1、 ETSI TR 102 793 V1.2.1 (2010-11)Technical Report Speech and multimedia Transmission Quality (STQ);Measurements of Call EstablishmentPerformance in IP NetworksETSI ETSI TR 102 793 V1.2.1 (2010-11) 2Reference RTR/STQ-00180 Keywords IP, packet mode, protocol, QoS, quality ETSI 650 Route des Lucioles F
2、-06921 Sophia Antipolis Cedex - FRANCE Tel.: +33 4 92 94 42 00 Fax: +33 4 93 65 47 16 Siret N 348 623 562 00017 - NAF 742 C Association but non lucratif enregistre la Sous-Prfecture de Grasse (06) N 7803/88 Important notice Individual copies of the present document can be downloaded from: http:/www.
3、etsi.org The present document may be made available in more than one electronic version or in print. In any case of existing or perceived difference in contents between such versions, the reference version is the Portable Document Format (PDF). In case of dispute, the reference shall be the printing
4、 on ETSI printers of the PDF version kept on a specific network drive within ETSI Secretariat. Users of the present document should be aware that the document may be subject to revision or change of status. Information on the current status of this and other ETSI documents is available at http:/port
5、al.etsi.org/tb/status/status.asp If you find errors in the present document, please send your comment to one of the following services: http:/portal.etsi.org/chaircor/ETSI_support.asp Copyright Notification No part may be reproduced except as authorized by written permission. The copyright and the f
6、oregoing restriction extend to reproduction in all media. European Telecommunications Standards Institute 2010. All rights reserved. DECTTM, PLUGTESTSTM, UMTSTM, TIPHONTM, the TIPHON logo and the ETSI logo are Trade Marks of ETSI registered for the benefit of its Members. 3GPPTM is a Trade Mark of E
7、TSI registered for the benefit of its Members and of the 3GPP Organizational Partners. LTE is a Trade Mark of ETSI currently being registered for the benefit of its Members and of the 3GPP Organizational Partners. GSM and the GSM logo are Trade Marks registered and owned by the GSM Association. ETSI
8、 ETSI TR 102 793 V1.2.1 (2010-11) 3Contents Intellectual Property Rights 4g3Foreword . 4g3Introduction 4g31 Scope 5g32 References 5g32.1 Normative references . 5g32.2 Informative references 5g33 Abbreviations . 6g34 State of Art . 6g35 Need for a new indicator 7g36 Measurements of call establishment
9、 performance . 7g36.1 Call Setup Delay s . 8g36.1.1 Abstract Definition . 8g36.1.2 Abstract Equation . 8g36.1.3 Trigger Points . 9g36.2 Media Establishment Delay ms . 9g36.2.1 Abstract Definition . 9g36.2.2 Abstract Equation . 9g36.2.3 Trigger Points . 10g36.2.4 Informer section 10g36.2.5 Special Co
10、nsiderations 10g3Annex A: Void 12g3Annex B: Alternative NAT Handling in IMS 13g3History 15g3ETSI ETSI TR 102 793 V1.2.1 (2010-11) 4Intellectual Property Rights IPRs essential or potentially essential to the present document may have been declared to ETSI. The information pertaining to these essentia
11、l IPRs, if any, is publicly available for ETSI members and non-members, and can be found in ETSI SR 000 314: “Intellectual Property Rights (IPRs); Essential, or potentially Essential, IPRs notified to ETSI in respect of ETSI standards“, which is available from the ETSI Secretariat. Latest updates ar
12、e available on the ETSI Web server (http:/webapp.etsi.org/IPR/home.asp). Pursuant to the ETSI IPR Policy, no investigation, including IPR searches, has been carried out by ETSI. No guarantee can be given as to the existence of other IPRs not referenced in ETSI SR 000 314 (or the updates on the ETSI
13、Web server) which are, or may be, or may become, essential to the present document. Foreword This Technical Report (TR) has been produced by ETSI Technical Committee Speech and multimedia Transmission Quality (STQ). Introduction The existing definitions of Call Establishment Delay are related only t
14、o the context of telephony on traditional PSTN networks. In VoIP context the Call Establishment may differ for several reasons. Firstly, the tones played by terminal do not necessarily correspond to the state of the network. For example, the user terminal may play the ring back tone even though it h
15、as not received any signalling message from the network informing that the called party has been alerted. In addition it can happen that some network resources are not available immediately after pick-up on caller part because the media channel needs to be established separately. In these conditions
16、 it is necessary to define new terms characterizing the Call Establishment as well as the related measurement methodology. ETSI ETSI TR 102 793 V1.2.1 (2010-11) 51 Scope The present document provides a state of art concerning definitions of call establishment delay existing in standardization. As ca
17、ll establishment in PSTN context and in IP context differ, the present document defines a new term of call establishment delay which better reflects the course of call establishment in VoIP networks. The call establishment delay can be reflected by two parameters: Call Setup Delay and Media Establis
