Escolar Documentos
Profissional Documentos
Cultura Documentos
Table of Contents
1 Introducing AudioCodes’ ITP .............................................................................7
1.1 Objective ...............................................................................................................7
1.2 Interoperability Test Laboratory Environment ...................................................7
1.2.1 AudioCodes’ Components ......................................................................................................7
1.2.2 Third Party Components .........................................................................................................7
1.2.3 Laboratory Topology...............................................................................................................8
1.2.4 Configuration ..........................................................................................................................9
1.3 Contact Information .............................................................................................9
1.4 Test Summary.......................................................................................................9
1.4.1 Summary of Results & Open Issues .......................................................................................9
1.4.2 AudioCodes’ Visual Intercept (VI) Records...........................................................................10
1.4.3 Avaya Bug Records ..............................................................................................................10
1.5 Recommendations & Conclusions....................................................................11
1.6 Conventions........................................................................................................11
2 Test Case List....................................................................................................13
2.1 Basic Calls ..........................................................................................................13
2.1.1 Inbound Basic to Third party phone 1 Calls ..........................................................................13
2.1.2 Inbound Basic to Third Party Phone 2 Calls .........................................................................14
2.1.3 Inbound Basic to PBX Phone................................................................................................15
2.1.4 Inbound Basic M2K to Third Party Calls ...............................................................................16
2.1.5 Basic Call – Caller ID............................................................................................................17
2.2 RTP Features – Codecs and DTMF....................................................................18
2.2.1 RTP Features – Codecs .......................................................................................................18
2.2.2 RTP Features – DTMF .........................................................................................................19
2.3 Supplementary Services ....................................................................................21
2.3.1 SIP Call Hold ........................................................................................................................21
2.3.2 Call Transfer .........................................................................................................................22
2.3.3 Call Waiting...........................................................................................................................24
2.4 Fax Calls .............................................................................................................25
2.4.1 Transparent Mode ................................................................................................................25
2.4.2 Fax T.38................................................................................................................................26
2.5 SIP Features........................................................................................................26
2.5.1 SIP Features - Registration and Authentication ....................................................................27
2.5.2 SIP Features - PRACK and Early Media...............................................................................28
2.5.3 Connection Mode Features...................................................................................................30
List of Tables
Table 1-1: AudioCodes' Components.......................................................................................................7
Table 1-2: Third Party Components .........................................................................................................7
Table 1-3: Test Summary .........................................................................................................................9
Table 1-4: Visual Intercept (VI) Records ................................................................................................10
Table 1-5: Bug Records of Avaya ..........................................................................................................10
Table 1-6: Conventions ..........................................................................................................................11
Table 2-1: Inbound Basic to Third party phone 1 Calls ..........................................................................13
Table 2-2: Inbound Basic to Third Party Phone 2 Calls .........................................................................14
Table 2-3: Outbound Basic Calls to PBX Phone ....................................................................................15
Table 2-4: Inbound Basic M2K to Third Party Calls ...............................................................................16
Table 2-5: Basic Call – Caller ID ............................................................................................................17
Table 2-6: Codecs ..................................................................................................................................18
Table 2-7: DTMF ....................................................................................................................................19
Table 2-8: Call Hold – re-INVITE (SIP) ..................................................................................................21
Table 2-9: Call Transfer..........................................................................................................................22
Table 2-10: Call Waiting .........................................................................................................................24
Table 2-11: Transparent Mode ...............................................................................................................25
Table 2-12: Fax T.38 ..............................................................................................................................26
Table 2-13: Registration and Authentication ..........................................................................................27
Table 2-14: SIP Features - PRACK and Early Media ............................................................................28
Table 2-15: Connection Modes ..............................................................................................................30
List of Figures
Figure 1-1: Layout of the Interoperability Test Environment with Avaya Equipment ...............................8
Figure B-1: Configuring AudioCodes Products in Avaya SIP Proxy. .....................................................39
Notice
This Interoperability Test Plan (ITP) presents the scenarios according to which
AudioCodes’ products are tested to determine the degree to which they’re interoperable
©
with the Avaya Communication Manager. Information contained in this document is
believed to be accurate and reliable at the time of printing. However, due to ongoing
product improvements and revisions, AudioCodes cannot guarantee accuracy of printed
material after the Date Published nor can it accept responsibility for errors or omissions.
