WebRTC Services Solution

WebRTC Services Solution

Web Real-Time Communications (WebRTC) enables web browsers to participate in audio, video and data communications, without any kind of additional plug-ins or application downloads. Realtime communication stack is built in browser. So, Using a WebRTC-enabled browser, users can place a call, participate in multi-party video and audio conferencing, and engage in screen sharing collaboration etc.

Although WebRTC enables real-time communications directly from the browser, it does not provide the server side signaling infrastructure needed for discovery, interconnection and session management of WebRTC sessions. Also there is a need to interconnect and interwork the webRTC communications with traditional SIP communication. This is where Sonus WebRTC Services Solution comes to rescue, providing the carrier grade reliable and secure session management infrastructure interconnecting webRTC and traditional SIP based networks. It is complemented by simple SDK for developing web and native IOS, Android mobile applications without worrying about intricacies of underlying browsers and Platforms. Making developing realtime communication enabled web and mobile applications a breeze.

Sonus has addressed the scalability and performance of real-time communications for next-generation networks by developing the Sonus WebRTC Services Solution, enabling web browsers to perform real-time communications by interworking with centralized web servers, applications, WebRTC desktop and mobile clients, and back-end SIP infrastructures. With the introduction of the Sonus WebRTC Services Solution, an enterprise and/or service provider can allow a user to place a call, participate in multi-party video and audio conferencing, engage in screen sharing collaboration, and share files through WebRTC-enabled browsers and mobile applications for real-time communications. Any WebRTC-enabled device can be used to communicate with another WebRTC-enabled device or with other endpoints (SIP, PSTN) via the Sonus WebRTC Gateway.

The Sonus WebRTC Services Solution is inclusive of:

  • Sonus WebRTC Gateway (WRTC) - performs session management and routing of the web-web and web-sip sessions. It maintain registry of webRTC users and performs user authentication. WRTC GW performs web-SIP signaling interworking.
  • Sonus WebRTC Software Development Kit (WRTC SDK) – toolkit to develop real time communication enabled web and mobile applications. Sonus WRTC SDK is available for web (JS), Android, IOS and desktop platforms.
  • Sonus Session Border Controller (SBC) - performs media policing, signaling adaptation, NAT traversal and media interworking to enable communication between WebRTC and SIP endpoints. Customer can use Sonus SBC SWe or SBC 5K/7K based on scaling and deployment requirements. Go to SBC Page.
  • Scalability: enables a network to adapt quickly to changes in the number of supported users in the virtual environment
  • Purpose-built for Cloud and Network Functions Virtualization (NFV) engagements
  • Security: built with secure WebSocket, hardened Connex-IP OS platform, ephemeral TURN credentials, and user authentication using Oauth 2.0, LDAP, and SIP
  • Secure media relay through SBC as encrypted media stream (DTLS-SRTP)
  • Wire rate policing of media streams
  • Management: Centralized cluster element management system (EMS) across the Sonus solution— including WebRTC Gateway, SBC, and PSX—reduces complexity of the end-to-end solution. The functionality includes monitoring, provisioning, statistics, troubleshooting, traps, and counters.
  • Fits seamlessly into the Sonus product portfolio, utilizing the Sonus SBC (5110/5210/7000/SWe), PSX and Insight EMS
  • Sonus SBC interfaces to the WebRTC gateway via SIP for signaling and acts as a WebRTC-to-SIP media gateway, enabling WebRTC users to communicate to any back-end SIP system and PSTN
  • Sonus WebRTC SDK provides support for non-WebRTC enabled browsers (Microsoft IE, Apple Safari) using plug-ins, Microsoft Edge using native ORTC.
  • Sonus WebRTC SDK provides tool kit for developing webRTC enabled applications on IOS, Android platforms.
  • High Availability – the WebRTC Gateway is highly available (Active, Active): in the event of failure of the WebRTC Gateway, another node in the cluster will take over the session
  • Session Rehydration – re-establishes WebSocket and session due to browser refresh, loss of IP connectivity, HTTP server crash
  • Session Mobility – ability to switch the user sessions to new device or network: user switching network (4G -> WiFi)
  • Multi-tenancy – allows for partitioning of access, policy, and user data, as well as customization of policy data as per enterprise needs
  • Multiple Point of presence – A User can login from multiple devices simultaneously and receive calls and messages.
  • User Authentication – allows for user authentication using OAuth-2 from social media networks such as Google and Facebook, LDAP using Enterprise Active Directory, customer application databases, webTokens etc., or SIP registrars
  • Admission Control – administers limits on the enterprise sessions, user sessions, media types, and subscribed applications
  • Support for lawful intercept and blacklisting of misbehaving endpoints

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