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Method of communicating a data stream over a communication network

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Publication number
WO2009000012A1
WO2009000012A1 PCT/AU2008/000744 AU2008000744W WO2009000012A1 WO 2009000012 A1 WO2009000012 A1 WO 2009000012A1 AU 2008000744 W AU2008000744 W AU 2008000744W WO 2009000012 A1 WO2009000012 A1 WO 2009000012A1
Authority
WO
Grant status
Application
Patent type
Prior art keywords
data
packet
stream
network
packets
Prior art date
Application number
PCT/AU2008/000744
Other languages
French (fr)
Inventor
Ronni Sahlberg
Mats Johan Dahlstedt
Original Assignee
Mobile Ip Pty Ltd
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date

Links

Classifications

    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L1/00Arrangements for detecting or preventing errors in the information received
    • H04L1/0001Systems modifying transmission characteristics according to link quality, e.g. power backoff
    • H04L1/0006Systems modifying transmission characteristics according to link quality, e.g. power backoff by adapting the transmission format
    • H04L1/0007Systems modifying transmission characteristics according to link quality, e.g. power backoff by adapting the transmission format by modifying the frame length
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L1/00Arrangements for detecting or preventing errors in the information received
    • H04L1/0001Systems modifying transmission characteristics according to link quality, e.g. power backoff
    • H04L1/0015Systems modifying transmission characteristics according to link quality, e.g. power backoff characterised by the adaptation strategy
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L1/00Arrangements for detecting or preventing errors in the information received
    • H04L1/12Arrangements for detecting or preventing errors in the information received by using return channel
    • H04L1/16Arrangements for detecting or preventing errors in the information received by using return channel in which the return channel carries supervisory signals, e.g. repetition request signals
    • H04L1/18Automatic repetition systems, e.g. van Duuren system ; ARQ protocols
    • H04L1/1867Arrangements specific to the transmitter end

Abstract

A method of communicating a data stream formed from a plurality of data packets over a communication network. A sending station determines a current packet loss value of the data stream. The sending station then decreases a current packet size of the data packets of the data stream if the current packet loss value of the data stream is greater than a predefined data packet loss value.

Description

"METHOD OF COMMUNICATING A DATA STREAM OVER A COMMUNICATION NETWORK"

FIELD OF THE INVENTION

The invention relates to a method of communicating a data stream of data packets over a communication network. In particular, although not exclusively, the invention relates to a method of communicating a data stream over a congested communication network.

BACKGROUND TO THE INVENTION

Various technological advancements in wireless communications have resulted in a proliferation of mobile communication devices that are connected to the Internet via mobile operator networks, such as the 3G Universal Mobile

Telecommunications System (UMTS) radio networks. Mobile operator networks use different technologies and, as a consequence, different interfaces to provide wireless mobile communication services. For example, the 3G UMTS radio networks use a Wideband Code Division Multiple Access (WCDMA) interface technology. Subscriber stations of the mobile operator networks connect to a radio base station of the mobile operator networks using wireless modems that are based on an interface technology of the mobile operator networks.

Mobile operator networks often provide a communication network that allows Internet access using Media Gateways, which convert data from a format used in the mobile operator networks to another format used in the Internet.

Thus, subscriber stations can access various internet services, such as World

Wide Web (WWW) services, video streaming services, and Voice over Internet

Protocol (VoIP) telephony services. Since communication systems in the communication network are normally packet-based communication systems, an application of a subscriber station needs to break application data into relatively small units of data called packets, also referred to as datagrams, so that they are compatible with an underlying communication layer, such as the Point-to-Point Protocol (PPP) data link layer or the Ethernet data link layer. The data can then be communicated over the communication network in a data stream of the data packets.

However, since a communication network provided by a mobile operator network is shared by multiple subscriber stations, there can be network congestion, resulting in frequent loss of data packets that are transmitted over the communication network. For example, hundreds of subscriber stations can be connected to a single access point, such as a radio base station coupled to be in communication with a core network and a media gateway of a 3G UMTS radio network. This means that the combined data rate of the subscriber stations can exceed, for example, the data rate of a shared network connection between the radio base station and the media gateway.

