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本文(ITU-T Y 1531-2007 SIP-based call processing performance《基于会话初始协议(SIP)的呼叫处理性能 12号研究组》.pdf)为本站会员(terrorscript155)主动上传,麦多课文库仅提供信息存储空间,仅对用户上传内容的表现方式做保护处理,对上载内容本身不做任何修改或编辑。 若此文所含内容侵犯了您的版权或隐私,请立即通知麦多课文库(发送邮件至master@mydoc123.com或直接QQ联系客服),我们立即给予删除!

ITU-T Y 1531-2007 SIP-based call processing performance《基于会话初始协议(SIP)的呼叫处理性能 12号研究组》.pdf

1、 International Telecommunication Union ITU-T Y.1531TELECOMMUNICATION STANDARDIZATION SECTOR OF ITU (11/2007) SERIES Y: GLOBAL INFORMATION INFRASTRUCTURE, INTERNET PROTOCOL ASPECTS AND NEXT-GENERATION NETWORKS Internet protocol aspects Quality of service and network performance SIP-based call process

2、ing performance ITU-T Recommendation Y.1531 ITU-T Y-SERIES RECOMMENDATIONS GLOBAL INFORMATION INFRASTRUCTURE, INTERNET PROTOCOL ASPECTS AND NEXT-GENERATION NETWORKS GLOBAL INFORMATION INFRASTRUCTURE General Y.100Y.199 Services, applications and middleware Y.200Y.299 Network aspects Y.300Y.399 Interf

3、aces and protocols Y.400Y.499 Numbering, addressing and naming Y.500Y.599 Operation, administration and maintenance Y.600Y.699 Security Y.700Y.799 Performances Y.800Y.899 INTERNET PROTOCOL ASPECTS General Y.1000Y.1099 Services and applications Y.1100Y.1199 Architecture, access, network capabilities

4、and resource management Y.1200Y.1299 Transport Y.1300Y.1399 Interworking Y.1400Y.1499 Quality of service and network performance Y.1500Y.1599 Signalling Y.1600Y.1699 Operation, administration and maintenance Y.1700Y.1799 Charging Y.1800Y.1899 NEXT GENERATION NETWORKS Frameworks and functional archit

5、ecture models Y.2000Y.2099 Quality of Service and performance Y.2100Y.2199 Service aspects: Service capabilities and service architecture Y.2200Y.2249 Service aspects: Interoperability of services and networks in NGN Y.2250Y.2299 Numbering, naming and addressing Y.2300Y.2399 Network management Y.240

6、0Y.2499 Network control architectures and protocols Y.2500Y.2599 Security Y.2700Y.2799 Generalized mobility Y.2800Y.2899 For further details, please refer to the list of ITU-T Recommendations. ITU-T Rec. Y.1531 (11/2007) i ITU-T Recommendation Y.1531 SIP-based call processing performance Summary ITU

7、-T Recommendation Y.1531 defines three performance parameters that may be used in specifying, measuring, and comparing the speed, accuracy, and dependability of call set-up processing in networks that employ the session initiation protocol (SIP), with other protocols, in establishing and terminating

8、 media sessions (“calls“) between users. The parameters may also be used in call processing performance apportionment or accumulation. This Recommendation does not specify numerical performance values. Source ITU-T Recommendation Y.1531 was approved on 13 November 2007 by ITU-T Study Group 12 (2005-

9、2008) under the ITU-T Recommendation A.8 procedure. ii ITU-T Rec. Y.1531 (11/2007) FOREWORD The International Telecommunication Union (ITU) is the United Nations specialized agency in the field of telecommunications, information and communication technologies (ICTs). The ITU Telecommunication Standa

10、rdization Sector (ITU-T) is a permanent organ of ITU. ITU-T is responsible for studying technical, operating and tariff questions and issuing Recommendations on them with a view to standardizing telecommunications on a worldwide basis. The World Telecommunication Standardization Assembly (WTSA), whi

11、ch meets every four years, establishes the topics for study by the ITU-T study groups which, in turn, produce Recommendations on these topics. The approval of ITU-T Recommendations is covered by the procedure laid down in WTSA Resolution 1. In some areas of information technology which fall within I

12、TU-Ts purview, the necessary standards are prepared on a collaborative basis with ISO and IEC. NOTE In this Recommendation, the expression “Administration“ is used for conciseness to indicate both a telecommunication administration and a recognized operating agency. Compliance with this Recommendati

13、on is voluntary. However, the Recommendation may contain certain mandatory provisions (to ensure e.g. interoperability or applicability) and compliance with the Recommendation is achieved when all of these mandatory provisions are met. The words “shall“ or some other obligatory language such as “mus

