(19) United States
`(12) Patent Application Publication (10) Pub. No.: US 2003/0131121 A1
`(43) Pub. Date:
`Jul. 10, 2003
`Quittek et al.
`
`US 2003O131121A1
`
`(54) METHOD FOR TIME-SYNCHRONOUS DATA
`TRANSFER
`(75) Inventors: Jurgen Quittek, Heidelberg (DE);
`Cristian Cadar, Heidelberg (DE)
`Correspondence Address:
`SUGHRUE MION, PLLC
`2100 Pennsylvania Avenue, NW
`Washington, DC 20037-3213 (US)
`(73) Assignee: NEC CORPORATION
`(21) Appl. No.:
`10/294,768
`(22) Filed:
`Nov. 15, 2002
`(30)
`Foreign Application Priority Data
`
`Nov. 16, 2001
`Jul. 4, 2002
`
`(DE)..................................... 101 56 115.6
`(DE)..................................... 10230 248.0
`
`Publication Classification
`
`... G06F 15/16
`(51) Int. CI.7.
`(52) U.S. Cl. .............................................................. 709/232
`
`
`
`(57)
`
`ABSTRACT
`
`A method for transferring time-synchronous data, particu
`larly voice and Video data, Over a network, particularly the
`Internet, between at least two terminals, where between the
`terminals a connection is established using a SIP Server and
`where the SIP protocol is used for establishing the connec
`tion, is-with respect to high Quality of Service for trans
`ferring time-synchronous data and with a technically simple
`and cost effective design-developed in a way that the SIP
`Server analyzes the connection and/or the terminals or the
`like and that based on the analysis it determines an ideal
`bandwidth for optimizing the transfer of time-synchronous
`data.
`
`Page 1 of 4
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`GOOGLE EXHIBIT 1011
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`US 2003/O131121 A1
`
`Jul. 10, 2003
`
`METHOD FOR TIME-SYNCHRONOUS DATA
`TRANSFER
`0001. The invention concerns a method for time-synchro
`nous data transfer, particularly of Voice and Video messages,
`over a network, particularly the Internet, between at least
`two terminals where a connection between the terminals is
`set up using the SIP protocol and at least one SIP server.
`0002 Methods for transfer of time-synchronous data,
`Such as voice data, over networks, particularly over the
`Internet, are gaining importance, because for private users as
`well as for enterprises cost Saving on telephone calls are
`possible. For conducting telephone calls between two ter
`minals over a network, particularly using the Internet pro
`tocol, means for Signalling call Set-up and tear-down are
`required. SIP, the Session Initiation Protocol is one of the
`protocols used for this purpose. It was Standardized by the
`IETF, the Internet Engineering Task Force.
`0003) A caller may send a SIP message for setting up a
`call by using his/her terminal. The message notifies the
`callee that the caller intends to Set up a call. The terminal of
`the callee would then for example ring and notify the
`terminal of the caller by another SIP message that ringing
`has Started. If the callee operates her/his terminal Such that
`it accepts the call, then the terminal sends another SIP
`message to the terminal of the caller for notifying it that now
`transmission of time-synchronous data, for example voice or
`video data, can start. The SIP protocol is also used for
`Signalling tear-down of a connection.
`0004 Establishing a concrete voice connection and cod
`ing and Sending time-synchronous voice data is not Sup
`ported by SIP For this, the terminals communicate with each
`other, for example, by negotiating about the kind of con
`nection or data transfer to use and a coding method for voice
`data. SIP Supports establishing a connection insofar, as it
`includes an exchange of terminal properties. This includes
`the kinds of Voice coding that the terminals Support, the
`addresses of the terminals to which voice traffic is to be sent,
`and Some other terminal Specific properties.
`0005 Another functionality of SIP is finding a callee at
`his/her current location. The first message from a caller to a
`callee if typically not sent directly to the callee's terminal,
`but to a SIP server, which is usually configured as SIP proxy
`Server. At this Server, a company XYZ provides an address
`Sip://customer(GXyZ.de to one of its customers.
`0006 Now, the customer can register her/his current
`terminal at the SIP proxy server provided by company XYZ.
