Voice Calling via Embedded Browser and WebRTC Integration
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Solution Overview
Problem
Connected devices, such as smart speakers and displays, lack SIM card slots and ports, making it difficult to directly connect to voice call carriers for placing voice calls.
Innovation Solution
Implementing an embedded browser that handles low-level protocols and an integration layer using WebRTC and SIP/SRTP to facilitate voice calls through a cloud computing system and voice call carrier interface, enabling voice command-based call initiation without the need for SIM cards.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Reliability
If connected devices use traditional SIM card slots and phone network ports, then direct voice call capability is achieved, but device complexity and manufacturing difficulty increase
Solution Approach 1:
The patent introduces a cloud computing system as an intermediary between the connected device and the voice call carrier. The device communicates with the cloud system via data network, which then interfaces with the carrier's SIP network. This mediator approach allows voice calling without requiring direct SIM card or phone network port connectivity, thereby reducing device complexity while maintaining reliability
Solution Approach 2:
The patent replaces the mechanical/physical connection system (SIM card slots and phone network ports) with a software-based network communication system. By using WebRTC for real-time audio data transmission over IP networks and SIP protocol for carrier integration, the device achieves voice calling capability through software rather than physical hardware interfaces, reducing manufacturing complexity
2Reliability
If connected devices include SIM card slots and phone network ports, then direct carrier connection is enabled, but ease of manufacture decreases
Solution Approach 1:
The cloud computing system serves as a mediator that handles the complex carrier connection protocols and authentication. The connected device only needs standard network connectivity to communicate with the cloud system, which then manages the SIP connection to the voice call carrier. This approach significantly simplifies the manufacturing process as devices don't require specialized SIM card slots or phone network ports
Solution Approach 2:
The patent makes the connected device utilize existing multi-functional network interfaces and processors for voice calling, rather than adding dedicated voice call hardware. The same network interface used for general data communication is leveraged for WebRTC audio streaming and SIP protocol exchange, making the device easier to manufacture by reusing existing components
3Device complexity
If connected devices lack SIM card slots and phone network ports, then device simplicity is maintained, but ability to place voice calls deteriorates
Solution Approach 1:
The cloud computing system acts as a mediator that bridges the gap between the simple connected device and the complex voice call carrier network. The device maintains simplicity by lacking SIM card slots and phone ports, while the cloud intermediary handles all carrier protocol complexities, ensuring reliable voice call capability is achieved
Solution Approach 2:
The patent substitutes the mechanical SIM card insertion/removal system and physical phone network ports with software-based authentication and network protocol communication. The device uses data network connectivity and software credentials to access voice calling services, maintaining hardware simplicity while achieving reliable carrier connection through software solutions
Data Source
AI summary
Systems and techniques are provided for voice calling with a connected device that does not include a SIM card or telephone port. Outgoing audio data may be received at an embedded browser running on a connected device, may be sent using Web Real Time Communications (WebRTC) from the embedded browser to an integration layer running within the embedded browser, and may be sent from the integration layer to a border controller for a voice call carrier over a Session Initiation Protocol (SIP) connection according to Secure Real Time Transport Protocol (SRTP). Incoming audio data may be received at the integration layer from the border controller for the voice call carrier over the SIP connection according to SRTP, may be sent using WebRTC from the integration layer to the embedded browser, and may be sent from the embedded browser to an audio output of the connected device which may output audio.