WO2000041527A2 - Method and apparatus for providing efficient multiplexing between gateways using dynamic timers - Google Patents
Method and apparatus for providing efficient multiplexing between gateways using dynamic timers Download PDFInfo
- Publication number
- WO2000041527A2 WO2000041527A2 PCT/US2000/000607 US0000607W WO0041527A2 WO 2000041527 A2 WO2000041527 A2 WO 2000041527A2 US 0000607 W US0000607 W US 0000607W WO 0041527 A2 WO0041527 A2 WO 0041527A2
- Authority
- WO
- WIPO (PCT)
- Prior art keywords
- buffer
- packetization
- timer associated
- packets
- samples
- Prior art date
Links
Classifications
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04J—MULTIPLEX COMMUNICATION
- H04J3/00—Time-division multiplex systems
- H04J3/24—Time-division multiplex systems in which the allocation is indicated by an address the different channels being transmitted sequentially
- H04J3/247—ATM or packet multiplexing
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L12/00—Data switching networks
- H04L12/64—Hybrid switching systems
- H04L12/6418—Hybrid transport
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L12/00—Data switching networks
- H04L12/64—Hybrid switching systems
- H04L12/6418—Hybrid transport
- H04L2012/6445—Admission control
- H04L2012/6459—Multiplexing, e.g. TDMA, CDMA
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L12/00—Data switching networks
- H04L12/64—Hybrid switching systems
- H04L12/6418—Hybrid transport
- H04L2012/6472—Internet
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L12/00—Data switching networks
- H04L12/64—Hybrid switching systems
- H04L12/6418—Hybrid transport
- H04L2012/6481—Speech, voice
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L12/00—Data switching networks
- H04L12/64—Hybrid switching systems
- H04L12/6418—Hybrid transport
- H04L2012/6489—Buffer Management, Threshold setting, Scheduling, Shaping
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04W—WIRELESS COMMUNICATION NETWORKS
- H04W88/00—Devices specially adapted for wireless communication networks, e.g. terminals, base stations or access point devices
- H04W88/16—Gateway arrangements
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04W—WIRELESS COMMUNICATION NETWORKS
- H04W92/00—Interfaces specially adapted for wireless communication networks
- H04W92/16—Interfaces between hierarchically similar devices
- H04W92/24—Interfaces between hierarchically similar devices between backbone network devices
Definitions
- This invention relates in general to a IP telephone, and more particularly to a method and apparatus for providing efficient multiplexing between gateways using dynamic timers
- PSTN Public switched telephone networks
- IP Internet Protocol
- IP packets are routed to the destination IP address contained within the header of each packet. Packets may travel over separate network paths before arriving at their final destination for reassembly and resequencing.
- the transmission speed between any two users can change dramatically based on the dynamic number of users sharing the common transmission medium, their bandwidth requirements, the capacity of the transmission medium, and the efficiency of the network routing and design.
- the IP is a network layer protocol that routes data across an Internet.
- the Internet Protocol was designed to accommodate the use of host and routers built by different vendors, encompass a growing variety of growing network types, enable the network to grow without interrupting servers, and support higher-layer of session and message-oriented services.
- the IP network layer allows integration of Local Area Network "islands".
- the Internet not only provides a medium for data transport, but also has developed as a medium for telecommunications.
- IP telephony is maturing as a technology and a service, which places it in direct conflict with the conventional public telephone network. New types of IP telephony equipment are being introduced and small Internet service providers and niche carriers are beginning to offer IP telephony services.
- IP telephony or voice-over-IP
- voice-over-IP has great potential to compete against the traditional public voice network
- many obstacles must first be overcome.
- the quality of Internet telephone calls is not as good as public network calls.
- IP telephony network equipment is new and lacks features that carriers desire.
- IP is the dominant transport protocol in access networks.
- the penetration of Internet and subsequent IP connectivity to homes and businesses have given impetus to integrated services over IP which means voice, data and video over a single IP network.
- IP networks offer a single class of service called best effort, which can not guarantee any Quality of Service (QoS) to applications.
- QoS Quality of Service
- IETF Internet Engineering Task Force
- IP telephone has emerged as a potential application to challenge the traditional phone companies by offering long distance telephone service over Internet for low prices.
- IP telephone gateways and accessories to provide IP telephony service to corporate customers and Internet Service Providers (ISPs).