14、t“ and the negative equivalents are used to express requirements. The use of such words does not suggest that compliance with the Recommendation is required of any party. INTELLECTUAL PROPERTY RIGHTS ITU draws attention to the possibility that the practice or implementation of this Recommendation ma

15、y involve the use of a claimed Intellectual Property Right. ITU takes no position concerning the evidence, validity or applicability of claimed Intellectual Property Rights, whether asserted by ITU members or others outside of the Recommendation development process. As of the date of approval of thi

16、s Recommendation, ITU had not received notice of intellectual property, protected by patents, which may be required to implement this Recommendation. However, implementers are cautioned that this may not represent the latest information and are therefore strongly urged to consult the TSB patent data

17、base at http:/www.itu.int/ITU-T/ipr/. ITU 2008 All rights reserved. No part of this publication may be reproduced, by any means whatsoever, without the prior written permission of ITU. ITU-T Rec. Y.1531 (11/2007) iii CONTENTS Page 1 Scope 1 2 References. 1 3 Definitions 2 4 Abbreviations and acronym

18、s 2 5 Conventions 2 6 SIP-based call processing function and outcomes 2 6.1 Definition of the call set-up function 3 6.2 Possible outcomes of a call set-up attempt. 3 7 SIP-based call processing performance parameters . 4 7.1 Call set-up delay. 4 7.2 Call misrouting probability. 5 7.3 Call set-up fa

19、ilure probability. 5 8 Security considerations. 6 Bibliography. 7 ITU-T Rec. Y.1531 (11/2007) 1 ITU-T Recommendation Y.1531 SIP-based call processing performance 1 Scope This Recommendation defines three call processing performance parameters call set-up delay, call misrouting probability and call s

20、et-up failure probability on the basis of session initiation protocol (SIP) message transfers that may be observed at calling and called user-network interfaces (UNIs). These parameters are intended to be used in describing the performance of IP-based networks in which SIP IETF RFC 3261 is used to e

21、stablish and terminate media sessions (“calls“) between users. In this context, SIP is regarded not as a purely application layer protocol, but as part of a comprehensive signalling system with associated network transport resource and admission control functions and lower layer protocols. Standardi

22、zed technologies that use SIP in this way include the IP multimedia subsystem (IMS) and IPCablecom 2. As noted in ITU-T I.350, end user concerns about the performance of a function fall in three general categories: speed, accuracy and dependability. These correspond, respectively, with the three gen

23、eral outcomes a discrete function can encounter: successful performance, incorrect performance and non-performance. The parameters call set-up delay, call misrouting probability, and call set-up failure probability address, respectively, these user concerns and corresponding performance outcomes. Th

24、ey thus provide a logically complete (although very basic) characterization of SIP-based call processing performance. The parameters are intended to be used in specifying and comparing user requirements with service provider offerings, and in measuring achieved performance levels. They may also be u

25、sed in apportioning or accumulating performance values among concatenated network sections delimited by SIP-based interfaces. ITU-T G.1040 provides methods for modelling the network contribution to transaction time, and is applicable to aspects of SIP call processing. This Recommendation is focused

26、on one particular SIP function (or “method“): the INVITE function. It does not address other SIP functions or other performance issues, some of which may also be of importance to network providers and users. These functions and issues may be considered in later work.12 References The following ITU-T

27、 Recommendations and other references contain provisions which, through reference in this text, constitute provisions of this Recommendation. At the time of publication, the editions indicated were valid. All Recommendations and other references are subject to revision; users of this Recommendation

28、are therefore encouraged to investigate the possibility of applying the most recent edition of the Recommendations and other references listed below. A list of the currently valid ITU-T Recommendations is regularly published. The reference to a document within this Recommendation does not give it, a

29、s a stand-alone document, the status of a Recommendation. ITU-T I.350 ITU-T Recommendation I.350 (1993), General aspects of quality of service and network performance in digital networks, including ISDNs. _ 1Among the SIP functions not addressed are querying for capabilities; registration; call redi

30、rection, forwarding, and queuing; multicast; forking; session description; offer/answer exchanges; retransmissions; and the security functions, e.g., authorization and authentication. Among the performance issues not addressed are apportionment and accumulation, handling of emergency services, call

31、processing interactions with priority access and restoral, and the impact of call processing on service availability. 2 ITU-T Rec. Y.1531 (11/2007) ITU-T G.1040 ITU-T Recommendation G.1040 (2006), Network contribution to transaction time. ITU-T Y.1540 ITU-T Recommendation Y.1540 (2002), Internet pro

32、tocol data communication service IP packet transfer and availability performance parameters. IETF RFC 3261 IETF RFC 3261 (2002), SIP: Session Initiation Protocol. 3 Definitions This clause is intentionally left blank. 4 Abbreviations and acronyms This Recommendation uses the following abbreviations

