WebRTC to SIP Gateway Protocol Conversion
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Solution Overview
Problem
Next Generation Networks (NGN) or Internet Protocol Multimedia Subsystem (IMS) networks are unable to establish real-time communication with terminals supporting WebRTC technology, as they typically rely on conventional SIP software and hardware, limiting interoperability with WebRTC-enabled browsers.
Innovation Solution
A method and gateway that establishes a Websocket link between a browser and a NGN/IMS network, converting WebRTC protocol-related media parameters in Session Initiation Protocol (SIP) packets to facilitate communication, using a gateway that includes a Websocket converter, SIP stack, and media server to translate between WebRTC and SIP protocols, enabling real-time communication between WebRTC browsers and SIP-based NGN/IMS networks.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Adaptability or versatility
If conventional SIP software and hardware terminals are used in NGN/IMS networks, then the network can maintain its existing architecture and operations, but it cannot establish real-time communication with WebRTC-enabled browsers
Solution Approach 1:
The patent introduces a gateway as an intermediary device between WebRTC browsers and the SIP-based NGN/IMS network. The gateway contains protocol conversion modules that translate WebRTC signaling and media protocols into SIP protocols, enabling communication without modifying the core network architecture. This mediator approach resolves the contradiction by adding compatibility functionality while preserving the existing network structure.
2Adaptability or versatility
If WebRTC protocol conversion is implemented in the gateway, then real-time communication between browsers and NGN/IMS networks is enabled, but the gateway complexity and processing requirements increase
Solution Approach 1:
The gateway is segmented into distinct functional modules: signaling conversion module for protocol translation, media conversion module for codec adaptation, and control module for call management. This segmentation allows each module to handle specific conversion tasks independently, reducing overall gateway complexity while enabling comprehensive protocol interoperability between WebRTC and SIP systems.
3Adaptability or versatility
If protocol conversion is performed for all SIP signaling data, then complete interoperability is achieved, but processing time and computational resources increase
Solution Approach 1:
The gateway implements selective protocol conversion by identifying and converting only the essential signaling elements required for call setup and maintenance (such as INVITE, ACK, and BYE messages), while allowing non-critical signaling data to pass through with minimal processing. This partial action approach achieves sufficient interoperability for real-time communication while reducing overall processing time and computational overhead.
Data Source
AI summary
The embodiments of the disclosure provide a method and gateway for communication between a browser and a telecommunication network. The method is applied to the gateway, and includes that: a first link with the browser is established, and a second link with a NGN or IMS network is established; first data from the browser is received on the first link, the first data is converted to obtain second data when the first data is call related Session Initiation Protocol (SIP) signaling, and the second data is sent to the NGN or IMS network on the second link; and third data from the NGN or IMS network is received on the second link, and when the third data is call related SIP signaling, and fourth data obtained by converting the third data is transparently transmitted to the browser on the first link. Fusion of an Internet technology and a communication technology is implemented.