To overcome some of the problems caused by network congestion, a mobile operator network often provides provisions for Quality of Service (QoS) over different types of network traffic. For example, radio communication links between radio base stations of the mobile operator network and a subscriber station can have different service level agreements (SLAs) for different types of network traffic. For instance, video and audio traffic can be given higher priority over data traffic. However, once the network traffic enters a core network or a media gateway of the mobile operator network and into the Internet, the service level agreements no longer applies and thus fails to provide agreed bandwidth. The amount of packet loss on a congested communication network may be increased as the data packet size becomes larger. For example, large data packets take longer to pass through the congested communication network than small data packets. Since data packets normally have time-to-live values, if the data packets do not arrive at a destination within the time-to-live, the data packets will be dropped before they arrive at the destination. This means that data packets will be retransmitted, unnecessarily consuming network resources and reducing overall quality of communication services. Using smaller data packets than necessary, however, will reduce efficiency of underlying network layers, such as the Ethernet physical layer, because each data packet normally carries additional control data, such as a header comprising destination information, source information, and other packet information, such as packet size.

OBJECT OF THE INVENTION Therefore, an object of the present invention is to overcome or alleviate one or more limitations of the prior art including providing an improved method of communicating a data stream of data packets over a communication network.

DISCLOSURE OF THE INVENTION

In one form, although it need not be the only or indeed the broadest form, the invention resides in a method of communicating a data stream formed from a plurality of data packets over a communication network, the method including the steps, at a sending station, of:

(i) determining a current packet loss value of the data stream; and (ii) decreasing a current packet size of the data packets of the data stream if the current packet loss value of the data stream is greater than a predefined data packet loss value.

In a further embodiment, the method further includes the steps of: (iii) transmitting a test data stream formed from a plurality of test data packets over the communication network if the current packet size of the data packets forming the data stream is less than a preferred packet size;

(iv) determining a packet loss value of the test data stream; and

(v) increasing the current packet size of the data packets forming the data stream is less than a predefined test pack loss value.

In another further embodiment, the current packet loss value of the data stream and the packet loss value of the test data stream over the communication network are determined by determining a number of lost data packets within a first time limit, wherein the lost data packets are data packets in the data stream or the test data stream that are not acknowledged by a receiving station within a second time limit.

In yet another further embodiment, the sending station continuously adjusts the current packet size of the data stream by decreasing the current packet size of the data stream while the current packet loss value of the data stream is larger than the predefined data packet loss value, and by increasing the current packet size of the data stream while the packet loss value of the test data stream is less than the predefined test data packet loss value.

In yet another further embodiment, the sending station is a network router, an Internet application server, a media gateway, or a network client. In yet another further embodiment, the communication network comprises a virtual tunnel.

In yet another further embodiment, the virtual tunnel is a Point-to-Point Protocol (PPP) connection.

Further features of the present invention will become apparent from the following detailed description.

BRIEF DESCRIPTION OF THE DRAWINGS

To assist in understanding the invention and to enable a person skilled in the art to put the invention into practical effect preferred embodiments of the invention will be described by way of example only with reference to the accompanying drawings, wherein:

FIG 1 shows a block diagram illustrating components of a communication network according to an embodiment of the invention; and

FIG 2 shows a general flow diagram illustrating a method for communicating a data stream of data packets over a communication network according to an embodiment of the invention.

DETAILED DESCRIPTION OF THE INVENTION

Embodiments of the present invention comprise a method and system for communicating a data stream of data packets over a communication network. Elements of the invention are illustrated in concise outline form in the drawings, showing only those specific details that are necessary to understanding the embodiments of the present invention, but so as not to clutter the disclosure with excessive detail that will be obvious to those of ordinary skill in the art in light of the present description.

In this patent specification, adjectives such as first and second, left and right, front and back, top and bottom, etc., are used solely to define one element or method step from another element or method step without necessarily requiring a specific relative position or sequence that is described by the adjectives. Words such as "comprises" or "includes" are not used to define an exclusive set of elements or method steps. Rather, such words merely define a minimum set of elements or method steps included in a particular embodiment of the present invention.