- IP telephone standards such as H.323, H.225 and H.245 have been standardized to enhance the rapid deployment of IP telephone services in global Internet.
- IP telephone is not a reality in public Internet today, it has been successful in Intranet and Virtual Private Networks (VPN) environments.
- VoIP Virtual Private Networks
- voice is carried in a circuit switched network with connection oriented model.
- the explosive growth of data traffic in the network has changed the notion of service specific networks (single network for single service).
- Voice over IP network enables the voice traffic from public branch exchanges (PBX) and public switched telephone network (PSTN) users to share the data network thus improving the network utilization and lowering the cost associated with long distance telephone network.
- PBX public branch exchanges
- PSTN public switched telephone network
- IP telephone gateways act as an interface between the existing PSTN and PBX networks and IP networks. This method allows one PSTN user to call another PSTN user connected through IP telephone gateways thus eliminating the need for long distance telephone network.
- RTP Real-time Transport Protocol/User Datagram Protocol/Internet Protocol
- RTP is an Internet protocol for transmitting real-time data such as audio and video.
- RTP itself does not guarantee real-time delivery of data, but it does provide mechanisms for the sending and receiving applications to support streaming data.
- RTP runs on top of the UDP protocol, although the specification is general enough to support other transport protocols.
- the User Datagram Protocol is a connectionless protocol that, like TCP, runs on top of IP networks. Unlike TCP/IP, UDP/IP provides very few error recovery services, offering instead a direct way to send and receive datagrams over an IP network.
- IP telephony gateways provide an interface between the existing circuit switched telephone networks (such as PSTN and GSM) and the packet switched IP data networks.
- PSTN and GSM circuit switched telephone networks
- IP telephony gateways In traditional IP telephony applications, telephone calls between PSTN users interconnected by a pair of IP telephony gateways to compress incoming PSTN speech samples generate packets with sizes ranging from 5 to 20 bytes per speech sample.
- G.723.1 the most popular IP telephony codec and the
- VoIP Voice over IP
- IMTC International Multimedia Teleconferencing Consortium's
- VoIP Voice over IP
- IMTC Voice over IP
- Many codecs used in cellular environment generate less than 10 byte packet per speech sample.
- Small size packets are subjected to large overhead when transferred using the Real time Transport Protocol (RTP).
- RTP/UDP/IP overhead is 40 bytes (12+8+20) for a simple speech packet.
- a 10 byte packet transferred via RTP/UDP/IP increases the overhead to 80% (40 byte overhead/50 byte overhead plus packet).
- a single UDP/IP connection (a pair of UDP ports) is established between the gateways requiring a large state (memory) to be maintained at the telephony gateways, thereby making these less scaleable.
- a PSTN-IP or GSM-IP gateway is needed at the network boundaries. In such a case, it is advantageous to multiplex several users (or voice calls) traversing the same pair of telephone gateways.
- RTCP Real Time Control Protocol
- the present invention discloses a flexible mechanism employing timers within the gateways to facilitate efficient multiplexing.
- the present invention solves the above-described problems by providing timers, wherein their value can be adjusted on a dynamic basis depending on factors such as network congestion that impact end-to-end delay.
- the extraction of data from the buffers is triggered when either the timer expires or when the accumulated data reaches a certain size.
- a method in accordance with the principles of the present invention includes receiving a stream of data packets on N channels, storing the packets for the Nth channel in an Nth input buffer having a timer associated therewith, moving the packets in the Nth input buffer to a packetization buffer having a timer associated therewith in response to the timer associated with the Nth input buffer, and transmitting the packets in the packetization buffer in response to the timer associated with the packetization buffer.
- timer associated with the packetization buffer is set to meet a first set of criteria and to meet an end-to-end delay associated with the packets being transmitted.
- timer associated with the Nth input buffer is set based on the setting of the timer associated with the packetization buffer and the end-to-end delay.
- the first set of criteria comprises maximizing the time the packets are held in the Nth input buffer, moving the packets in the Nth input buffer to the packetization buffer according to a first threshold, and transmitting the packets in the packetization buffer according to a second threshold.
- the first threshold comprises expiration of the timer associated with the Nth input buffer.
- the timer associated with the Nth input buffer is set at a value that expires before the Nth input buffer overflows.
- the first threshold comprises detection of an overflow condition for the Nth input buffer.
- the second threshold comprises expiration of the timer associated with the Nth input buffer.