33、and acronyms: ACK Acknowledgement B2BUA Back-to-Back User Agent CMP Call Misrouting Probability CMTS Cable Modem Termination System CSD Call Set-up Delay CSFP Call Set-up Failure Probability DNS Domain Name System DSLAM Digital Subscriber Line Access Multiplexer IAD Integrated Access Device IMS IP M

34、ultimedia Subsystem IP Internet Protocol IPS Interruptions per Second LS Location Server MTA Message Transfer Agent NGN Next Generation Network PS Proxy Server PSTN Public Switched Telephone Network SIP Session Initiation Protocol UAC User Agent Client UAS User Agent Server UNI User-Network Interfac

35、e 5 Conventions None. 6 SIP-based call processing function and outcomes It is customary to describe network performance for a specified communication function by first, formally defining the function in terms of reference events observable at the network interfaces of ITU-T Rec. Y.1531 (11/2007) 3 i

36、nterest; second, delineating a set of relevant possible outcomes of an individual “trial performance“ of the function; and finally, defining one or more parameters to characterize the performance attributes of each outcome.2This clause defines the SIP-based call processing function and outcomes. The

37、 SIP-based call processing performance parameters are defined in clause 7. 6.1 Definition of the call set-up function The interfaces of primary interest in this Recommendation are the user-network interfaces (UNIs) that separate the SIP-based customer equipment from the IP network (Figure 1). Such i

38、nterfaces include, for example, the physical interface between a SIP user agent (e.g., SIP phone, residential gateway, IAD, or MTA) on customer premises and the physical link that connects it to an associated network access concentrator (e.g., DSLAM or CMTS). Other interfaces are shown in Figure 1 t

39、o provide transaction details. SIP is assumed to be used at all of the illustrated interfaces.3The reference events of interest in describing call processing performance in SIP-based networks are transfers of SIP messages across the relevant interfaces in accordance with the standardized SIP protoco

40、ls. A SIP message is either a request from a client to a server or a response from a server to a client. Equipment implementing SIP (e.g., SIP phones, access gateways) can function as both servers and clients. The six SIP request messages defined in IETF RFC 3261 are REGISTER, OPTIONS, INVITE, ACK,

41、BYE and CANCEL. Two of these request messages are involved in a normal (successful) call set-up attempt: INVITE and ACK. The standardized SIP response messages are divided into six categories, distinguished by status codes: 1xx (provisional), 2xx (success), 3xx (redirection), 4xx (client error), 5xx

42、 (server error), and 6xx (global failure). One of these response messages is involved in a normal (successful) call set-up attempt: 200 OK. The exact times of occurrence of reference events (e.g., first bit versus last bit of message transfer) are specified using the conventions defined in ITU-T Y.1

43、540. The communication function of interest in this context is call set-up.4Referring to Figure 1, the call set-up function begins when the calling user issues an INVITE request (event A) and ends (successfully) when the called user receives the corresponding ACK to its final 200 OK (event F). The A

44、CK is considered to be part of the call set-up function for the reasons explained in clause 6.2 below. 6.2 Possible outcomes of a call set-up attempt Three possible outcomes of a call set-up attempt are distinguished, by observation of reference events, to provide a basis for the call set-up perform

45、ance parameter definitions.5Successful call set-up is defined to occur when the reference events (A, B, C, D, E, F) occur, as illustrated in Figure 1, within a specified maximum call set-up time, Tm(to be specified). Call misrouting is defined to occur when event D occurs, but events B and C do not

46、occur, within Tm. Call set-up _ 2In statistical terms, the set of outcomes is a sample space and the parameters are random variables defined on it. 3The events depicted in Figure 1 include a DNS query by the callers proxy server and a location service query by the callees proxy server. These exchang

47、es are typical but not necessarily present in every call. Other message exchanges may occur at interfaces between the UNIs during a call set-up attempt. UNI is one of the set of called user interfaces to which the call can properly be routed. 4In this Recommendation, the term call set-up is used in

48、preference to INVITE to emphasize that the defined function includes the calling users ACK to the called users 200 OK. The ACK is described as a separate transaction in IETF RFC 3261. 5Correspondence among the SIP messages used in defining these reference events and associated outcomes can be verifi

49、ed by comparison of header fields. The outcome definitions assume that “forking“ of INVITE messages does not occur. 4 ITU-T Rec. Y.1531 (11/2007) failure is defined to occur when one of the following event sequences is observed during the call set-up attempt: 1) Both events B and D do not occur. 2) Event C occurs, but event D does not. 3) Event E occurs, but event F does not. A fourth outcome category, excluded trial, is defined to omit from network performance assessment any call set-up attempt

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