Updates to this document and other documents can be viewed by registered Technical
Support customers at www.audiocodes.com under Support / Product Documentation.
© Copyright 2006 AudioCodes Ltd. All rights reserved.
This document is subject to change without notice.
Refer to the current release notes that may be included with your documentation or hardware
delivery.
Date Published: May-10-2006 Date Printed: May-16-2006
Tip: When viewing this manual on CD, Web site or on any other electronic
copy, all cross-references are hyperlinked. Click on the page or section
numbers (shown in blue) to reach the individual cross-referenced item
directly. To return to the point from where you accessed the cross-
reference, press Alt + ←.
Trademarks
AC logo, Ardito, AudioCoded, AudioCodes, AudioCodes logo, IPmedia, Mediant,
MediaPack, MP-MLQ, NetCoder, Stretto, TrunkPack, VoicePacketizer and VoIPerfect, are
trademarks or registered trademarks of AudioCodes Limited.
All other products or trademarks are property of their respective owners.
WEEE EU Directive
Pursuant to the WEEE EU Directive, electronic and electrical waste must not be disposed
of with unsorted waste. Please contact your local recycling authority for disposal of this
product.
Customer Support
Customer technical support and service are provided by AudioCodes’ Distributors,
Partners, and Resellers from whom the product was purchased. For Customer support for
products purchased directly from AudioCodes, contact support@audiocodes.com.
Privacy Information
Full details pertaining to the specific telephony companies’ products with which
AudioCodes’ Media Gateways have been proven to interoperate, and to what degree of
interoperability, are presented in the AudioCodes Interoperability List, LTRT-10000.
Note: The ITP is open, modular and flexible, depending on the requirements of
the specific interoperability project. Some test scenarios can be waived, for
example, when testing for basic-level interoperability. Conversely, testers
can invent and add additional test scenarios, depending on network
conditions/requirements and third party features/capabilities.
1.1 Objective
In this subsection, the primary objective of this interoperability test is described.
The MP-114 FXS MediaPack gateway and Mediant 2000 digital gateway Version 4.8
tested in SIP with Avaya Communication Manager (CM) and SIP proxy.
Figure 1-1: Layout of the Interoperability Test Environment with Avaya Equipment
Describe (in words) the layout of the test environment shown in Figure 1-1, including
configured IP addresses and phone numbers of the components.
1.2.4 Configuration
Refer to Appendix A – AudioCodes’ ini File and Appendix B - Configuration File for
Third Party Devices on page 33 and 39 respectively.
FAILED = The tested feature is supported by all parties’ devices and the
appropriate configuration was performed, but the test failed due to
(for example) proprietary implementation of the feature by the third
party.
N/T = Not Tested. The feature is supported by all parties’ devices
according to product specifications, but the scenario was not run due
to (for example) scope constraints, etc.
N/S = Not Supported. The feature presently isn’t supported by one of the
parties’ devices, but future support is possible.
1.6 Conventions
Complete the column ‘Refers to’ in Table 1-6. If your interoperability test requires it, add
more conventions (for example, Third Party Phone 2) and define what they refer to.
Convention Refers to
1 AC1 MP-114 User
2 M2K Mediant 2000
3 Third Party Phone 1 Avaya SIP IP Phone User
4 Third Party Phone 2 Avaya H.323 IP Phone User
5 CM analog phone user connected to analog
PBX Phone
card in the CM
6 Third Party Server Avaya SIP Proxy
Reader’s Notes
2 DTMF transport from AC1 supporting 1. AC1 declares its capability for an N/T
RFC 2833. AC2 doesn’t support RFC 2833 RTP telephone event.
RFC 2833 2. AC2 replies without an RTP
telephone event
3. DTMF passes in-band transport
3 DTMF transport from AC1 to PBX DTMF passes through an RFC Pass
phone (RFC 2833) 2833 RTP telephone event
Verify DTMF digit quality, including
fast digit dialing
This section tests the capability of the system to hold the call using these two methods.
