Abstract: A method and apparatus are disclosed for congestion management in a multi-branch Internet Protocol-based private branch exchange switch. The multi-branch Internet Protocol-based private branch exchange switch is interconnected through (i) a packet network referred to as the primary network, such as a wide area network, and (ii) an alternate network, such as the public switched telephone network. Packet phone adapters associated with each packet telephone unit monitor packet telephone calls and report delay information to communication servers. The communication server can reroute the packet telephony calls through the secondary network upon detection of congestion in the underlying primary network, thereby preserving voice quality. The packet phone adapter will discard records collected from calls whose duration is below a minimum value, to ensure reliable congestion information. Each communication server records reported voice quality of service information in a congestion control database.

Abstract: A method and apparatus are disclosed for alleviating congestion and overload in a distributed call-processing system interconnected through a packet based network. The illustrative Internet Protocol network includes a plurality of end terminals and distributed call processors. According to an aspect of the invention, the call processor will determine whether to process a call request or to forward the request to another call processor. Generally, the call processor will declare an overload condition if sufficient resources (including processing or memory resources) are not available to process a given call. If a call processor determines that it is too congested to process a call, the call processor enters an overload condition, selects an alternate call processor and forwards the request to the alternate call processor. Each call processor maintains an ordered list of call processors that indicates whether or not each call processor is overloaded.

Abstract: A method and apparatus are disclosed for dynamically adapting the play-out delay for voice packets transmitted over a local area network using the Ethernet standard as an access mechanism. It has been observed that the distribution of voice packet delays in a LAN (shared by voice and data traffic) follows a log-normal distribution. An adaptive algorithm is disclosed to estimate the parameters of the log-normal distribution and to apply a dynamic play-out delay to improve the quality of packetized voice conversations, quantified by minimum delay and packet loss. For example, the size, B, of the play-out buffer can be established to ensure that the packet loss does not exceed one percent (1%). The distribution parameters are continuously updated and the size of the play-out buffer, B, is modified at the beginning of every calibration interval according to the illustrative 99% percentile of the delay distribution.

Abstract: A digital cellular/personal communications service (PCS) application incorporates a logical link connection (LLC) server. In this approach, there are two ATM connections in the path of an AAL-2 connection: one between a base station and the LLC Server and the other between the LLC Server and a vocoder group. All LLCs from, or to, a given base station use a common ATM connection irrespective of the vocoder used at the other end. Similarly, all LLCs from, or to, a given vocoder group use a common ATM connection irrespective of the destination base station at the other end. At the LLC Server, LLC packets from many base stations destined for the same vocoder group are extracted and bundled into the ATM connection between the LLC Server and the destination vocoder group. Similar treatment is given to the packets originating at vocoders and destined for base stations.