Betting on WebRTC:
Technology Overview and Architectural Impacts
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Mobile Networks Reference Poster
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WebRTC
Preparing networks for a traffic explosion
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WebRTC


 

Our Solution

  • Simulate millions of  WebRTC  and IMS VoLTE clients Generate and analyze line rate (1G and 10G) RTP, SDES-SRTP and DTLS-SRTP media streams with mix of codecs such as OPUS, VP8, H.264 and G.711
  • Supports up to 1.28 million of concurrent secure signalling and media sessions in real-time per QA-805 platform
  • Validate the end-to-end service including the Network Address Traversal mechanism using ICE and STUN
  • Measure key performance indicators (KPIs) with high resolution accuracy, upto (10 ns) 

 

 

Access to the internet via web browsers has become a fundamental requirement  for both personal and business lives.

More than a billion devices (tablets, smartphones, PCs) already support browsers, and that number is expected to increase to four billion by 2016

MNOs can capitalize on this significant growth by extending their IMS services to internet users and enabling end-to-end communication between IMS and web clients. This is made possible by the convergence of WebRTC and IMS technologies.

WebRTC is a technology that enables real-time communication via internet browsers.  It uses protocols such as HTTP, WebSocket and DTLS-SRTP.

IMS is a technology that delivers real-time communication services to telecom users.  It uses protocols such as SIP, Diameter and RTP.

The WebRTC gateway enables the convergence between IMS and WebRTC and must support critical functions such as interworking between protocols including authentication, policy and charging, and QoS functions.

Since it plays such an important role, it is crucial to characterize the behavior of the WebRTC gateway in the labs before deployment by emulating real-world traffic scenarios that include IMS and WebRTC users.

WebRTC Diagram

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