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Testing WebRTC: Mitigate Live Network Failures

  • febrero 16, 2015

Kader Khan

Director, Simulators Business Unit

Kader Khan is Director of Service Assurance, Systems and Services division at EXFO. He joined the company in 2005, dividing his 15 years between senior roles in R&D Management, Site Management, Product Management, Business Development, Partnership, and Alliances. He is a passionate leader, an expert in intelligent technologies, loves winning deals, and seeing customers thrive.

Kader has over 22 years of experience in the information communication sector, specializing in wireless technology. He has a Bachelor of Engineering Degree in computer science and has completed various technical and management certification courses.

Service providers are adopting web real-time communications (WebRTC) technology that provides many new monetizing opportunities to extend existing communication services (voice, video and data), and to create and augment wholly new and robust real-time services by leveraging existing investments on high-speed networks, such as long-term evolution (LTE) and IP multimedia subsystem (IMS).

Although WebRTC creates many great opportunities, service providers need to be careful what they wish for—customer expectations in today’s highly competitive environment leave little room for failure when it comes to the delivery of such innovative technologies and services. Operators can only accomplish their goals by offering uncompromising performance. This is best achieved by understanding the best practices for testing WebRTC networks, devices and services, as such knowledge will help ensure reliable network infrastructure in labs, in turn mitigating live network failures and avoiding customer dissatisfaction. 

In this webinar, we will deep-dive into the bits and bytes of testing end-to-end WebRTC services, as well as WebRTC Gateway (SBC), which is the focal point of convergence between telecoms and the Internet world. This webinar will also provide insight into signaling protocols such as SipOverWebsocket, JSONoverWebsocket and NAT, in addition to media protocols such as ICE, STUN and DTLS-SRTP.

Who Should Attend

Network architects, designers, lab managers and testers working for service providers, large enterprises and manufacturers, as well as members of the press and analysts who are looking to understand the essentials of testing WebRTC in labs for stellar live network and service performance. 

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