- the second threshold comprises a maximum packet payload size, the packets in the packetization buffer being transmitted when the packets in the packetization buffer exceeds the maximum packet payload size.
- FIG. 1 illustrates one possible embodiment of an advanced communication network
- Fig. 2 illustrates a system block diagram showing the use of gateways between a circuit-switched network (CSN) and a packet-switched network (PSN); and Fig. 3 illustrates a PSTN-IP/Mobile-IP gateway depicting the various buffers and timers according to the present invention.
- CSN circuit-switched network
- PSN packet-switched network
- Fig. 3 illustrates a PSTN-IP/Mobile-IP gateway depicting the various buffers and timers according to the present invention.
- the present invention provides a flexible mechanism employing timers within the gateways to facilitate efficient multiplexing.
- the timers include values that can be adjusted on a dynamic basis depending on factors such as network congestion that impact end-to-end delay.
- the extraction of data from the buffers is triggered when either the timer expires or when the accumulated data reaches a certain size.
- Fig. 1 illustrates one possible embodiment of an advanced communication network 100, e.g., a third generation GSM evolution. Those skilled in the art will recognize that the present invention is not meant to be limited to use with GSM mobile communication systems, but is applicable to other mobile communication systems. However, the present invention will be described herein using GSM as an example.
- GRAN Generic Radio Access Network
- SGM/UMTS core network which has been evolved from the GSM core network by integrating new third generation capabilities.
- the evolved GSM network elements are referred to as 3G MSC and 3G SGSN.
- a mobile unit 110 receives and sends signals to a base station (BS) 112.
- Base stations 112 are in turn coupled to a radio network controller (RNC) 114 in the radio access network (RAN) 116.
- the RAN interfaces with GSM/UMTS core network 120 via the lu-interface 122, which corresponds to the GSM A-interface and GPRS Gb-interface.
- radio access 130 is isolated from the core network 120, and the goal is that the GSM/UMTS core network would have the flexibility to support any radio access scheme.
- Circuit switched services are routed via the GSM MSC 140, and the packet switched services via the GPRS part 150 of the GSM/UMTS core network.
- Fig. 2 illustrates a system block diagram 200 showing the use of gateways 210, 212 between a circuit-switched network (CSN) 220 and a packet-switched network (PSN) 230.
- CSN circuit-switched network
- PSN packet-switched network
- Multiple users traversing the same pair of gateways 210, 212 are multiplexed between the two gateways 210, 212 for traversal over the PSN 230.
- Multiplexing voice streams 240 between IP telephony gateways 210, 212 needs to be done in an efficient manner while not compromising the user's performance requirements.
- Fig. 3 illustrates a PSTN-IP/Mobile-IP gateway 300 depicting the various buffers and timers according to the present invention.
- multiple streams 310, 312, 314 are received at corresponding processing units (PU) 320, 322, 324.
- Buffers 330, 332, 334 are associated with each voice stream 310, 312, 314.
- Timers 340, 342, 344 are associated with each buffer 330, 332, 334 for providing efficient multiplexing between gateways 210, 212 as shown in Fig. 2.
- the buffers 330, 332, 334 provide packets to the multiplexing controller unit (CU) 350.
- CU multiplexing controller unit
- the multiplexing controller unit 350 then packetizes the voice streams 310, 312, 314 at a packetization buffer 360 having a packetization timer 370 associated therewith.
- the times 340, 342, 344 of the PSTN-IP/Mobile-IP gateway 300 provide efficiency in multiplexing voice steams between such IP gateways.
- the network operator may choose the value based on the known approximation of end-to-end delay.
- the multiplexing controller 350 has the capability to extract the network delay information from Real-Time Control Protocol (RTCP) reports.
- RTCP Real-Time Control Protocol
- Fig. 3 the gateway functionality is not illustrated in its entirety, but is drawn merely for illustration purposes.
- the buffer organization could be arbitrary, so that the buffers could either be physically separate buffers or logical buffers that use some dynamic sharing principle.
- Each incoming voice stream x 310-314 is processed by a processing unit 320-324 prior to being stored in buffer B x 330-334. Such processing could include transcoding and encryption.
- each voice stream buffer or input buffer B x 330-334 is a timer T x 340-344, while a timer T p 370 is associated with the packetization buffer B p 360.
- the control unit (CU) 350 moves voice samples from the voice stream buffers 330-334 to the packetization buffer 360.
