statistics are related to the media sender. This includes the packets sent, bytes sent, round-trip time, and more.
  3. Receiver RTP Statistics: Receiver RTP statistics are related to the media receiver. This includes packets received, bytes received, packets discarded, packets lost, jitter, and more.
  4. Receiver Media Render Statistics: Receiver media render statistics are related to the media rendering. This includes frames lost, frames discarded, frames rendered, playout delay, and more.

The Importance of End-to-end Monitoring

At callstats.io, we gather metrics from real-time communications at the endpoints and middleboxes. By collecting data at multiple points, we are able to get a comprehensive picture of the path, and can diagnose the exact origin of the problem.

We look to measure both the network performance of the path between each pair of WebRTC devices and the media performance at each WebRTC device. This covers a slew of metrics related to the playback and rendering of the media streams to give a comprehensive picture. Metrics are aggregated independently for each connection, participant, and conference.

Get Started with WebRTC Monitoring

getStats contains powerful meetrics that can be used in any WebRTC service. To reap the full benefits of the metrics and collect fine-grained time series of a conference requires a significant amount of resources to collect the data, organize the data, and run diagnostics. There are many metrics to monitor, so if you want to roll your own solution, it’s best to start small and choose a few key metrics. Get started today!

Allie Mellen is the technical content marketer for callstats.io, a software-as-a-service company that provides products that measure and manage the performance of real-time media communication.

6/11/2018