Abstract: A network is defined with several alternative softswitches/proxies, which may be used for communication. Each softswitch/proxy has a unique Internet Protocol (IP) address. The softswitches/proxies receive configuration data from a centralized user-profile server, which maintains user-profile information. A centralized call-detail record (CDR) server is connected to each softswitch/proxy and maintains CDRs on each user on each softswitch/proxy. Based on the network configuration, an end-device configuration system generates a provisioning file. The provisioning file includes the IP addresses of each softswitch/proxy. The provisioning file is communicated to user devices. Each user device accesses the provisioning file and uses the IP address for communication. Should the communication fail for any reason, the user device may autonomously access the provisioning file and initiate another call using the next IP address in the provisioning file.

Abstract: A network is defined with several alternative softswitches/proxies, which may be used for communication. Each softswitch/proxy has a unique Internet Protocol (IP) address. The softswitches/proxies receive configuration data from a centralized user-profile server, which maintains user-profile information. A centralized call-detail record (CDR) server is connected to each softswitch/proxy and maintains CDRs on each user on each softswitch/proxy. Based on the network configuration, an end-device configuration system generates a provisioning file. The provisioning file includes the IP addresses of each softswitch/proxy. The provisioning file is communicated to user devices. Each user device accesses the provisioning file and uses the IP address for communication. Should the communication fail for any reason, the user device may autonomously access the provisioning file and initiate another call using the next IP address in the provisioning file.

Abstract: An improved telephony adapter compresses voice data, creates IP packets, and prioritizes the voice IP packets over the data IP packets. Preferably, the compression and packetization interval is such that the bandwidth occupied by the voice IP packets is approximately half of the minimum average available bandwidth in the upstream direction, thereby maintaining acceptable latency and voice quality of the speech. Further enhancement is achieved by causing the ISP to also give priority to voice packets that are destined to the telephony adapter, over the data packets that are destined to the telephony adapter.

Abstract: A terminal adapter for guaranteeing the quality of service of both voice and data packets is disclosed. When a data packet is received in a first data input queue of a terminal adapter, a determination is made whether a voice packet is present in a voice input queue. Another determination is made as to whether the sum of the size of the data packet and the size of all packets in a terminal adapter output queue would exceed a first size threshold established for the output queue. If voice packets are present in the voice input queue, or if the aforementioned sum exceeds the size threshold, the data packet is not forwarded to the output queue. If no voice packets are present in the voice input queue and if the aforementioned sum is below the first size threshold, then the data packet is forwarded to the output queue.

Abstract: Voice over Internet Protocol (VoIP) calls received in a Hybrid Fiber Coax (HFC) network (12) maintained by a provider of HFC VoIP telephony services are advantageously translated into a Time-Division Multiplex (TDM) format by an Internet Protocol Device Terminal (IPDT) (26) in the HFC network. Once translated into a TDM format, the call passes to the Public Switched Telephone Network (PSTN) (28) for call processing to afford the call features subscribed to by the called party, such as caller-ID and call waiting. Once processed, the PSTN routes the call to the destination. Likewise, a call destined for an HFC VoIP customer can be processed within the PSTN to afford the call features subscribed to by the HFC VoIP customer. In this way, the HFC VoIP telephony service can offer full-featured VoIP cable telephony without the need to perform call processing in its own network.

Abstract: Voice over Internet Protocol (VoIP) calls received in a Hybrid Fiber Coax (HFC) network (12) maintained by a provider of HFC VoIP telephony services are advantageously translated into a Time-Division Multiplex (TDM) format by an Internet Protocol Device Terminal (IPDT) (26) in the HFC network. Once translated into a TDM format, the call passes to the Public Switched Telephone Network (PSTN) (28) for call processing to afford the call features subscribed to by the called party, such as caller-ID and call waiting. Once processed, the PSTN routes the call to the destination. Likewise, a call destined for an HFC VoIP customer can be processed within the PSTN to afford the call features subscribed to by the HFC VoIP customer. In this way, the HFC VoIP telephony service can offer full-featured VoIP cable telephony without the need to perform call processing in its own network.

Abstract: A terminal adapter for guaranteeing the quality of service of both voice and data packets is disclosed. When a data packet is received in a first data input queue of a terminal adapter, a determination is made whether a voice packet is present in a voice input queue. Another determination is made as to whether the sum of the size of the data packet and the size of all packets in a terminal adapter output queue would exceed a first size threshold established for the output queue. If voice packets are present in the voice input queue, or if the aforementioned sum exceeds the size threshold, the data packet is not forwarded to the output queue. If no voice packets are present in the voice input queue and if the aforementioned sum is below the first size threshold, then the data packet is forwarded to the output queue.

Abstract: An improved telephony adapter compresses voice data, creates IP packets, and prioritizes the voice IP packets over the data IP packets. Preferably, the compression and packetization interval is such that the bandwidth occupied by the voice IP packets is approximately half of the minimum average available bandwidth in the upstream direction, thereby maintaining acceptable latency and voice quality of the speech. Further enhancement is achieved by causing the ISP to also give priority to voice packets that are destined to the telephony adapter, over the data packets that are destined to the telephony adapter.

Abstract: A terminal adapter for guaranteeing the quality of service of both voice and data packets is disclosed. When a data packet is received in a first data input queue of a terminal adapter, a determination is made whether a voice packet is present in a voice input queue. Another determination is made as to whether the sum of the size of the data packet and the size of all packets in a terminal adapter output queue would exceed a first size threshold established for the output queue. If voice packets are present in the voice input queue, or if the aforementioned sum exceeds the size threshold, the data packet is not forwarded to the output queue. If no voice packets are present in the voice input queue and if the aforementioned sum is below the first size threshold, then the data packet is forwarded to the output queue.

Abstract: An improved telephony adapter compresses voice data, creates IP packets, and prioritizes the voice IP packets over the data IP packets. Preferably, the compression and packetization interval is such that the bandwidth occupied by the voice IP packets is approximately half of the minimum average available bandwidth in the upstream direction, thereby maintaining acceptable latency and voice quality of the speech. Further enhancement is achieved by causing the ISP to also give priority to voice packets that are destined to the telephony adapter, over the data packets that are destined to the telephony adapter.

Abstract: A terminal adapter for guaranteeing the quality of service of both voice and data packets is disclosed. Such quality is ensured by inserting gaps between successive data packets in a stream of multiplexed data and/or voice packets. A gap after a particular data packet is proportional to the size of that particular data packet. In this way, bandwidth is preserved for any voice packets that may have arrived during the transfer of the data packet as well as for any voice packets that arrive during the gap. The unconstrained upstream data bandwidth and the bandwidth used by voice calls may each be estimated by taking a plurality of instantaneous measurements of the available bandwidth and/or taking individual direct measurements. The size of data packets may be limited to a maximum size in order to ensure that time-sensitive voice packets experience only an acceptable delay in queue for transmission.

Abstract: An architecture and technique for creating self-installable and portable telephony (dial tone) service that can be moved between any two locations that has access to both a voice communication network and a data network. A telephony adapter is used as a subscriber premises device that is connected between a conventional telephone set and both a voice network and a data network. A provisioning server communicates with the telephony adapter through the data network and maintains a record of the subscriber's local telephone number and IP address of the telephony adapter. As the subscriber moves from one location to another, the telephony adapter (once turned “on”) will communicate with the provisioning server and re-establish phone service, always using the same local phone number of the subscriber.