HTTP Intermediary for SIP VoIP Cloud Load Balancing
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Solution Overview
Problem
Current cloud computing platforms are not well-suited for Voice over Internet Protocol (VoIP) applications, particularly those requiring interconnection with the public switched telephone network (PSTN), as SIP-based applications lack access to HTTP-based cloud platform capabilities, leading to barriers in adoption and complex provisioning processes, including issues with robocall abuse.
Innovation Solution
The Realtime Internet Peering for Telephony (RIPT) protocol, which operates over HTTP, enables secure, scalable, and self-provisioned VoIP communications by utilizing HTTP/3, integrating with cloud platforms like AWS, Azure, and Google Cloud, providing secure caller ID, automated provisioning, and built-in security features.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Adaptability or versatility
If SIP-based applications are deployed on cloud platforms, then VoIP interconnection with PSTN is enabled, but the applications cannot access HTTP-based cloud platform capabilities such as load balancing, autoscaling, and security features
Solution Approach 1:
The patent introduces an HTTP intermediary layer that translates SIP signaling into HTTP requests. This intermediary enables SIP-based VoIP applications to access HTTP-based cloud platform capabilities (load balancing, autoscaling, security) without requiring the applications themselves to be HTTP-based, thus resolving the contradiction between adaptability to cloud platforms and deployment complexity
Solution Approach 2:
The patent creates a universal interface layer that allows SIP applications to leverage multiple cloud platform capabilities simultaneously. By translating SIP protocols into HTTP, the system enables a single application deployment to access load balancing, security, scaling, and other HTTP-based services, achieving multi-functionality without increasing application complexity
2Reliability
If SIP servers are deployed on bare metal or VMs with custom-built load balancing and security technologies, then VoIP functionality is achieved, but the barrier to entry increases and provisioning becomes complex
Solution Approach 1:
The patent enables cloud platforms to automatically provision and configure VoIP services through HTTP-based interfaces. The system allows automated self-service deployment where the cloud platform handles load balancing, security configuration, and service provisioning without requiring manual setup of SIP servers on bare metal or VMs, thus improving ease of deployment while maintaining reliability
Solution Approach 2:
The HTTP intermediary layer abstracts away the complexity of SIP server deployment by providing a standardized interface to cloud platform services. This mediator handles the translation and integration automatically, eliminating the need for manual configuration of load balancing and security technologies, thereby improving ease of deployment while maintaining service reliability
3Productivity
If traditional SIP trunking is used, then VoIP communication is established, but provisioning operations are complex and require exchange of static IPs and ports taking weeks
Solution Approach 1:
The patent replaces the manual, mechanical SIP trunking provisioning process (exchange of static IPs and ports) with an automated HTTP-based system. The cloud platform handles provisioning through HTTP APIs, automatically configuring services without manual intervention, thus dramatically improving provisioning speed from weeks to minutes while reducing complexity
Solution Approach 2:
The system enables automated self-service provisioning where the cloud platform automatically configures VoIP services through HTTP interfaces. The platform handles service activation, configuration, and resource allocation automatically without requiring manual exchange of static IPs and ports, achieving rapid provisioning while simplifying the process
4Reliability
If SIP-based applications are deployed without HTTP protocol, then VoIP functionality is achieved, but the applications are vulnerable to robocall abuse and lack built-in security features
Solution Approach 1:
The patent introduces an HTTP intermediary layer that provides security features (including protection against robocall abuse) while maintaining SIP protocol flexibility. The intermediary translates SIP requests into HTTP, enabling the use of HTTP-based security mechanisms without requiring the VoIP applications themselves to change from SIP to HTTP, thus achieving security improvement while preserving protocol adaptability
Data Source
AI summary
Systems and methods are described herein for providing a Voice over Internet Protocol (VoIP) call. In an embodiment, a load balancing processor receives a re-initiated HTTP request from a client processor upon detection that an initial call server is no longer active, and sends the re-initiated HTTP request to a second call server. The second server generates updated call resource information that identifies the second server as the new server resource for the call, and sends the updated call resource information over the IP network to the client processor. Subsequent HTTP requests from the client processor for sending and receiving signaling and media data for the call are received at the second server using the updated call resource information.