AC3 and back to AC2 call waiting ring back tone and
AC1 has a CWI
3. AC1 presses flash, AC2 has
MOH if available
4. AC1 and AC3 converse
5. AC1 presses flash, AC3 has an
MOH if available
6. AC1 and AC2 converse
3 AC1 in call with AC2, PBX phone 1. AC1 and AC2 converse N/T
calls AC1, AC1 holds AC2 and 2. PBX phone calls AC1, PBX
answers PBX phone and back to phone receives a call waiting ring
AC2 back tone and AC1 has a CWI
3. AC1 presses flash, AC2 has an
MOH if available
4. AC1 and the PBX phone
converse
5. AC1 presses flash, PBX phone
has MOH if available
6. AC1 and AC2 converse
4 AC1 in call with AC2, third party 1. AC1 and AC2 converse N/T
phone calls AC1, AC1 holds AC2 2. Third party phone calls AC1,
and answers third party phone and third party phone receives a call
back to AC2 waiting ring back tone and AC1
has a CWI
3. AC1 presses flash, AC2 has an
MOH if available
4. AC1 and third party phone
converse
5. AC1 presses flash, third party
phone has an MOH if available
6. AC1 and AC2 converse
Reader’s Notes
;**************
;** Ini File **
;**************
[SYSTEM Params]
SyslogServerIP = 149.49.140.245
EnableSyslog = 1
[BSP Params]
PCMLawSelect = 3
RoutingTableHopsCountColumn = 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
0, 0
[ATM Params]
[Analog Params]
FXSLoopCharacteristicsFilename = 'MP11x10-1-fxs.dat'
CallProgressTonesFilename = 'usa_tones_11.dat'
[ControlProtocols Params]
[MGCP Params]
[MEGACO Params]
EP_Num_0 = 0
EP_Num_1 = 1
EP_Num_2 = 0
EP_Num_3 = 0
EP_Num_4 = 0
[SS7 Params]
IdlePCMPattern = 85
RFC2833PayloadType = 127
[WEB Params]
LogoWidth = '339'
[SIP Params]
LOCALSIPPORT = 5060
PLAYRBTONE2IP = 0
REGISTRATIONTIME = 3600
SIPT1RTX = 500
SIPT2RTX = 4000
ISPROXYUSED = 1
ISREGISTERNEEDED = 1
SIPDESTINATIONPORT = 5060
PLAYRBTONE2TEL = 2
DETFAXONANSWERTONE = 0
CHANNELSELECTMODE = 0
GWDEBUGLEVEL = 5
ENABLERPIHEADER = 0
ENABLEEARLYMEDIA = 1
ISUSERPHONE = 1
SIPSESSIONEXPIRES = 0
PROXYNAME = '149.49.140.210'
SIPGATEWAYNAME = '149.49.140.210'
PRACKMODE = 0
ALTROUTINGTEL2IPMODE = 0
SIPMAXRTX = 7
ASSERTEDIDMODE = 0
ISUSERPHONEINFROM = 0
ADDTON2RPI = 1
USESOURCENUMBERASDISPLAYNAME = 0
MINSE = 90
IPALERTTIMEOUT = 180
ISFAXUSED = 0
SIPTRANSPORTTYPE = 0
TCPLOCALSIPPORT = 5060
TLSLOCALSIPPORT = 6061
ENABLESIPS = 0
USERAGENTDISPLAYINFO = ''
SESSIONEXPIRESMETHOD = 0
USEDISPLAYNAMEASSOURCENUMBER = 0
USESIPTGRP = 0
SIPSUBJECT = ''
CODERNAME = g711Alaw64k,20,$$,$$,0
CODERNAME = g711Ulaw64k,20,$$,$$,0
CODERNAME = g729,20,$$,$$,0
CALLERDISPLAYINFO0 = 3000,0