- the control unit 350 prepends a suitable header to the voice sample(s) taken from the voice stream buffers 330-334 prior to storing it in the packetization buffer 360.
- the voice stream timers T x 340-344 are reinitialized and the packetization timer T p 370 is suitably adjusted. Based on the value of the timer T p 370 and on the occupancy of the packetization buffer 360, data from the packetization buffer 360 is sent out into the network after suitably prepending it with the necessary headers. Upon doing so, the timer T p 370 is re-initialized.
- the total end-to-end delay D experienced by a packet equals the sum of the delay on the circuit switched network ( c ) 250, the delay within the voice stream buffer (d vb ) 252, the delay within the packetization buffer (d pb ) 254, other processing delays within the initiating gateway (d p/ ) 256, transmission delay from the initiating gateway onto the network (d t ) 258, propagation delay over the packet switched network (d p ) 260, and all processing delays in the remote gateway (dps) 262.
- the delay over the circuit switched network d c 250 can be determined and is relatively unchanging. Processing delays at the gateways while depending on the load at the gateways, do not vary significantly. Hence, d p , 256 and d p2 258 are relatively constant and can be determined.
- the transmission delay d, 258 depends on the packet size and the link speed, and the propagation delay d p 260 depends on several factors including the congestion in the network. Feedback about congestion in the network can be obtained from several mechanisms including RTCP (Real Time Control Protocol) reports, thus giving a handle on d p 260.
- RTCP Real Time Control Protocol
- d n d p + d 1 and d n is the network delay, d p is the Propagation delay, d? is the Transmission delay, and f(.) denotes some function.
- the network delay value d n is extracted from the RTCP reports and used to set the value for c (in essence values for ⁇ 252 and d pb 254).
- c in essence values for ⁇ 252 and d pb 254.
- Voice samples should be held in the voice buffers 330-334 for as long as possible prior to moving them to the packetization buffer 360.
- a larger amount of data i.e., several voice samples from the same source
- when moved from the voice buffers 330-334 to the packetization buffer 360 incurs less overhead due to a single header being prepended to a larger amount of data.
- Voice samples are moved from the voice buffers 330-334 to the packetization buffer 360 as soon as the corresponding timers 340-344 expire or when the amount of data in the voice buffers 330-334 is such that a new incoming sample would result in the data in the voice buffers 330-334 exceeding that which can be handled by the length indicator in the prepended header.
- voice samples placed in a voice buffers 330-334 should not exceed the buffer size.
- the buffer timers 340-344 should be set such that the voice samples are removed before an overflow occurs.
- Data from the packetization buffer 360 is transmitted onto the network as soon as the timer T p 370 expires or when the amount of data in the packetization buffer 360 just exceeds the maximum packet payload size.
- the new incoming data from the voice buffers 330-334 that resulted in the packet payload size exceeding the limit is transmitted in the next packet.
- the maximum packet payload size is governed by the MTU (maximum transmission unit) within the network. This is usually 1500 bytes for IP datagrams over the public Internet.
- the common packetization buffer timer T p 370 is established with the end-to- end delay D x of each voice stream x and the above specified guidelines in mind. Having established the value that T p 370 needs to be initialized with, the value that each voice buffer timer T x 370 is initialized with is ascertained merely by subtracting T p 370 from the known constant c associated with the particular voice stream x 310- 314. When the first voice sample for voice stream x 310-314 is stored in the corresponding voice stream buffer B x 330-334, the timer Tx 340-344 associated with the voice stream buffer 330-334 is re-initialized.
- the samples are transferred from the voice buffers 330-334 to the packetization buffer 360 after prepending the suitable header.
- the timers T x 340-344 associated with the voice buffers B x 330-334 are is then reinitialized.
- the data from the packetization buffer 360 is sent out to the network when the timer T p 370 expires or when the amount of data within the packetization buffer 360 matches the MTU (taking into account packetization headers).
- the timer T p 370 is re-initialized.