`His current terminal might be his work phone, his home
`phone, his mobile phone, or any other SIP-enabled phone.
`The terminal of the caller then sends the first message to
`Sip://customer(GXyZ.de. There the proxy server forwards this
`message to the terminal that the customer has registered. The
`SIP server would also forward the reply of customer's
`terminal in the opposite direction.
`0007 AS for the conventional technique, the following
`references are known.
`0008 Reference for Differentiated Services:
`0009 “RFC 2475 An Architecture for Differentiated
`Service,” S. Blake, D. Black, M. Carlson, E. Davis,
`Z. Wang, W. Weiss, December 1998 (Format: TXT
`94786 bytes) (Updated by RFC3260) (Status:
`INFORMATIONAL)
`
`0010) Reference for RSVP:
`0.011 “RFC 2205 Resource ReServation Protocol
`(RSVP), Version 1 Functional Specification,” R.
`Braden, Ed., L. Zhang, S. BerSon, S. Herzog, S.
`Jamin, September 1997 (Format: TXT-223974
`bytes) (Updated by RFC2750) (Status: PROPOSES
`STANDARD);
`0012 “RFC 2210 The Use of RSVP with IETF
`Integrated Services,” J. Wroclawski, September
`1997 (Format: TXT-77613 bytes) (Status: PRO
`POSED STANDARD)
`0013) Reference for SIP:
`0014) “RFC 3261 SIP: Session Initiation Protocol,”
`J. Rosenberg, H. Schulzrinne, G. Camarillo, A.
`Johnston, J. Peterson, R. Sparks, M. Handley, E.
`Schooler, June 2002 (Format: TXT-647976 bytes)
`(Obsoletes RFC 2543) (Updated by RFC 3265)
`(Status: PROPOSED STANDARD).
`0015 For the known methods of transferring time-syn
`chronous data over the basic Internet, there is the particular
`problem that they give no guarantees for Quality of Service
`(QoS), for example available bandwidth and packet delay of
`a connection. This may lead to bad quality of Voice trans
`mission, because packets containing the coded Voice arrive
`in an order different to the one they were Sent in, because
`packets are damaged or dropped during transfer, or because
`packets are transferred with high delay. For time-synchro
`nous data, Such as voice or Video, this leads to bad QoS. The
`lack of QoS is one of the reasons for the limited acceptance
`of time-synchronous Services, particularly Internet tele
`phony, so far. Also SIP does not have any built-in mecha
`nism to support Quality of Service (QoS) to the time
`Synchronous data transfer it signals.
`0016 Enhancements of the basic Internet, such as Inte
`grated Services and Differentiated Services support QoS for
`Internet connections, but it requires additional Signalling and
`network management functions. Integrating SIP Signalled IP
`telephony or video transfer with these methods for QoS
`provisioning would be a significant technology improve
`ment and it would increase the acceptance of Internet
`telephony, but its available is very limited, So far.
`0017 Existing suggestions on how to perform this inte
`gration are based on the idea that the telephony terminals
`themselves try to reserve resources for their calls by using
`other means of Signalling, independent of SIP. An example
`is the Resource reServation Protocol (RSVP) of Integrated
`Services. However, this approaches do not Scale Sufficiently
`with an increasing number of users, Such that existing QoS
`provisioning Systems cannot deal with a high rate of reser
`Vation requests.
`0018. Therefore, this invention is targeted at defining a
`method for transferring time-synchronous data with high
`quality, which is technically simple and cost-effective.
`0019. According to the invention, the goal described
`above is achieved by a method for transferring time-Syn
`chronous data according to patent claim 1. This method is
`designed and developed in a way that the SIP server ana
`lyzes the connections and/or the terminals in a way that
`based on this analysis an ideal bandwidth for optimizing the
`transfer of time-synchronous data is determined.