18、hment Delay. 2 References References are either specific (identified by date of publication and/or edition number or version number) or non-specific. For specific references, only the cited version applies. For non-specific references, the latest version of the reference document (including any amen
19、dments) applies. Referenced documents which are not found to be publicly available in the expected location might be found at http:/docbox.etsi.org/Reference. NOTE: While any hyperlinks included in this clause were valid at the time of publication ETSI cannot guarantee their long term validity. 2.1
20、Normative references The following referenced documents are necessary for the application of the present document. Not applicable. 2.2 Informative references The following referenced documents are not necessary for the application of the present document but they assist the user with regard to a par
21、ticular subject area. i.1 ETSI EG 202 765-2: “Speech Processing, Transmission and Quality Aspects (STQ); QoS and network performance metrics and measurement methods; Part 2: Transmission Quality Indicator combining Voice Quality Metrics“. i.2 ITU-T Recommendation E.800: “Definitions of terms related
22、 to quality of service“. i.3 ITU-T Recommendation E.600: “Terms and Definitions of Traffic Engineering“. i.4 ITU-T Recommendation H.323: “ Packet-based multimedia communications systems“. i.5 IETF RFC 3261: “SIP: Session Initiation Protocol“. i.6 IETF RFC 5245: “Interactive Connectivity Establishmen
23、t (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols“. i.7 IETF RFC 5389: “Session Traversal Utilities for NAT (STUN)“. i.8 ETSI TS 123 228: “Digital cellular telecommunications system (Phase 2+); Universal Mobile Telecommunications System (UMTS); LTE; IP Mul
24、timedia Subsystem (IMS); Stage 2 (3GPP TS 23.228 version 9.3.0 Release 9)“. i.9 ETSI TS 124 229: “Digital cellular telecommunications system (Phase 2+); Universal Mobile Telecommunications System (UMTS); LTE; Internet Protocol (IP) multimedia call control protocol based on Session Initiation Protoco
25、l (SIP) and Session Description Protocol (SDP); Stage 3 (3GPP TS 24.229 version 9.3.1 Release 9)“. i.10 TIA FS 1037C: “Telecommunications: Glossary of Telecommunication Terms“. ETSI ETSI TR 102 793 V1.2.1 (2010-11) 6i.11 IETF RFC 3389: “Real-time Transport Protocol (RTP) Payload for Comfort Noise (C
26、N)“. i.12 IETF RFC 3551: “RTP Profile for Audio and Video Conferences with Minimal Control“. 3 Abbreviations For the purposes of the present document, the following abbreviations apply: ETSI European Telecommunications Standards Institute IP Internet Protocol ISDN Integrated Services Digital Network
27、 ICE Interactive Connectivity EstablishementIMS IP Multimedia Subsystem ITU-T International Telecommunication Union - Telecommunication standardization sector MTSI Multimedia Telephony Service for IMS NAT Network Adress Translation PSTN Public Switched Telephone Network RTP Realtime Transport Protoc
28、ol SBC Session Boarder Controller SDP Session Description Protocol SIP Session Initiation Protocol STUN Session Traversal Utilities for NAT TIA Telecommunications whereas from network perspective different names are used: post-selection delay, call set-up time or call delay. 5 Need for a new indicat
29、or The aforesaid discussion reflects well the call establishment delay definitions in PSTN and thus can be directly employed for measurements of this indicator. In PSTN networks the invitation to dialling signifies that the necessary resources to pass a call are available. In addition the ring back
30、tone signal indicates that the call is established and that a circuit is reserved. The call can begin at any time which depends only on called party, as the network has prepared everything for the call to start. The situation in IP networks is different. Not only the signal of invitation to dialling
31、 does not signify that the resources are available, also the ring back tone does not indicate that the network has prepared everything for the call to start. In fact, due to longer transmission delay in IP networks the call establishment for VoIP may take longer than in PSTN. Some devices are theref
32、ore inclined to diminish the waiting by groundlessly introducing ring back tone signal before the network has actually informed that the called party has been alerted. Therefore the fact of hearing the ring back tone signal in handset does not necessarily indicate that the telephone rings at the cal
33、led party. In addition, no matter which signalling protocol is used for call establishment, in VoIP telephony a media channel needs to be established separately. This may take less or more time. In some situations the interlocutors are obliged to wait for several milliseconds after the called party