Abstract
Description
Claims
Priority Applications (2)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
EP00903219A EP1145509A2 (en) | 1999-01-12 | 2000-01-11 | Method and apparatus for providing efficient multiplexing between gateways using dynamic timers |
AU24996/00A AU2499600A (en) | 1999-01-12 | 2000-01-11 | Method and apparatus for providing efficient multiplexing between gateways usingdynamic timers |
Applications Claiming Priority (2)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
US22872399A | 1999-01-12 | 1999-01-12 | |
US09/228,723 | 1999-01-12 |
Publications (2)
Publication Number | Publication Date |
---|---|
WO2000041527A2 true WO2000041527A2 (en) | 2000-07-20 |
WO2000041527A3 WO2000041527A3 (en) | 2000-11-30 |
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Application Number | Title | Priority Date | Filing Date |
---|---|---|---|
PCT/US2000/000607 WO2000041527A2 (en) | 1999-01-12 | 2000-01-11 | Method and apparatus for providing efficient multiplexing between gateways using dynamic timers |
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Country | Link |
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EP (1) | EP1145509A2 (en) |
AU (1) | AU2499600A (en) |
WO (1) | WO2000041527A2 (en) |
Cited By (4)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
EP1168757A1 (en) * | 2000-06-20 | 2002-01-02 | TELEFONAKTIEBOLAGET L M ERICSSON (publ) | Packet multiplexing using a dynamic buffer delay timer |
EP1296488A2 (en) * | 2001-09-20 | 2003-03-26 | Lucent Technologies Inc. | Multiplexing IP data packets within a MPLS frame |
WO2004054182A1 (en) * | 2002-12-12 | 2004-06-24 | Koninklijke Philips Electronics N.V. | A system and method for adapting transmission rate of a multimedia streaming server using a 'virtual clock' |
EP1707446A2 (en) | 2005-04-01 | 2006-10-04 | Keihin Corporation | Communication control apparatus and communication control method |
Citations (1)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
US5541926A (en) * | 1992-10-02 | 1996-07-30 | Kabushiki Kaisha Toshiba | ATM cell assembly and disassembly device with enhanced data handling flexibility |
-
2000
- 2000-01-11 AU AU24996/00A patent/AU2499600A/en not_active Abandoned
- 2000-01-11 EP EP00903219A patent/EP1145509A2/en not_active Withdrawn
- 2000-01-11 WO PCT/US2000/000607 patent/WO2000041527A2/en not_active Application Discontinuation
Patent Citations (1)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
US5541926A (en) * | 1992-10-02 | 1996-07-30 | Kabushiki Kaisha Toshiba | ATM cell assembly and disassembly device with enhanced data handling flexibility |
Non-Patent Citations (2)
Title |
---|
KITAMURA Y ET AL: "IMPLEMENTATION OF AAL2 FOR LOW BIT-RATE VOICE OVER ATM" ISS. WORLD TELECOMMUNICATIONS CONGRESS. (INTERNATIONAL SWITCHING SYMPOSIUM),CA,TORONTO, PINNACLE GROUP, 21 September 1997 (1997-09-21), pages 271-276, XP000720533 * |
NAKAMURA H ET AL: "APPLYING ATM TO MOBILE INFRASTRUCTURE NETWORKS" ISS. WORLD TELECOMMUNICATIONS CONGRESS. (INTERNATIONAL SWITCHING SYMPOSIUM),CA,TORONTO, PINNACLE GROUP, 21 September 1997 (1997-09-21), pages 73-80, XP000720509 * |
Cited By (7)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
EP1168757A1 (en) * | 2000-06-20 | 2002-01-02 | TELEFONAKTIEBOLAGET L M ERICSSON (publ) | Packet multiplexing using a dynamic buffer delay timer |
EP1296488A2 (en) * | 2001-09-20 | 2003-03-26 | Lucent Technologies Inc. | Multiplexing IP data packets within a MPLS frame |
EP1296488A3 (en) * | 2001-09-20 | 2003-05-14 | Lucent Technologies Inc. | Multiplexing IP data packets within a MPLS frame |
WO2004054182A1 (en) * | 2002-12-12 | 2004-06-24 | Koninklijke Philips Electronics N.V. | A system and method for adapting transmission rate of a multimedia streaming server using a 'virtual clock' |
EP1707446A2 (en) | 2005-04-01 | 2006-10-04 | Keihin Corporation | Communication control apparatus and communication control method |
JP2006282053A (en) * | 2005-04-01 | 2006-10-19 | Keihin Corp | Communication control device and communication control method |
EP1707446A3 (en) * | 2005-04-01 | 2010-12-01 | Keihin Corporation | Communication control apparatus and communication control method |
Also Published As
Publication number | Publication date |
---|---|
WO2000041527A3 (en) | 2000-11-30 |
EP1145509A2 (en) | 2001-10-17 |
AU2499600A (en) | 2000-08-01 |
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