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Web real-time communications (WebRTC) technology enables real-time communication via browsers. Service providers have deployed WebRTC and are doing trials with the technology to extend their IMS services such as voice over IP (VoIP), voice over LTE (VoLTE), video over LTE (ViLTE) and rich communication services (RCS) to Internet (web) users, hence enabling real-time communication between IMS and web users such as voice, video and chat. The interworking between IMS and web users is made possible by the WebRTC gateway, which translates signaling and transcodes media between these two technology domains.
A WebRTC gateway is typically implemented and deployed in one of the two following variations:
1. Session border controller (SBC) supporting WebRTC signaling (eP-CSCF), media (e-IMS-AGW) and NAT (STUN/TURN/ICE) functions.
2. SBC supporting only WebRTC media (eIMS-AGW) function, and a separate device supporting WebRTC signaling (WebRTC signaling gateway) and STUN/TURN/ICE (NAT) functions.
WebRTC specifications do not define standard web signaling protocol, leaving web application architects and developers to decide. The signaling protocol is a requirement to transmit and exchange user device capabilities such as codec and ICE credentials from one endpoint to another in order to set up the endpoint for a communication session. Despite the flexibility and freedom provided to the application designers to choose the signaling protocol, having no standard signaling protocol introduces challenges in delivering interoperable WebRTC services.
To overcome the interoperatbility challenge, Open Mobile Alliance (OMA) has defined a standardized RESTful HTTP network API that allows a a web-user application (e.g., Javascript running in a WebRTC-enabled browser) to signal video, voice and data session over IP with another communication endpoint in the network. It includes common data types, naming conventions, fault definitions and namespaces.
The httpFlex application emulates RESTful HTTP WebRTC enpoints in compliance with OMA RESTful Network API for WebRTC signaling v1.0 specifications. Its graphical user interface is flexible and easy to use, enabling customers to use out-of-box pre-canned call flows or to define their own custom call flows. It can be used to simulate real-world normal and abnormal calls during various lifecycle phases from lab testing to live network testing. The httpFlex application runs over the QualityAssurer QA-805 test platform.
The QualityAssurer QA-805 test platform–the industry’s leading high-performance and high-capacity platform–supports the comprehensive set of applications and features to test the WebRTC gateway as well as the network end to end.