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Enhancing Real-Time Communication with WebRTC: A Detailed Overview

Explore the capabilities of WebRTC for setting up media directly between browsers, handling NAT/FW traversal, and overcoming firewall challenges. Learn how WebRTC integrates with SIP PBX for enterprise communication solutions.

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Enhancing Real-Time Communication with WebRTC: A Detailed Overview

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  1. What WebRTC Does NOT Do: • “No Numbers” No rendezvous – “no addressing” at all. Not like SIP • ------------ • More islands? Yes, but it is adding high quality real-time communication where we already are in contact. What WebRTC Does: • Sets up media directly between browsers (SDP/RTP like SIP) – typically on same web application. • “Handles” NAT/FW traversal (ICE, STUN, TURN) – fooling firewalls (like Skype). Voice Video Data “For free!”

  2. WebRTC Like All Real-Time Communication Protocols has a NAT/Firewall Traversal Problem • Firewalls do not allow unknown incoming traffic and media is a “surprise” (just like SIP) • SBCs are Firewalls that know SIP and take it into the LAN, but WebRTC prescribes ICE/STUN/TURN to fool the firewall to let the RTC traffic through (similar to Skype.) • Web sockets, WS/WSS, often used to hold the signaling channel open • There are issues… • Getting through • Quality signaling media media Company Web Server Company Web Server STUN TURN SERVER WS/WSS ICE LAN LAN

  3. The TURN Server IN the Firewall Fixes Traversal, Quality and can Measure Usage: Q-TURN in the Firewall or an “EW-SBC” • A novel Ingate view: • Knock-knock; Give my media a Quality Pipe • Regard ICE as a request for real-time traffic through the Firewall. Interpret the STUN & TURN signals in the Firewall • Have the STUN/TURN server functionality IN the Firewall and setup the media flows under control • Security is back in the right place - The firewall is in charge of what is traversing • The Enterprise firewall can still be restrictive Q-TURN • Q-TURN Enables QoS and More: • Prioritization and Traffic Shaping • Diffserve or RVSP QoS over the Net • Authentication (in STUN and TURN) • Accounting (usage of this pipe)

  4. The WebRTC & SIP PBX Companion Adding WebRTC capabilities to the enterprise PBX / UC-solution This is for an enhanced Ingate platform, running on existing HW, on virtual machines and even ported on to embedded CPEs. It is considered as a PBX vendor OEM-product and as service provider deployed access device. media Company Web Server Company Web Server WS media LAN LAN SIP

  5. This is in the Companion SIP Trunking E-SBC Web Server WebRTC Gateway Q-TURN Web Server

  6. WebRTC Hooks to the Ingate SIP Proxy E-SBC The WebRTC traffic comes as (i) SIP from the browser (which has got its SIP client code from some webpage) OR (ii) according to the simple Ingate RTCgate protocol (Where the SIP UA, the client, is in the Ingate product). In both cases the transport is via websockets (WS or WSS). All Ingate SIP and SBC features becomes available. The WebRTC client Java Script code may be on any web server (also in the Ingate). The server executing the RTCgate protocol runs on the Ingate. There is more than just SIP, e.g. PBX extensions available through the RTCgate protocol. With Ingate’s Q-TURN (more about that later), important features such as Security, QoS and Accounting (Billing) becomes available also for WebRTC end-to-end (not converting to SIP).

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