34、has answered the phone before they can actually hear themselves speaking, as the media channel is not yet ready. It seems natural to claim that the call has been established when the network has provided the means for both interlocutors to converse and hear one another. As this delay is usually perc
35、eived by users it should therefore be taken into consideration when defining call establishment indicators. Media establishment delay can be equal to zero if the media flow has been established before answering the phone. Notice that in situations where media flow establishment begins as soon as the
36、 called user is being alerted, this indicator may be user dependant. In fact, in such cases the more time it will take for the called party to answer the phone, the smaller this indicator may be. However this should not be seen as a drawback as the purpose of this indicator is mainly to evaluate use
37、r perception. Consequently, it seems important to define a new term of call establishment delay which will better reflect the course of call establishment in VoIP networks. The call establishment delay can be reflected by two parameters: Call Setup Delay and Media Establishment Delay. 6 Measurements
38、 of call establishment performance The measurements of call establishment performance in a passive way are done at one point while analysing the IP flows. The different phases of call establishment and associated signalling messages for SIP and H.323 have been presented on figure 1. ETSI ETSI TR 102
39、 793 V1.2.1 (2010-11) 8Figure 1: Call establishment phases in VoIP context 6.1 Call Setup Delay s 6.1.1 Abstract Definition Time between sending of complete address information and receipt of call set-up notification. 6.1.2 Abstract Equation ()s t- t=sDelay Setup Callhandseton button send pressesuse
40、r destablisheconnect NOTE: This parameter is not calculated unless the telephony call setup attempt is successful. Note that a call with “user busy“ is also considered as a successful call setup. ETSI ETSI TR 102 793 V1.2.1 (2010-11) 96.1.3 Trigger Points Table 1 Event from abstract equation Trigger
41、 point from users point of view Technical description/protocol part tuser presses send button on handset: Time of call attempt Start: Push send button (e.g. #) or end of timeout. (See note 3). Start (H.323): The forming of first CONNECTION REQUEST message SETUP. Start (SIP): The forming of first CON
42、NECTION REQUEST message INVITE. tconnection established: Time when connection is established (successful call attempt) Stop: Alerting tone is heard by the A-party coming from the B-party. Stop (H.323): The message informing of ring back tone ALERTING or that the B-party user is occupied RELEASE COMP
43、LETE with release code 17 is received from the B-party. Stop (SIP): The message informing of ring back tone 180 RINGING or that the B-party user is occupied 486 Busy Here or 600 Busy Everywhere is received from the B-party. NOTE 1: With automatic tools there is not a significant difference between c
44、onsidering the alerting or the connect message, as the answer machine should always answer immediately. NOTE 2: For the trigger points of the technical description/protocol part, see figure 1. NOTE 3: For technical reasons if the push send button is not pressed by user telecom systems wait a timeout
45、 before processing the dialled number. This timeout is excluded from the measurement as it can be cut short by the user. 6.2 Media Establishment Delay ms 6.2.1 Abstract Definition Time between the moment the B-party answers the phone and the moment when media flows are established in both directions
46、. 6.2.2 Abstract Equation ()ms 0; t- tmax =msdelay nt Establisme Mediahook-offhandset party -Bdestablishe flows mediaNOTE 1: This parameter is not calculated unless the telephony call setup attempt is successful and the call was established (e.g. no busy tone). NOTE 2: In some systems the media flow
47、 is established before the B-party handset is off-hook. In such situations the Media Establishment delay should be equal to zero (not negative). See annex B for additional information about such situations. ETSI ETSI TR 102 793 V1.2.1 (2010-11) 106.2.3 Trigger Points Table 2 Event from abstract equa
48、tion Trigger point from users point of view Technical description/protocol part tB-party handset off hook: Time of taking the handset off-hook at the B-party Start: B-party answers the phone and end of ring back tone at the A-party. Start (H.323): The receiving of CONNECTION message CONNECT. Start (
49、SIP): The receiving of CONNECTION message 200 OK. Tmedia flows established: Time when media flows are established Stop: A-party can hear B-party and inversely. Stop: The reception of first media packet. (See note 5). NOTE 1: With automatic tools there is not a significant difference between considering the alerting or the connect message, as the answer machine should always answer immediately. The user dependant time is therefore negligible. NOTE 2: This parameter cannot be evaluated at analog interface as the connec