`
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`US 2003/O131121 A1
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`Jul. 10, 2003
`
`0020. Within the invention it is found that for achieving
`sufficient QoS, no additional functionality is required for the
`terminals and/or the network, but that the required QoS can
`be achieved in a particularly simple and cost-effective way
`by extending the SIP Server Such that the connection, par
`ticularly the SIP message exchanged for Signalling, and or
`the terminals are analyzed. Based on this analysis and with
`knowledge of the coding and decoding methods, an optimal
`bandwidth for optimizing the transfer of time-synchronous
`data is determined. A Sufficient QoS for the data transfer can
`therefore be achieved without requiring additional function
`ality of terminals or the network, which Saves costs signifi
`cantly.
`0021 Now, in a very advantageous way, based on the
`analysis of the connection and or the terminals, a bandwidth
`reservation for the transfer of time-synchronous data can be
`made for ensuring QoS. The reservation of bandwidth could
`be performed by the SIP server, which is extended by this
`functionality. The QoS would be ensured with respect to loss
`of data during data transfer, as far as the transfer data rate is
`within the reserved bandwidth. Particularly for voice and
`Video transmission, this method ensures good QoS.
`0022. In order to determine the required bandwidth to be
`reserved in a very simple way, at the analysis, the properties
`of the connection and/or the terminals and/or the used
`coding method and/or the used decoding method for time
`Synchronous data transfer could be determined. The analysis
`of the used coding method and/or decoding method is very
`Simple, Since these methods are mentioned explicitly in the
`SIP messages.
`0023. In an again very simple way, the reservation of
`bandwidth could be performed using a QoS management
`System. The QoS management System could be external to
`the SIP server and receive reservation requests from the SIP
`Server. The QoS management System could, for example, be
`designed as a bandwidth broker, also called QoS Server.
`Then bandwidth broker performs the complex task of for
`warding the reservations to the individual devices in the
`network.
`0024 Now, if several connections between different
`devices are established, then these bandwidth reservations
`for different connections can be aggregated to at least one
`traffic trunk, preferably by the SIP server. This could be used
`for achieving scalability of the methods described above in
`the network.
`0025. With respect to high flexibility, the bandwidth of
`the traffic trunk could be chosen larger than the actual
`bandwidth required by the connections. This would allow
`quickly providing bandwidth to new connections to be
`established, without requesting additional bandwidth at the
`bandwidth management System.
`0026. With respect to an effective reservation of band
`width, a traffic trunk could aggregate reservations between
`two end points, particularly access routers or edge routers.
`Access routers or edge routers could be devices connecting
`a large number of terminals to the Internet.
`0027. With respect to high flexibility concerning new
`connections to be established, a new connection between a
`first terminal and a Second terminal via at least two end
`points could be mapped to the respective traffic trunk. This
`trunk would be the already existing traffic trunk between the
`
`two end points. This would largely avoid the creation of new
`traffic trunk at connection establishment as well as the
`related effort.
`0028. For ensuring good QoS, the SIP server could reject
`establishment of a new connection, if the required band
`width of the new connection exceeds the remaining avail
`able bandwidth of the trunk. This would ensure that the QoS
`of already existing connections would not be deteriorated.
`0029) Alternatively, the SIP server could reserve addi
`tional bandwidth for the concerned traffic trunk, if the
`required bandwidth of a new connection to be established
`exceeds the remaining available bandwidth of the trunk.
`This would ensure that very few requested connections
`would be rejected, and that in most cases a connection would
`be established.
`0030. With respect to cost-effectiveness, the SIP server
`could reduce the reserved bandwidth for a traffic trunk if the
`bandwidth required by the connections is much less that the
`reserved bandwidth.
`0031. With respect to flexibility, the reservation of band
`width, particularly between all end points server by the SIP
`Server, could be performed before any connection is estab
`lished. This can be based on an analysis of reservations in
`the past.
`0032. When the invented method is available, terminals
`have-with Some restrictions-a free choice of the SIP
`server to use. The SIP server is realized In general as a SIP
`proxy server. A service provider could offer two different SIP
`Servers, one offering QoS according to the methods
`described above, and a conventional one without this func
`tion. Then the Service provider could charge a higher price
`for connections with QoS guarantees, while also offering
`connections without QoS guarantees for a lower price or free
`of charge to users who do not require QoS guarantees.
