Patent classifications
H04M7/0066
Using STIR/SHAKEN ID Headers to Allow Access into VoIP Networks
The subject matter described herein provides systems and techniques for adding an identity (ID) header to IP packets associated with a VoIP call. This ID header may be used to authenticate the source provider/originator of a VoIP call, may be used to traceback to the source provider/originator of the VoIP call, and may be used to create a relationship between the source provider/originator and the destination provider/destination of the VoIP call. Such steps may be performed by a public proxy/platform. The ID header may include a certificate and/or a public encryption key, from a public certificate authority (CA) infrastructure, which assists in authenticating the source provider/originator of the VoIP call. The public proxy/platform may directly route authenticated VoIP calls through a VoIP network towards its destination, bypassing a public switched telephone network (PSTN).
Processing sensitive information over VoIP
This invention relates to a method of processing sensitive information over VoIP. The method provides a method of processing, by a call processor, a media call comprising the steps of: receiving a first signalling stream from a first entity; creating a second signalling stream to a second entity; forwarding signals received from the first signalling stream to the second signalling stream; receiving a third signalling stream from the second entity; creating a fourth signalling stream to the first entity; and forwarding signals received on the third signalling stream to the fourth signalling stream; the first signalling stream containing instructions to set up a media call between the first entity and the second entity such that media is transmitted over a first media stream from the first entity to the second entity and a media is transmitted over a second media stream from the second entity to the first entity without intervention by said call processor.
APPLICATION ROUTING BASED ON USER EXPERIENCE METRICS LEARNED FROM CALL TRANSCRIPTS
In one embodiment, a device obtains call transcripts from an online application. The device detects cues within the call transcripts that are indicative of poor user experience. The device generates, based in part on the cues detected within the call transcripts, a model trained to predict poor user experience from network path telemetry for the online application. The device causes traffic for the online application to be routed along a particular network path, based on a prediction by the model.
PROCESSING SENSITIVE INFORMATION OVER VOIP
This invention relates to a method of processing sensitive information over VoIP. The method provides a method of processing, by a call processor, a media call comprising the steps of: receiving a first signalling stream from a first entity; creating a second signalling stream to a second entity; forwarding signals received from the first signalling stream to the second signalling stream; receiving a third signalling stream from the second entity; creating a fourth signalling stream to the first entity; and forwarding signals received on the third signalling stream to the fourth signalling stream; the first signalling stream containing instructions to set up a media call between the first entity and the second entity such that media is transmitted over a first media stream from the first entity to the second entity and a media is transmitted over a second media stream from the second entity to the first entity without intervention by said call processor.
PROCESSING SENSITIVE INFORMATION OVER VOIP
The invention relates to a method of processing sensitive information over VoIP. The method provides a method of processing, by a call processor, a media call comprising the steps of: receiving a first signalling stream from a first entity; creating a second signalling stream to a second entity; forwarding signals received from the first signalling stream to the second signalling stream; receiving a third signalling stream from the second entity; creating a fourth signalling stream to the first entity; and forwarding signals received on the third signalling stream to the fourth signalling stream; the first signalling stream containing instructions to set up a media call between the first entity and the second entity such that media is transmitted over a first media stream from the first entity to the second entity and a media is transmitted over a second media stream from the second entity to the first entity without intervention by said call processor.
Large volume voice over in internet protocol services for an aircraft
A voice over internet protocol (VoIP) system for an aircraft includes a ground gateway, an aircraft gateway disposed on the aircraft, and a service provider network disposed on the aircraft. The ground gateway is in communication with the aircraft gateway via the service provider network. The aircraft gateway includes a first proxy agent, and the ground gateway includes a second proxy agent. The first proxy agent and the second proxy agent communicate a network packet for a number streams. The network packet includes a header and voice payloads for the streams.
NETWORK EXCEPTION SYSTEMS AND METHODS FOR PACKET-SWITCHED TELEPHONY
Asynchronous and/or synchronous telephony protocol systems and methods may include an asynchronous signaling node (ASN) and/or a call duration time quota from a charging onset to place and complete a call based on a first device call request as received from a first user mobile device on a packet switched network. The asynchronous systems include instructions to automatically modify the telephony address with a prefix and destination address when the first device has insufficient or independent balance or upon a network exception; route the modified call signal to the ASN; and deliver and automatically disconnect the call when the call is completed. The synchronous systems are balance-independent and include instructions to automatically set the call duration time quota upon such exception, and deliver and automatically disconnect the call from the second user telephony device when the call is completed or when the call duration time quota is exceeded.
Communication using communication tokens, such as QR codes
A VoIP communication service using communication tokens, such as QR codes, to identify the receivers can provide callers with the ability to communicate with the receivers using VoIP technology, without the need to pre-registering with the VoIP service provider that hosts the communication. The tokens can be customized by having a profile, which can include conditions imposed by the receivers on the callers. The tokens can be used in a call center to provide VoIP communication services.
DYNAMIC VOICE OVER INTERNET PROTOCOL PROXY FOR NETWORK BANDWIDTH OPTIMIZATION
Examples include receiving, from a VoIP server, identification information of the virtual meeting and information of a plurality of client devices participating in the virtual meeting, associating each client device of the plurality of client devices to a network device in a set of network devices based on the identification information of the virtual meeting and capabilities of the set of network devices, and creating a proxy agent in each network device in the set. The proxy agent is configured to receive, from the VoIP server, VoIP streaming data of the virtual meeting. Additionally, examples include, configuring the proxy agent to replicate and transmit the VoIP streaming data to the plurality of client devices.
SESSION BORDER CONTROLLER (SBC) SYSTEM
A session border controller (SBC) system includes an SBC signaling and media processing module and an SBC signaling access module. The SBC signaling access module is configured to: receive signaling sent by a terminal and forward the signaling to the SBC signaling and media processing module; and when the terminal is disconnected from a network during a call, re-register and reconnect the terminal, and after the terminal is reconnected, receive an information request from the terminal during the call and forward the information request to the SBC signaling and media processing module. The SBC signaling and media processing module is configured to: process the signaling and media streams from the terminal, and when the terminal is disconnected from the network during the call, learn, based on the information request, a transmission link for communication with the terminal after the terminal is reconnected upon the disconnection from the network.