FIG 1 shows a block diagram illustrating components of a communication network according to an embodiment of the invention. Subscriber stations 105 are in communication with a mobile operator network 110 either directly or via an access point 115. Subscriber stations 105 are suitably in the form of a wireless network device, such as a smart phone equipped with a Wireless Local Area Network (WLAN) interface, or a wired network device, such as a personal computer (PC) with an Ethernet interface.

The mobile operator network 110 suitably provides access to other networks, such as the Internet 120. For example, the mobile operator network 110 can comprise a radio base station 125 coupled to be in communication with a core network 130, which is then coupled to be in communication with a media gateway 135. The media gateway 135 provides a gateway between the mobile operator network 110 and the Internet 120 by providing communication data conversions between a format used in the mobile operator network 110 and another format used in the Internet 120.

Preferably, the radio base station 125 provides different network interfaces, such as the Wideband Code Division Multiple Access (W-CDMA) interface, the Wireless Local Area Network (WLAN) interface, and the Code Division Multiple Access (CDMA) interface. Preferably, the access point 115 provides network routing services or gateway services for the subscriber stations 105 by establishing one or more virtual tunnels, for example, with the media gateway 135 of the mobile operator network 110. Alternatively, each subscriber station 105 establishes a virtual tunnel with the media gateway 135 to access the Internet 120. A virtual tunnel is suitably a Point-to-Point Protocol (PPP) connection or a variant of a PPP connection over a network interface, such as a W-CDMA interface, a WLAN interface, or a CDMA interface. More than one virtual tunnel may be established with one or more mobile operator networks 110 to provide improved data rate by sharing network traffic over the established virtual tunnels.

Therefore, the subscriber stations 105 are able to access rich content of the Internet 120 by transmitting and receiving data over the communication network 100. For example, each subscriber station 105 is able to access various internet services provided by an application server 140, such as World Wide Web (WWW) services, file transfer services, and audio/video data streaming services. Furthermore, each subscriber station 105 can act as an audio/video streaming server. The access point 115 may also act as an intelligent access point providing audio/video streaming server services.

When the subscriber stations 105 or the access point 115 n eeds to upload or download large application data, such as an image file, the data is divided into smaller units of data packets, before they are transmitted over the communication network 100. The data packets are then transmitted over the communication network 100 in a stream of the data packets.

For example, the access point 115 can divide network traffic data for a large image file into small units of data packets and transmit them over one or more of virtual tunnels that are established between the access point 115 and one or more of media gateways 135. The media gateways 135 then route the data packets to the Internet 120, and the Internet 120 in turn routes the data packets to the application server 140. As illustrated in FIG 1 , the communication network 100 may be shared by multiple subscriber stations 105. Thus, when multiple subscriber stations 105 compete for shared network bandwidth, network congestion can happen at a transport network of the communication network 100. For example, a network connection between the radio base station 125 and the media gateway 135 can be congested. This means that a virtual tunnel between the access point 115 and the media gateway 135 can be congested, and consequently some transmitted data packets may be dropped.

Therefore, preferably the access point 115 determines a current packet loss value of a data stream of data packets over a virtual tunnel. If the current packet loss value of the data stream is larger than a predefined data packet loss value, the access point 115 then reduces a current packet size of the data stream.

However, using smaller data packets than necessary will reduce efficiency of underlying network layers, such as the PPP data link layer. Thus, if the current packet size of the data stream is less than a preferred packet size, preferably the access point 115 sends a test data stream of test data packets that is preferably larger than the current packet size of the data stream over the congested virtual tunnel to determine a packet loss value of the test data stream. The access point 115 then increases the current data packet size if the packet loss value of the test data stream is less than a predefined test data packet loss value.

