Abstract:
IP telephony communications are conducted by sending both audio data produced by a CODEC that represents received spoken audio input, and a textual representation of the spoken audio input. A receiving device utilizes the textual representation of the spoken audio input to help recreate the spoken audio input when a portion of the CODEC data is missing. The textual representation can be generated by a speech-to-text function. Alternatively, the textual representation can be a notation of extracted phonemes.
Abstract:
When a voice over Internet protocol (VOIP) telephone call is being conducted by a mobile telephony device, measurements of at least one condition that exists for the mobile telephone device during the VOIP telephone call are taken during the duration of the telephone call. The measurements could be taken periodically as the VOIP telephone call progresses. The measured condition is one that could affect the perceived quality of the VOIP telephone call. The measurements of the at least one condition are recorded against the telephone call for later use and analysis. The recorded information may be analyzed to determine how to modify a setting of the mobile telephony device to improve the quality of VOIP telephone calls conducted with the mobile telephony device.
Abstract:
An Internet protocol (IP) telephony system terminates calls to certain groups of telephone numbers via multiple different telephony carriers. The IP telephony system will discontinue using a telephony carrier to terminate calls if the quality provided by the carrier falls below a threshold level. The IP telephony system includes a quality monitoring unit that determines when a particular group of telephone numbers are intrinsically impaired, such that no carrier could provide high quality when terminating calls to those numbers. In these circumstances, the IP telephony system adjusts the quality threshold to which a carrier's quality is compared when completing calls to the impaired numbers to account for the intrinsic impairment of the telephone numbers.
Abstract:
A method and system for integrating telecommunication session output with one or more applications are provided herein. The method for integrating telecommunication session output with one or more applications includes communicating with one or more second devices in a telecommunication session using a first application disposed on a first device; tracking attributes associated with the telecommunication session; recording at least a portion of the telecommunication session on the first device to produce a session recording; and storing the attributes and at least one of the session recording, or a text transcription of the session recording in a second application on the first device.
Abstract:
A system and method is disclosed herein for providing a unidirectional outage bypass for outgoing communications from a session initiation protocol (SIP) device in a hosted Voice-over Internet Protocol (VoIP) private branch exchange (PBX) system. An outage monitoring system is in communication with both a bypass configuration system and one or more bypass enablers that act at the direction of the bypass configuration system. The outage monitoring system detects outages and overloads, as well as, network failures between network components, the VoIP PBX, Client Devices and the public switched telephone network (PSTN). The bypass configuration system, in response to a detection of an outage, determines error-handling procedures for the unidirectional outage bypass system based on bypass configuration data. The bypass enablers forward outgoing communications between a client device and the PSTN, in accordance the error-handling procedures, by bypassing components that are currently experiencing failures or overloads.
Abstract:
Systems and methods of preventing an Internet service provider from identifying a stream of data packets as carrying a voice over Internet protocol telephony communication can make use of encryption techniques to prevent the Internet service provider from examining the content of the data packets. Also, multiple communications channels may be established between a telephony device and elements of an IP telephony system. A stream of data packets bearing the media of an IP telephony communication is then separated into sub-streams, and each sub-stream is sent through a different one of the communications channels. This prevents an Internet service provider from identifying a stream of data packets as bearing the media of an IP telephony communication based on a pattern in the data traffic.
Abstract:
When IP telephony devices that make use of an Internet protocol (IP) based private branch exchange (PBX) service provider resister with the PBX service provider, they furnish local area network address information that indicates how the telephony devices can be reached directly on the local area network to which they are connected. This information is provided to other telephony devices within the same business or organization. As a result, the telephony devices within a business or organization can contact one another directly to setup and conduct telephony communications over a local area network, without the need for such telephony communications to pass over a public data network, or though assets of the IP based PBX service provider.
Abstract:
Methods and systems for intelligently terminating calls are provided herein. In some embodiments, a method for intelligently terminating calls may include receiving a call request directed to a communication identifier associated with a first user, determining a call termination action to associate with the call request based on (a) information associated with the call request and (b) previous call termination patterns associated with the first user, and terminating the call to one or more devices associated with the communication identifier based on the determined call termination action.
Abstract:
A method and apparatuses for full sender-side rate control include receiving a data stream at a forwarding unit, determining a bandwidth estimation for communicating the data stream between the forwarding unit and a sender of the data stream, determining respective bandwidth estimations for communicating the data stream between the forwarding unit and at least two receivers, aggregating the bandwidth estimations determined for communicating the data stream between the forwarding unit and the at least two receivers, and limiting the bandwidth estimation determined for communicating the data stream between the forwarding unit and the sender of the data stream based on the aggregated bandwidth estimations. In some embodiments, the bandwidth estimation determined for communicating the data stream between the forwarding unit and the sender of the data stream is limited by communicating a limitation message regarding the aggregated bandwidth estimation of the receivers to the sender.
Abstract:
Methods and system for automating conferencing in a communication session are provided herein. In some embodiments, a method for automating conferencing in a communication session includes establishing a communication session between a first end-user device and a second end-user device; receiving a first notification message from the first end-user device indicating that the communication session is at least one of visible or joinable by one or more devices associated with the first end-user device, wherein the one or more devices are associated with the first end-user device when the one or more devices are one of (i) pre-registered and stored with an association to the first end-user device, (ii) pre-authenticated to tether the one or more devices to the first end-user device, or (iii) pre-associated with a same user account profile; and sending a second notification message to the one or more devices associated with the first end-user device.