TRUNKGROUP = 1-1,3006,0
PROXYIP = 149.49.140.210
AUTHENTICATION_0 = 3006,123456
TXDTMFOPTION = 4
[VXML Params]
[IPsec Params]
[PSTN-SDH Params]
;**************
;** Ini File **
;**************
;Profile: NONE
;Key features:;Max SW Ver: 5.0;Board Type: TrunkPack 1610;SS7 Links: MTP2=2 MTP3=2
M2UA=2 M3UA=1 ;Security: IPSEC MediaEncryption StrongEncryption
EncryptControlProtocol ;Coders: G723 G729 G728 NETCODER GSM-FR GSM-EFR AMR
EVRC-QCELP G727 ;Control Protocols: MGCP MEGACO H323 SIP
;E1Trunks=2;T1Trunks=2;Channel Type: RTP DspCh=60;PSTN Protocols: IUA=1 ;IP Media:
VXML ;Default features:;Coders: G711 G726;
;------------------------------
[SYSTEM Params]
SyslogServerIP = 149.49.140.245
EnableSyslog = 1
[BSP Params]
PCMLawSelect = 1
LocalOAMIPAddress = 149.49.140.243
RoutingTableHopsCountColumn = 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
0, 0
[ATM Params]
[Analog Params]
[ControlProtocols Params]
[MGCP Params]
[MEGACO Params]
EP_Num_0 = 0
EP_Num_1 = 1
EP_Num_2 = 0
EP_Num_3 = 0
EP_Num_4 = 0
[PSTN Params]
ProtocolType = 1
ClockMaster = 1
TerminationSide_0 = 1
TerminationSide_1 = 0
FramingMethod = c
LineCode = 2
[SS7 Params]
IdlePCMPattern = 85
[WEB Params]
LogoWidth = '339'
[SIP Params]
LOCALSIPPORT = 5060
PLAYRBTONE2IP = 0
SIPT1RTX = 500
SIPT2RTX = 4000
ISPROXYUSED = 1
ISREGISTERNEEDED = 1
SIPDESTINATIONPORT = 5060
PLAYRBTONE2TEL = 2
DETFAXONANSWERTONE = 0
CHANNELSELECTMODE = 1
GWDEBUGLEVEL = 5
ENABLERPIHEADER = 0
ENABLEEARLYMEDIA = 0
ISUSERPHONE = 1
SIPSESSIONEXPIRES = 0
PROXYNAME = '149.49.140.210'
SIPGATEWAYNAME = '149.49.140.210'
USERNAME = '3006'
PASSWORD = '123456'
PRACKMODE = 1
ALTROUTINGTEL2IPMODE = 0
SIPMAXRTX = 7
ASSERTEDIDMODE = 0
ISUSERPHONEINFROM = 0
ADDTON2RPI = 1
USESOURCENUMBERASDISPLAYNAME = 0
MINSE = 90
IPALERTTIMEOUT = 180
ISFAXUSED = 0
SIPTRANSPORTTYPE = 0
TCPLOCALSIPPORT = 5060
SIP183BEHAVIOUR = 0
PLAYBUSYTONE2ISDN = 0
TLSLOCALSIPPORT = 5061
ENABLESIPS = 0
USERAGENTDISPLAYINFO = ''
SESSIONEXPIRESMETHOD = 0
USEDISPLAYNAMEASSOURCENUMBER = 0
PLAYRBTONE2TRUNK_0 = -1
PLAYRBTONE2TRUNK_1 = 0
PLAYRBTONE2TRUNK_2 = -1
PLAYRBTONE2TRUNK_3 = -1
PLAYRBTONE2TRUNK_4 = -1
PLAYRBTONE2TRUNK_5 = -1
PLAYRBTONE2TRUNK_6 = -1
PLAYRBTONE2TRUNK_7 = -1
USESIPTGRP = 0
SIPSUBJECT = ''
CODERNAME = g711Alaw64k,20,0,$$,0
CODERNAME = g711Ulaw64k,20,0,$$,0
CODERNAME = g729,20,0,$$,0
TRUNKGROUP = 0-0/1-31,4000,0
TRUNKGROUP = 1-1/1-31,5000,0
PROXYIP = 149.49.140.210
[SCTP Params]
[VXML Params]
[IPsec Params]
[PSTN-SDH Param
www.audiocodes.com