Abstract: A method for managing communications while an application location changes between a first wireless network and a second wireless network is provided. The method includes mapping an internal address to a first external address for the first wireless network. A second wireless network is determined to use for sending communications for the application. The internal address is then mapped to a second address for the second wireless network.

Abstract: A system, method, and apparatus are directed towards routing a Voice over IP (VOIP) messages over a network. The VOIP messages are sent by a source client device to a destination client device through a portal service that has access to multiple routing services such as peering partners, carriers, etc. A VOIP system of the portal service aggregates call quality data after each VOIP call over each routing service. The call quality data is analyzed to determine a perception factor for each routing service at various times of day, days of the week, day of the year, geographic areas, and the like. When a VOIP call is requested through the portal service, the VOIP system determines a current cost, a current quality of service, and the perception factor for each routing service. A weighting is applied to each criterion and a routing service is selected for routing the VOIP call.

Abstract: A system, method, and apparatus are directed towards managing a Voice over IP (VOIP) message over a network. A computing device may be configured to select a network connection for which to send the message to a destination based on a variety of factors. Duplicate message packets may be communicated to the destination device through multiple network connections. The multiple network connections may include a peer-to-peer network connection, a peer network connection, an ad-hoc network connection, or the like. Metrics may be collected about the multiple network connections. A determination may be made based, in part, on the metrics whether one network connection is optimal over another network connection. If so, that network connection may be selected to continue to provide the message packets, and the communication of the duplicate packets is ceased.

Abstract: A system, method, and apparatus are directed towards a system, method, and apparatus for managing a communication session by dynamically selecting a CODEC. A client device requests a communication session with a receiver device. If available, historical information may be used to establish an initial CODEC and its associated sampling frequency for the communication session. Signals, such as a white noise signal, and/or a ring tone may be sent between the client device and the receiver to determine a metric for the communication session. The metric may be used to adjust the CODEC and/or its sampling frequency for the communication session. In one embodiment, if it is determined that the sampling frequency is less than a minimum determined value, a message may be sent to the client device advising that the current communication session be terminated.

Abstract: A system, method, and apparatus are directed towards managing a Voice over IP (VOIP) messages over a network, employing the Real-time Transport Protocol (RTP) and Session Initiation Protocol (SIP) over the Transmission Control Protocol (TCP). The VOIP messages are sent by a source device to a destination device through a relay server. The relay server may throttle the VOIP messages employing buffer management. When the buffer is substantially full, the relay server will drop packets from the source device. Indication of the lost packets may be provided to the source device through a Real-time Transport Control Protocol (RTCP) report. The source device may then employ the RTCP report to modify a type of codec employed, and thereby adjust a rate of flow of VOIP packets sent towards the destination device. Additionally, the relay server may provide port translation services for RTP/RTCP packets between the source and destination devices.

Abstract: A system, method, and apparatus are directed towards managing a Voice over IP (VOIP) message over a network, where the VOIP message may employ the Real-time Transport Protocol (RTP) and possibly Session Initiation Protocol (SIP) over the User Datagram Protocol (UDP). The invention enables a VOIP client device, such as an IP phone, and the like, to communicate a message to a local proxy residing on a local computing device. The communications between the VOIP device and the local proxy may employ SIP/RTP over UDP. Upon receipt of the communications, the local proxy converts the transport protocol to another transport protocol, such as Transmission Control Protocol (TCP). The local proxy may also perform a port translation on the message. The converted communications may then be sent to a remote server, where it may be employed in its present SIP/RTP over TCP format, or be converted back to using UDP.

Abstract: A system, method, and apparatus are directed towards providing integrated access to call information to an end-user using any of a variety of client applications, including an Instant Messaging (IM), a web-browser, and/or a VOIP client. An end-user may employ any of the above client applications to establish a communication session and enable an integrated access display for use in managing such call information as audio messages, call histories, and instant messaging conversations. The integrated access display may be managed through an intelligent integration client that may reside on a client device, such as a personal computer, a mobile device, a VOIP phone, and the like. Through the intelligent integration client, the end-user may search, retrieve, and/or view the call information. For example, using the VOIP phone, the end-user may search for and access a stored audio message, such as a voice mail message, or the like.

Abstract: A system, method, and apparatus are directed towards managing an audio message, such as a Voice over Internet Protocol (VOIP) message over a network. The invention employs a statistical mechanism to automatically optimize a gain control for setting a volume of an audio message being sent by a client device. An initial gain value is automatically adjusted based, in part, on a statistical sampling of energy levels in the audio message. Environmental factors, such as a sound card within the client device, background noise, and the like, may also be considered through a setting of a servo coefficient that may be used to map between volume levels and decibel levels. The servo coefficient may also be adjusted based, at least in part, on decibel (dB) feedback information from a destination device for which the audio message is intended.

Abstract: An integrated VoIP unified message processing system includes a voice platform that processes data in native VoIP format. There is no use of hardware telephone interface cards (TICs) or software transcoding to transform data to PCM or other formats. Cost reductions are achieved by the elimination of expensive dedicated hardware and scalability is achieved by obviating the need for software transcoding.

Abstract: Techniques for using instant messaging preferences and settings are used for predictive communications, including routing of data traffic, are provided. A voice call can be routed over a packet-based network if a voice network is not available, or for other reasons. In addition to voice calls, embodiments of the invention can route video calls, multimedia communications and other communications. In some implementations, an instant message presence can be queried for in determining how to route a voice-over-IP or PSTN telephone call. Thus, a call might be routed to a computer or computing device unrelated to the location of the PSTN telephone line being called, if an associated instant messenger presence is detected on that computer or computing device.

Abstract: A method for managing communications while an application location changes between a first wireless network and a second wireless network is provided. The method includes mapping an internal address to a first external address for the first wireless network. A second wireless network is determined to use for sending communications for the application. The internal address is then mapped to a second address for the second wireless network.

Abstract: Techniques for using billing services to manage a personal address book are provided. The techniques comprise receiving a call detail record (CDR) for a plain old telephone system (POTS) or a voice-over-IP (VOIP) call by a user. The CDR is created by a telecommunication carrier that routed the call. The CDR may be stored on a telecommunications server associated with the telecommunications carrier. Address information for the call is determined from the CDR. This address information is added to the personal address book for the user. This address information may be available for the user when using an IM client.

Abstract: An integrated VoIP unified message processing system includes a voice platform that processes data in native VoIP format. There is no use of hardware telephone interface cards (TICs) or software transcoding to transform data to PCM or other formats. Cost reductions are achieved by the elimination of expensive dedicated hardware and scalability is achieved by obviating the need for software transcoding.

Abstract: A unified, intelligent message store is scalable and is optimized to provide both streaming and non-streaming data, as required, to telephony and web-based applications. A message application server (MAS) includes an expandable number of message pools, each message pool having message blocks that can be linked to store variable length messages. Multiple MASs can be used to extend the system.

Abstract: A voice converter includes upstream and downstream resource managers for allocation half-duplex resources of a full duplex DSP transcoder. The resource managers schedule available upstream or downstream resources and provide transcoded voice data to requesting applications.

Abstract: An intelligent voice bridging system and process that scaleably extends the functionality of a VOIP telecommunication application by bridging data from external components into the VOIP data stream.