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Voice over IP Consortium
SIP Trunking Interoperability Test Suite
Version 1.0
Last Updated: February 7, 2008
VoIP Consortium
University of New Hampshire
InterOperability Laboratory
121 Technology Drive, Suite 2
Durham, NH 03824
Phone: +1-603-862-0186
Fax: +1-603-862-4181
http://iol.unh.edu
2007 University of New Hampshire InterOperability Laboratory
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The University of New Hampshire
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Modification Report
Version Date Editors Comments
0.1 September 13, 2007 James Swan Initial draft created
0.2 September 13, 2007 Stephen Kolacz Modified Test Suite template to be morespecific to trunking
0.3 September 18, 2007 Stephen Kolacz Added Groups 1-5 and associated tests.
0.4 September 25, 2007 Stephen Kolacz Added Observable Results to all tests.
Imported Test Setup image.
1.0 January 31, 2008 James Swan Updated overall document.
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Acknowledgements
The University of New Hampshire would like to acknowledge the efforts of the following individuals in the
development of this test suite.
Stephen Kolacz University of New HampshireJames Swan University of New Hampshire
Timothy Winters University of New Hampshire
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Table of Contents
Modification Report
Acknowledgements
Table of Contents 4
Introduction
References
Terms: Definitions and Abbreviations 7Definitions 7Abbreviations 8
Test Setups 9Test Setup 1: General Test Setup 9
Group 1: PSTN Gateway Call Origination 10SIP_Trunking_IO_1.1: Call Setup and Termination via PSTN Gateway 11
SIP_Trunking_IO_1.2: Call Cancellation via PSTN Gateway 12SIP_Trunking_IO_1.3: Call Termination via SIP Server 13SIP_Trunking_IO_1.4: Call Rejection via SIP Server 14
Group 2: SIP Server Call Origination 15SIP_Trunking_IO_2.1: Call Setup and Termination via SIP Server 16SIP_Trunking_IO_2.2: Call Cancellation via SIP Server 17SIP_Trunking_IO_2.3: Call Termination via PSTN Gateway 18SIP_Trunking_IO_2.4: Call Rejection via PSTN Gateway 19
Group 3: SIP Call Trunk Failover Testing 20SIP_Trunking_IO_3.1: SIP Trunk Failover Test 21
Group 4: Codec Testing 23SIP_Trunking_IO_4.1: Call Setup and Termination Using G.711U Codec 24
SIP_Trunking_IO_4.2: Call Setup and Termination Using G.711A Codec 26SIP_Trunking_IO_4.3: Call Setup and Termination Using G.723 Codec 28SIP_Trunking_IO_4.4: Call Setup and Termination Using G.726 Codec 30SIP_Trunking_IO_4.5: Call Setup and Termination Using G.729 Codec 32
Group 5: SIP Trunk Busy Testing 34SIP_Trunking_IO_5.1: SIP Trunk Capacity 35
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Introduction
OverviewThe University of New Hampshires InterOperability Laboratory (UNH-IOL) is an institution designed to improve
the interoperability of standards based products by providing an environment where a product can be tested against
other implementations of a standard. This suite of tests has been developed to help implementers evaluate the
conformance of their SIP Endpoint Implementations. The success metrics of these tests are based on the standardsreferenced in this document. Successful completion of all tests contained in this suite does not guarantee that the
tested device will operate with other devices. However, these tests provide=de a reasonable level of confidence that
the Device Under Test will function well in most multi-vendor environments.
Organization of TestsEach test contains an identification section that describes the test and provides cross-reference information. The
discussion section covers background information and specifies why the test is to be performed. Tests are grouped
in order to reduce setup time in the lab environment. Each test contains the following information:
Test Number
The Test Number associated with each test follows a simple grouping structure. Listed first is the Test Group
Number followed by the tests number within the group. This allows for the addition of future tests to the
appropriate groups of the test suite without requiring the renumbering of subsequent tests.
PurposeThe purpose is a brief statement outlining what the test attempts to achieve. This also includes background
information on why one needs to perform such a test to show that the device complies with the standard.
References
The references section lists standards and other documentation that might be helpful in understanding and evaluating
the test and results.
Resource RequirementsThe requirements section specifies the hardware and test equipment that will be needed to perform the test. The
items contained in this section are special test devices or other facilities, which may not be available on all devices.
