Multiplex communications – Data flow congestion prevention or control – Flow control of data transmission through a network
Reexamination Certificate
2000-05-24
2004-06-08
Yao, Kwang Bin (Department: 2667)
Multiplex communications
Data flow congestion prevention or control
Flow control of data transmission through a network
C370S395210
Reexamination Certificate
active
06747953
ABSTRACT:
FIELD OF INVENTION
The invention pertains to congestion control for communications networks and, particularly, packet-based voice communications networks.
BACKGROUND OF THE INVENTION
The use of packet-based networks for communicating data between various locations is well known. Packet-based networks also are increasingly being used to carry voice or combined voice and data traffic. Particularly, public telephone companies are using packet-based networks as part of the public telephone networks. The two types of packet-based networks in most common use today are ATM (Asynchronous Transfer Mode) networks and IP (Internet Protocol) networks. The small but growing number of networks which use either of these types of packet-based networks for transmitting voice (or voice and data) are termed VTOA (Voice Traffic Over ATM) and VoIP (Voice-over-IP) networks, respectively.
In packet-based networks, data streams transferred between two nodes of the network are transferred in discrete packets. Packets may be of consistent size or may be variably sized. Each packet includes a field of data which may be preceded and/or followed by non-data information such as preamble and housekeeping information (such as data source and destination addresses and the like). The data source destination fields of the packet are used by the switches and routers in the network to route the data through network nodes from source node to destination node.
The destination node reconstructs the data in the proper order based on the housekeeping information to accurately recreate the transmitted data stream. Accordingly, a continuous stream of data can be generated at a source node of the network, transmitted in separate packets (which can even travel separate routes) to the destination node and be reassembled at the receiving node to recreate a continuous stream of receive data. The fact that the data was packetized for transmission and reassembled is transparent to the users at the source and destination nodes.
However, when congestion on the network exceeds the physical capabilities of the network, i.e., the amount of data that must be transmitted via the network for all of its source-destination node pairs exceeds the physical capabilities of the network, packets get dropped and the apparently seamless transmission of data falls apart. Particularly, using a voice link as an example, transmission links in which packets are being dropped will have noticeably reduced audio quality. The receiving party may perceive noise, delay jitter and even gaps in the received voice transmission. Delay jitter occurs when the packets do not arrive quickly enough to be recreated seamlessly to produce a continuous stream of data. Noise and gaps appear when packets are completely dropped.
FIG. 1
is a block diagram of an exemplary communication network
10
, including a packet-based communication sub-network
12
. The packet-based sub-network
12
of
FIG. 1
forms a portion of an overall telephone communications system
10
in this particular example. The packet-based network
12
interconnects a plurality of public service telephone networks (PSTNs)
14
. In other words, some nodes of the packet-based network
12
are gateway nodes through which data from other networks, for example, time division multiplexed (TDM) PSTN networks
14
is transferred. The network nodes through which the PSTNs are coupled to the packet-based network are termed gateways. The gateways
18
convert voice data from the protocol of the PSTN networks
14
(e.g., a TDM protocol) to the protocol of the packet-based network
12
(i.e., packets).
Using voice data being transmitted from PSTN
14
a
to PSTN
14
b
as an example, gateway
18
a
converts the TDM voice into packet voice and transmits it to an edge switch or edge router
20
a
. From edge router
20
a
, the data is directed to edge router
20
b
, preferably, directly. However, the data may be routed through one or more switches and/or routers
24
. Edge router
20
b
directs the packets to gateway
18
b
, where packet voice data is converted to the TDM format and then forwarded on the PSTN network
14
b.
The term edge switch or edge router refers to a device that is primarily responsible for performing network access functions for the packet-based network, such as low level traffic classification, aggregation and management. As its name implies, an edge switch or edge router typically is coupled directly to a gateway into the network, i.e., it is at the edge of the packet-based network. The network further comprises core switches and core routers
24
. Both of these terms refer to devices that perform larger scale traffic classification, aggregation and management.
There are several known mechanisms for handling congestion in communications networks. For instance, call blocking is a mechanism by which new calls (i.e., source/destination node pairs) are not accepted by the network when the quality of existing calls starts degrading due to congestion. Another method known for reducing congestion at the congested point is by rerouting calls in the PSTN networks. Particularly, the TDM networks may be coupled to the central packet-based network at multiple gateways. If a path through one particular gateway is congested, the PSTN network can reroute calls to another gateway through which less congested paths in the packet-based network can be accessed.
It is also possible to use compression algorithms to reduce the bandwidth needed for a call and thus allow more calls to be accommodated. With respect to voice communications, many voice compression algorithms are known. However, compression typically leads to loss of information. Generally, the greater the amount of compression, the greater the loss of data. In connection with voice communications, generally, the greater the amount of compression, the lower the audio quality of the call.
Call gapping in the PSTN networks is another possible mechanism for reducing congestion.
It is an object of the present invention to provide a network congestion management method and apparatus for use in connection with packet-based networks.
SUMMARY OF THE INVENTION
The invention is a method and apparatus for congestion management in a packet-based network that interconnects a plurality of peripheral networks. We disclose a method and apparatus by which the amount of congestion is quantified and particular corrective actions and the level of those corrective actions necessary to eliminate or reduce data loss is generated. The generated data may be transferred to those network elements (or the peripheral networks) which can implement the indicated corrective actions or may be implemented automatically by a centralized network congestion manager circuit (e.g., a dedicated digital signal processor). The system operates essentially in real time. Among the corrective measures supported by the invention are reconfiguration of bandwidth pipes, call blocking, rerouting in the peripheral networks (e.g., PSTN networks that are coupled to a central packet-based network at multiple gateways), selective voice compression on new calls and call gapping in the peripheral network.
We disclose a method and apparatus for determining the congestion at the ports of a network and determining the level of call blocking needed to provide a predetermined quality of service for calls.
The network congestion management in accordance with the present invention may be a centralized system or it may be a distributed system in which multiple nodes of the network contain elements of the network congestion management hardware and software.
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Hernandez-Valencia Enriq
Park Kun I.
Qureshi Muhammed A.
Wang Yung-Terng
Lucent Technologies - Inc.
Yao Kwang Bin
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