Abstract:
The present invention provides a method for evaluating a communication link between a first communication site and a second communication site. The first and second communication sites increment a sent counter and reset a received counter upon sending a message to the other communication site. Each communication site also increments a received counter and reset a sent counter upon receiving a message from the other communication site. When the sent counter exceeds a sent threshold, the communication system site sends an alert that the communication link is down. When the received counter exceeds a predetermined percentage of the other communication site's sent counter, the site sends a message to the other communication site to let the other communication system site know that it is still receiving messages.
Abstract:
A communication network (100) includes a first server (124) and a second server, such as any of the servers (121-23), connected via a common network (101). First server (124) owns a record data (180) associated with at least one process running for serving a client device (174.) The second server keeps a copy of the record data. First server (124) performs a Hashing function over record data (180) to produce a first Hash value. The second server similarly performs the same or similar Hashing function over the copy of the record data to produce a second Hash value. First server (124) sends the first Hash value to the second server for comparison. If the first Hash value fails to match to the second Hash value, a latest copy of the record data (180) is sent from first server (124) to the second server upon request.
Abstract:
A method of switching a call to a multipoint conference call includes sending a message (201) from a first terminal (105-T1) to a gatekeeper (102) which provides address translation and control access to a shared network medium (101). The call is initially established as a point to point communication between first terminal (101) and a second terminal (105-TN) over shared network medium (101) while complying with H.323 standard. Message (201) contains a request for the multipoint conference call. The method furthermore includes selecting a multipoint control unit (104) connected to shared network medium (101) to allocate resources for the multipoint conference call, and then switching the call to the multipoint conference call via the allocated resources. Thereby, the initial call is switched to a multipoint conference call without interrupting the initial call.
Abstract:
A method and apparatus for data transmission is provided herein. In accordance with the preferred embodiment of the present invention a loss-ratio estimator (105) estimates a current loss (L) for a communication channel (108). Once the actual loss for the channel is known, a generator (104) compares the actual loss (L) to a target loss (T). A retransmission control parameter (R) is then adjusted by the generator (104) and output to a transmitter 103 where it is used to control the retransmission behavior and to determine when to abort a bad frame. When a bad frame is aborted, transmitter 103 indicates the abortion to a receiving device (102). A receiver then utilizes the indication to stop reporting the bad frame in all subsequent ACK/NAKs.
Abstract:
Methods and systems are provided for routing callers to agents in a call-center routing environment. An exemplary method includes identifying caller data for a caller in a queue of callers, and jumping or moving the caller to a different position within the queue based on the caller data. The caller data may include one or both of demographic data and psychographic data. The caller can be jumped forward or backward in the queue relative to at least one other caller. Jumping the caller may further be based on comparing the caller data with agent data via a pattern matching algorithm and/or computer model for predicting a caller-agent pair outcome. Additionally, if a caller is held beyond a hold threshold (e.g., a time, “cost” function, or the like) the caller may be routed to the next available agent.
Abstract:
An RTP payload header (408) is compressed by utilizing a codebook (514) that holds one or more indexable payload header variations and sending it to a receiving device (504). An RTP packet (400) that includes the payload header (408) with information pertaining to the packet is generated by a sending device (502) and the codebook is searched for the payload header information. If the information is found, the payload header (408) is replaced with a short index in the codebook (514). At the receiving device (504) the index is used to retrieve a payload header variation that corresponds to the index and the variation is placed back into the payload header (408) to uncompress the header.
Abstract:
In the present technique of streaming a main media stream that has been requested, an anti-shadow stream (36) that represents a backup copy of the main media stream (24) is sent along with an output media stream (34) that represents an output copy of the main media stream. The content of the anti-shadow stream (36) is preferably forward-shifted in time from the output media stream (34) so as to provide replacement of loss data of the output stream. Put differently, sequenced data frames of the output stream (34) are delayed by order compared to that of the anti-shadow stream (36).
Abstract:
Methods and systems are provided for routing callers to agents in a call-center routing environment. An exemplary method includes pooling incoming callers, and causing a caller from the pool of callers to be routed. The caller may be routed from the pool of callers to an agent, placed in another pool of callers, or placed in a queue of callers. The caller data may include demographic or psychographic data. The caller may be routed from the pool of callers based on comparing the caller data with agent data associated with an agent via a pattern matching algorithm and/or computer model for predicting a caller-agent pair outcome. Additionally, if a caller is held beyond a hold threshold (e.g., a time, “cost” function, or the like) the caller may be routed to the next available agent.
Abstract:
Various embodiments are described to enable improved inter-network/inter-technology handover of mobile devices. A network device (131, 132) collects dynamic information corresponding to mobile devices (101, 102), such as wireless measurement information at the device's location, and/or information corresponding to wireless network nodes (121-124), such loading levels/loading distributions. The network device then sends some or all of the dynamic information collected and/or statistical information generated from the dynamic information collected to a neighboring network information server (150) for access by other communication networks. By maintaining dynamic and/or statistical information in a neighboring network information server, such information can be made available to all the communication networks in a given region. One potential benefit to making this information available is improved inter-network handoff decision-making.
Abstract:
Methods and systems are provided for routing callers to agents in a call-center routing environment. An exemplary method includes identifying caller data for a caller in a queue of callers, and jumping or moving the caller to a different position within the queue based on the caller data. The caller data may include one or both of demographic data and psychographic data. The caller can be jumped forward or backward in the queue relative to at least one other caller. Jumping the caller may further be based on comparing the caller data with agent data via a pattern matching algorithm and/or computer model for predicting a caller-agent pair outcome. Additionally, if a caller is held beyond a hold threshold (e.g., a time, “cost” function, or the like) the caller may be routed to the next available agent.