Last ModificationThis specifies the date of the last modification to this test.
Test LayoutThis setup section describes the configuration of the test environment. Small changes in the configuration should be
included in the test procedure.
DiscussionThe discussion section is optional. It is a general discussion of the test and relevant section of the specification,
including any assumptions made in the design or implementation of the test as well as known limitations.
ProcedureThe procedure section of the test description contains the step-by-step instruction for carrying out the test It
provides a cookbook approach to testing, and may be interspersed with observable results.
Test MetricsThe test metrics section lists the necessary parameters for success in a given test. When multiple values are possible
for a specific event, this section provides a short discussion on how to interpret them. The tests are structured so that
failure of one test metric will result in a failure for the entire test, or a request to refer to comments.
Possible ProblemsThis section contains a description of known issues with the test procedure, which may affect test results in certain
situations.
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References
[SIPCONN] SIP Forum. Technical Recommendation sf-adopted-twg-IP_PBX_SP_Interop-sibley-sipconnect: IP
PBX / Service Provider Interoperability, March 2006.
[RFC3261] Internet Engineering Task Force (IETF). Request for Comments (RFC) 3261: Session InitiationProtocol (SIP), June 2002.
[RFC3264] Internet Engineering Task Force (IETF). Request for Comments (RFC) 3264: An Offer/Answer
Model with the Session Description Protocol (SDP), June 2002.
[RFC4733] Internet Engineering Task Force (IETF). Request for Comments (RFC) 4733: RTP Payload for
DTMF Digits, Telephony Tones and Telephony Signals, December 2006.
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Terms: Definitions and Abbreviations
Definitions
Callee The endpoint at which the call is received.
Caller The endpoint at which the call process is started.Firewall The firewall provides packet filtering and general security services at the Service
Provider and Enterprise network edges.
IP PBX (PBX) The IP PBX constitutes an Enterprises collection of network elements that provides
packetized voice call origination and termination services using SIP for signaling and
RTP for media traffic. The definition of an IP PBX for the purposes of this test suite
includes any hard wired (physically connected) phones as well as any IP Phones
under the IP PBX Systems control.
IP Phones IP Phones are devices that are capable of originating and terminating packetized voice
calls using the Enterprises IP PBX. For the purposes of this test suite, IP Phones are
considered part of the IP PBX System itself and therefore subject to the same overall
requirements.
IPv4 Network The IPv4 network constitutes a combination of the physical and logical elements (i.e.
circuits, routers, switches, etc.) required to route and/or switch IPv4 packets betweenthe Service Provider and Enterprise network edges.
Message Data sent between SIP elements as part of the protocol. SIP messages are either
requests or responses.
Request A SIP message sent from a client to a server, for the purpose of invoking a particular
operation.
Response A SIP message sent from a server to a client, for indicating the status of a request sent
from the client to the server.
Signaling Gateway (SGW) The Signaling Gateway performs translation of SIP signaling to SS7 signaling.
SIP Application Server
(SAS)
The SIP Application Server is a server or group of servers within the Service
Providers network that provides PSTN call origination / termination services to
Enterprises using SIP.
SIP Proxy Server (SPS) The SIP Proxy Server is a server or group of servers that provides SIP message
routing and TLS termination services at the Service Provider and Enterprise networkedges.
Trunking Gateway (TGW) The Trunking Gateway interfaces with PSTN switches and converts packetized voice
samples to TDM voice samples.
UA (User Agent) A logical entity that can act as both a user agent client and a user agent server.
URI (Universal Resource
Indicator)
Generally, a formatted string which provides information on how to access a
particular resource. A SIP URI is in the form: sip:[email protected]
and is the primary method of locating users on the Internet.
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Abbreviations
LAN Local Area Network
RTP Real Time Protocol
SDP Session Description Protocol
SIP Session Initiation Protocol
UDP User Datagram ProtocolURI Uniform Resource Identifier
VLAN Virtual LAN
WAN Wide Area Network
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Test Setups
Test Setup 1: General Test Setup
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Group 1: PSTN Gateway Call Origination
Scope:
This set of tests is aimed at verifying basic call origination from a PSTN Gateway to a SIP Server over a SIP trunk.