FIG 2 shows a general flow diagram illustrating a method for communicating a data stream of data packets over a communication network according to an embodiment of the invention. Once a sending station has network traffic data to transmit, at step 205, the sending station begins packetizing the network traffic data into a plurality of data packets and begins transmitting the data packets in a data stream over a communication network to a receiving station. For example, the access point 115 can be a sending station that transmits a data stream of data packets over a virtual tunnel, such as a PPP link to the media gateway 135. The media gateway 135 forwards the network traffic data to a receiving station, such as the application server 140, through the Internet 120.

Alternatively, the subscriber station 105 can be a sending station and the media gateway 135 can be a receiving station. Furthermore, the media gateway 135 or the application server 140 can be a sending station, and the access point 115 or the subscriber station 105 can be a receiving station.

At step 210, the sending station determines a current packet loss value of the data stream. For example, the access point 115 can determine a current packet loss value of the data stream over a virtual tunnel between the access point 115 and the media gateway 135, by determining a number of lost data packets in the data stream within a first time limit.

The lost data packets are data packets in the data stream that are not acknowledged by a receiving station within a second time limit. For example, the application server 140 sends an acknowledgement to the access point 115 for each data packet it receives from the access point 115. In addition, each data packet has a time-to-live value, within which time the data packet must be delivered. If a data packet does not arrive at a receiving station, the data packet will be dropped by an intermediate node. Thus, by determining a number of data packets that are not acknowledged by the application server 140, the access point 115 can determine the current packet loss value of the data stream.

At step 220, the sending station decreases a current packet size of the data stream if the current packet loss value of the data stream is larger than a predefined data packet loss value. According to an embodiment of the present invention, the predefined data packet loss value is zero. For example, if the current packet loss value of the data stream is larger than zero, the access point 115 can reduce the current packet size of the data stream.

One of the reasons for reducing the current packet size is that smaller data packets are more likely to pass through a congested communication network and arrive at a receiving station within their time-to-live than larger data packets. For example, a data packet of 2000 bits and its time-to-live value set to 1 second will be dropped if it is transmitted over a PPP link having data rate of 1000 bits per second (bps), but a data packet of 500 bits can be transmitted over the same PPP link without being dropped.

At step 225, a test data stream of test data packets is transmitted if the current packet size of the data stream is less than a preferred packet size. For example, if the access point 115 determines that the current packet size of the data stream is less than a packet size that is normally compatible with a non- congested PPP data link layer, the application server 140 can transmit a test data stream of test data packets that are larger than the current packet size of the data stream to the media gateway 135 or the application server 140. This way the access point 115 can test whether the communication network 100 is still congested and/or whether a larger packet size than the current packet size of the data stream can be transmitted to the application server 140 without packet loss. At step 230, a packet loss value of the test data stream is determined.

For example, the access point 115 can determine the packet loss value of the test data stream over the communication network 100 by determining a number of lost test data packets in the test data stream within a first time limit. The lost test data packets can be obtained by determining data packets in the test data stream that are not acknowledged by a receiving station, such as the application server 140, within a second time limit.

At step 235, the current packet size of the data stream is increased if the packet loss value of the test data stream is less than a predefined test packet loss value. For example, if the packet loss value of the test data stream is zero, the access point 115 can assume that larger data packets can be transmitted over the communication network 100 because the packet size of the test data stream is larger than the current packet size of the data stream. Thus, the access point 115 can increase the current packet size of the data stream, for example, to the packet size of the test data stream. Finally, the sending station can repeat the steps of 205, 210, and 215, and/or the steps of 205, 220, 225, and 230 until the communication of the data stream is completed. For example, the access point 115 can keep decreasing the current packet size of a data stream over a congested virtual tunnel until no more packet loss is observed. While the current packet size of the data stream is less than the preferred packet size, the access point 115 can transmit a test data stream of test data packets over the virtual tunnel to determine network congestion and/or data transfer rate of the virtual tunnel by determining a packet loss value of the test data stream^ Thus, the access point 115 also can keep increasing the current packet size of the data stream as the network congestion and/or the data transfer rate of the virtual tunnel are improved. This means that the access point 115 can achieve an optimal packet size of a data stream over a virtual tunnel.

