摘要:
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.
摘要:
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.
摘要:
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.
摘要:
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.
摘要:
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.
摘要:
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.
摘要:
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.
摘要:
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.
摘要:
Telephone calls are directed from a first telecommunications platform to a second telecommunications platform, together with an identifier which uniquely identifies the call and enables telecommunication switches to monitor and maintain control over the call. The unique identifier enables the call to be returned from the second platform to the first platform. Once returned to the first platform, the call can be directed to a third telecommunications platform (and, subsequently, back to the first platform), without requiring the caller to place another call or to re-verify the caller's identity. The call identifier is used either to automatically return the call to the first platform after the occurrence of a predetermined event, or upon detecting predetermined, caller-dialed keystrokes indicating that the caller wishes to return to the interexchange carrier platform.
摘要:
A method and an apparatus for detecting relocation of endpoint devices used for accessing services provided on packet networks, such as Voice over Internet Protocol (VoIP) networks are disclosed. For example, the method receives a report of a detected inertial movement of a customer endpoint device, and the method then uses the report of the detected inertial movement for determining whether a relocation of the customer endpoint device has occurred.