`0033 Several variants of the invention can be imple
`mented. Particularly, for the selection of the SIP server that
`performs the analysis, particularly of the SIP messages, and
`the reservation of resources for a traffic trunk as well as the
`modifications of reservations for the traffic trunks. The SIP
`server could be selected out of a chain of SIP servers
`involved in forwarding SIP Signalling messages. The reser
`vation and modification of reservation for traffic trunks
`could be based on the observation of actual observed usage
`of resources.
`0034. Another variant would be an explicit signalling of
`QoS requirements of coding methods and decoding methods
`used by the terminals to the SIP server. For this purpose, for
`example an extension of the SIP protocol could be defined
`and used. This would imply that the SIP server no longer
`would need to analyze SIP messages with respect to prop
`erties of the terminals, particularly concerning the used
`coding method in order to determine the required QoS
`parameters of the connection. Instead, it would receive these
`parameters directly from the terminals using the SIP exten
`Sions for Signalling.
`0035. The invention offers a technically feasible and
`Scalable way of integrating SIP-Signalled transfer of time
`Synchronous data with QoS guarantees in networkS. Prefer
`ably, the QoS parameters are derived from an analysis of SIP
`messages while the messages are forwarded by the SIP
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`US 2003/O131121 A1
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`Jul. 10, 2003
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`Server. An advantage of the invention is that no extensions
`of the SIP protocol are required for offering QoS guarantees.
`ReSource reservations for individual connections are aggre
`gated by the SIP server into traffic trunks. These traffic
`trunkS can then be adapted to the observed usage of the
`trunks. This way, the invention is a highly Scalable method
`that can be used for installing large Internet telephony
`networks. The investment in the new technology is relatively
`small, because only the SIP server needs to be extended. A
`change of the terminals is not required.
`1. Method of transmitting time-synchronous data, particu
`larly voice and/or Video data, over a network, particularly
`over the Internet, between at least two terminals, where
`between the terminals a connection is established using a
`SIP server and where the SIP server and where the SIP
`protocol is used for establishing the connection, wherein the
`SIP server analyzes the connection and/or the terminals such
`that based on this analysis and ideal bandwidth is deter
`mined for optimizing the transfer of time-synchronous data.
`2. Method as claimed in claim 1, wherein based on the
`analysis of the connection and/or the terminals a reservation
`of bandwidth for the transfer of time-synchronous data is
`made for guaranteeing Quality of Service (QoS).
`3. Method as claimed in claim 2, wherein at the analysis
`the properties of the connection and/or the terminals and/or
`the used coding method and/or the used decoding method
`are determined.
`4. Method as claimed in claim 3, wherein the reservation
`of bandwidth by the SIP server is performed using a QoS
`management System.
`5. Method as claimed in claim 4, wherein the reservations
`of bandwidth for different connections, preferably by the SIP
`Server, are aggregated into at least one traffic trunk.
`
`6. Method as claimed in claim 5, wherein the bandwidth
`of the traffic trunk is chosen larger than the actual required
`bandwidth.
`7. Method according to claim 5, wherein, in traffic trunks,
`the reservations between two end points, preferably access
`routers or edge routers, are aggregated.
`8. Method as claimed in claim 7, wherein a new connec
`tion being established between a first terminal and a Second
`terminal via at least two endpoints is mapped to the respec
`tive traffic trunk during establishment.
`9. Method as claimed in claim 8, wherein additional
`connections are rejected by the SIP server if the required
`bandwidth of the additional connection exceeds the remain
`ing available bandwidth of the traffic trunk.
`10. Method as claimed in claim 8 and where applicable to
`claim 4, wherein the SIP server requests additional band
`width, preferably using a QoS management System and
`particularly for the affected traffic trunk, if the required
`bandwidth of the additional connection exceeds the remain
`ing available bandwidth of the traffic trunk.
`11. Method according to claim 5, wherein the SIP server
`reduces the reserved bandwidth for a traffic trunk if the
`bandwidth required by the connections is much less than the
`reserved bandwidth.
`12. Method according to claim 5, wherein the reservation
`of bandwidth, preferably between all end points served by
`the SIP server, is performed before connections are estab
`lished.
`
`Page 4 of 4
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