According to an embodiment of the present invention, the sending station adjusts the current packet size of the data stream dynamically by decreasing the current packet size while the current packet loss value of the data stream is larger than the predefined data packet loss value, and by increasing the current packet size of the data stream while the packet loss value of the test data stream is less than the predefined test data packet loss value.

Advantages of the invention include an improved method of communicating a data stream of data packets over a communication network by dynamically adjusting the current packet size of the data stream. This is because the invention achieves reduced packet loss of the data stream while keeping current packet size of the data stream as close as possible to the preferred packet size, which is normally a packet size that is compatible with an underlying data link layer of a communication network. If the current packet size is too small, data transmission at an underlying communication layer, such as the Ethernet data link layer, becomes inefficient. Reduced packet loss means less frequent retransmission of the data packets that are dropped by an intermediate node. Reduced packet loss also means increase data rate as more packets will be arriving at a receiving station rather than being dropped. Overall network resources therefore can be conserved. In particular, the present invention can thus increase overall quality of service (QoS) at receiving stations, such as the application server 140 or the subscriber stations 105, and reduced network resource requirement at sending stations and intermediate network nodes.

Throughout the specification the aim has been to describe the invention without limiting the invention to any one embodiment or specific collection of features. Persons skilled in the relevant art may realize variations from the specific embodiments that will nonetheless fall within the scope of the invention. For example, some or all functions could be implemented in custom logic on one or more of field-programmable gate arrays (FPGAs) or application specific integrated circuits (ASICs), or in a computer readable code components configured to be executed by a processor. Of course, a combination of the two approaches could be used. It will be appreciated that various other changes and modifications may be made to the embodiment described without departing from the spirit and scope of the invention.

Claims

1.A method of communicating a data stream formed from a plurality of data packets over a communication network, the method including the steps, at a sending station, of: (i) determining a current packet loss value of the data stream; and
(ii) decreasing a current packet size of the data packets of the data stream if the current packet loss value of the data stream is greater than a predefined data packet loss value.
2. The method of claim 1 , the method further including the steps of:
(iii) transmitting a test data stream formed from a plurality of test data packets over the communication network if the current packet size of the data packets forming the data stream is less than a preferred packet size;
(iv) determining a packet loss value of the test data stream; and (v) increasing the current packet size of the data packets forming the data stream is less than a predefined test pack loss value.
3. The method of claim 1 , wherein the current packet loss value of the data stream is determined by calculating a number of lost data packets forming the data stream within a first time period.
4. The method of claim 3, wherein data packets are characterised as lost data packets in the event that a data packet is not acknowledged by a receiving station within a second time period
5. The method of claim 2, wherein the packet loss value of the test data stream is determined by calculating a number of lost test data packets forming the test data stream within a first time period.
6. The method of claim 5, wherein test data packets are characterised as lost test data packets in the event that a data packet is not acknowledged by a receiving station within a second time period.
7. The method of claim 1 , wherein the sending station continuously decreases the current packet size of the data packets forming the data stream by decreasing the current packet size of each data packet whilst the current packet loss value of the data stream is larger than the predefined data packet loss value.
8. The method of claim 2, wherein the sending station continuously adjusts the current packet size of the data packets forming the data stream by increasing the current packet size of each data packet whilst the packet loss value of the test data stream is less than the predefined test data packet loss value.
9. The method of claim 1 , wherein the sending station is in the form of a network router.
PCT/AU2008/000744 2007-06-22 2008-05-27 Method of communicating a data stream over a communication network WO2009000012A1 (en)

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US9888401B2 (en) 2017-04-03 2018-02-06 At&T Intellectual Property I, L.P. Media content delivery over a wireless network

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WO2015080910A1 (en) * 2013-11-27 2015-06-04 At&T Intellectual Property I, L.P. Adaptive pacing of media content delivery over a wireless network
US9237467B2 (en) 2013-11-27 2016-01-12 At&T Intellectual Property I, L.P. Adaptive pacing of media content delivery over a wireless network
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