摘要:
A method and apparatus are used in a gateway to discard selected frames received with a selected encoded-information-type from a communication link with a larger bandwidth to avoid overflowing an internal delay variance removing queue used for protocol translation to a communication link with a smaller bandwidth. The discarded frames do not decrease the quality of translated information. A visual delay variance removing queue congestion indicator is included to indicate three levels of congestion in the delay variance removing queue for received frames. The method and apparatus are used in a multimedia gateway which is translating audio/video conferencing protocols (e.g., H.320, H.323/LAN H.323/PPP and H.324) received from a communication link with a large bandwidth and sent to a communication link with a smaller bandwidth.
摘要:
Gateway routers for real-time networks have the ability to collect delay, loss, and jitter statistics on a per-connection basis. It is possible to use this information not only to monitor the quality of individual voice calls and other real-time connections, but also to evaluate the overall performance of the underlying network. This paper describes a method for monitoring and managing the performance of a real-time data network that supports voice, video and other real-time services. In the described embodiments, the RTCP mechanisms of RTP for sender and receiver reporting be used to relay performance information to one or more network monitoring sites for analysis and interpretation. In addition, gateway routers are organized and managed within a hierarchy that allows the monitoring function to localize it view of network conditions within the hierarchy; and the monitoring of network performance can occur on various time scales.
摘要:
A system and method for providing telephone service to a user of a telecommunications device using a data network service provider. The data network service provider has a local service host that is accessible by a local access identifier. A caller uses a telecommunications device to dial the local access identifier to connect to the local service host. In response to a prompt, the caller dials a telephone extension that identifies the callee's telecommunications device. The local service host receives the telephone extension and verifies that the callee is a subscriber. The local service host then retrieves the gateway nearest the callee telecommunications device and opens a voice-over-data channel between the callee and caller gateways. The telephone conversation then proceeds between the callee and caller telecommunications devices over a public switched telephone network connection to the caller gateway, the voice-over-data channel and the PSTN connection to the callee telecommunications device.
摘要:
A system and method for providing location information and other information about a calling telephone to the caller during a telephone connection in a data network telephony system. Data network telephones may be provisioned and otherwise configured for operation with an extensive database and other user account information. The user's account may include a location identifier that identifies the physical location of the telephone. The location identifier may provide address information, latitude and longitude configuration, directions, or other information. In one embodiment of the present invention, the location information is communicated in a data communications channel between the caller and the callee. In another embodiment of the present invention, the location information is communicated to the caller during call setup. The embodiments of the present invention are useful in providing emergency dispatch services, such as 911 in a data network telephony system.
摘要:
A computationally simple yet powerful forward error correction code scheme for transmission of real-time media signals, such as digitized voice, video or audio, in a packet switched network such as the Internet. An encoder at the sending end derives p redundancy blocks from each group of a k payload blocks and concatenates the redundancy blocks, respectively, with payload blocks in the next group of k payload blocks. At the receiving end, a decoder may recover up to p missing packets in a group of k packets, provided with the p redundancy blocks carried by the next group of k packets. The invention thereby enables correction from the loss of multiple packets in a row, without significantly increasing the data rate or otherwise delaying transmission.
摘要:
A method and apparatus for communication system buffer size and error correction coding selection. A method includes the steps of receiving a stream of data packets by a real time receiver that includes a buffer management device, a first plurality of jitter buffers, and a second plurality jitter buffers. The first and second plurality of jitter buffers are evaluated and a first and a second optimal jitter buffer is chosen. The first and the second optimal jitter buffer has an associated conditional optimal performance characteristic. The conditional characteristics are compared and a preferred buffer of the receiver is selected. The apparatus includes a receiving device including a first set of jitter buffers and a second set of jitter buffers with error coding. The first set includes a plurality of buffers and a second plurality of buffers maintained in the second set of buffers. The apparatus also includes a means for comparing the first plurality of buffers and the second plurality of buffers, a means for selecting a first optimal buffer from the first plurality of buffers, and a means for selecting a second optimal buffer from the second plurality of buffers. Either the first or the second selected optimal decoder determines the receiver buffer size and whether forward error correction is utilized.
摘要:
A method and apparatus for improving the speed and quality of end-to-end data or real-time media transmissions over an internet is disclosed. A data stream representing a media signal at a given level of compression is processed just before the data stream enters the internet. A less compressed data stream representing the same media signal is generated transmitted through the internet. Due to the lower level of compression, the underlying media signal is less sensitive to packet loss in the internet and, as a result, the media signal that arrives at the receiving end will tend to be more continuous and clear.
摘要:
An improved system for identifying the loudest speech signal in a G.723.1 based audio teleconferencing link is disclosed. The system selects the loudest of several analog audio signals by directly analyzing the encoded G.723.1 bit streams representing those signals, rather than by decoding the encoded speech signal in the G.723.1 bit streams and then re-encoding the signal as a selected output bit stream. The system uses the excitation gain parameters encoded in G.723.1 frames to approximate frame gains for respective bit streams and then estimates a short term speech energy for each bit stream by averaging the approximate frame gains over time. The system then compares the estimated speech energy levels and outputs to each conference participant the signal with the highest estimated speech energy as the next portion of an output signal.
摘要:
A system and method for accessing a data network telephony account using a wireless personal information device (PID). The user may connect over the wireless cellular infrastructure to a telephony control server via a data network for access to the user's telephony account, which indicates the user's telephone number in a telephone number entry. Once the connection is made, the user issues a command to set the telephone number entry in the user's data network telephony account to a specific telephone number. The user may then invoke a contacts application in the wireless PID and select a person to call from the contacts list. The user selects the entry to send a command to initiate a telephone connection between the party selected and the user at the telephone designated by the user at the telephony control server.
摘要:
A system and method of accessing radio programming from radio stations that communicate radio programming on a data network. The radio programming is accessed as radio or audio signals formatted in radio-over-data packets to a data network telephone. The data network telephone is a telephone that uses voice-over-data communications channels over a data network to make telephone connections with other data network telephones. The data network telephone includes a display, a keypad, a handset and an optional speaker output. The data network telephone advantageously permits simultaneous access to radio programming and communication on a telephone connection. The data network telephone also includes an interface to a portable information device. A radio control application may be used on the portable information device that communicates control information to a radio application on the data network telephone. The PID may be used to set the desired radio station, volume and other settings that may be communicated to the radio application on the data network telephone.