Overview:This section is aimed at testing the interoperability between a PSTN gateway and a SIP Server using a SIP Trunk
where a single call is originated at the PSTN gateway and terminated at a SIP server. These tests involve
establishing, canceling, and terminating a call originated at a PSTN gateway.
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SIP_Trunking_IO_1.1: Call Setup and Termination via PSTN Gateway
Purpose: This test verifies that a SIP Server can properly setup and tear down a call initiated and terminated by the
PSTN Gateway.
References:
[RFC3261] Section 13[RFC3261] Section 15
Resource Requirements:
SIP Server
PSTN Gateway
PSTN Simulator
Last Modification: 01/29/2008
Test Layout:
Test Setup 1
Procedure:
Part A: PSTN originates and terminates call to the SIP Server
1. Configure PSTN gateway to use DTMF method RFC 2833 and G.711U codec.
2. Phone A calls Phone X.
3. Observe traffic on all networks.
4. Phone X answers the call.
5. Observe traffic on all networks.
6. Phone A goes on-hook to terminate the call.
7. Observe traffic on all networks.
Observable Results:
Part AStep 3: PSTN gateway must transmit an INVITE request to the SIP Server. The SIP Server must
respond with 180 Ringing response to the PSTN Gateway. Caller and Called Parties phone numbers
must be displayed in the INVITE on the SIP Server trunk side.
Step 5: The SIP Server must transmit a 200 OK to the PSTN Gateway. The PSTN Gateway must
respond with an ACK request to the SIP Server. A 2-way RTP stream must be established.
Step 7: The PSTN gateway must transmit a BYE request to the SIP Server. The SIP Server must
respond with a 200 OK response. Multimedia traffic must not be flowing between the SIP Server and
the PSTN Gateway.
Possible Problems:
None
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SIP_Trunking_IO_1.2: Call Cancellation via PSTN Gateway
Purpose: This test verifies that the SIP Server can properly handle the cancellation of a call by the PSTN Gateway.
References:
[RFC3261] Section 9
Resource Requirements:
SIP Server
PSTN Gateway
PSTN Simulator
Last Modification: 01/29/2008
Test Layout:
Test Setup 1
Procedure:
Part A: PSTN originates and terminates call to SIP Server
1. Configure PSTN gateway to use DTMF method RFC 2833 and G.711U codec.
2. Phone A calls Phone X.
3. Observe traffic on all networks.
4. Place Phone A on-hook before Phone X answers.
5. Observe traffic on all networks.
Observable Results:
Part A
Step 3: PSTN gateway must transmit an INVITE request to the SIP Server. The SIP Server must
respond with 180 Ringing response to the PSTN Gateway. Caller and Called Parties phone numbers
must be displayed in the INVITE on the SIP Server trunk side.Step 5: PSTN gateway must transmit a CANCEL request to the SIP Server. The SIP Server must
respond to the CANCEL request with a 200 OK response. The SIP Server must respond to the
INVITE request with a 487 Request Terminated response. The PSTN Gateway must respond to the
487 Request Terminated response with an ACK request to the SIP Server. Multimedia traffic must not
be flowing between the SIP Server and the PSTN Gateway.
Possible Problems:
None
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SIP_Trunking_IO_1.3: Call Termination via SIP Server
Purpose: This verifies that calls originated by the PSTN Gateway are properly terminated when the SIP Server
initiates termination.
References:
[RFC3261] Section 15
Resource Requirements:
SIP Server
PSTN Gateway
PSTN phone
Last Modification: 01/29/2008
Test Layout:
Test Setup 1
Procedure:
Part A: PSTN originates and call is established, SIP Server terminates
1. Phone A calls Phone X.
2. Observe traffic on all networks.
3. Phone X answers call from Phone A.
4. Phone X terminates call with Phone A.
5. Observe traffic on all networks.
Observable Results:
Part A
Step 2: The PSTN gateway must transmit an INVITE request to the SIP Server. The SIP Server must
respond with 180 Ringing response to the PSTN Gateway. Caller and Called Parties phone numbers
must be displayed in the INVITE on the SIP Server trunk side.Step 5: The SIP Server transmits a BYE request to the PSTN Gateway. The PSTN gateway responds
with a 200 OK. Multimedia traffic must not be flowing between the SIP Server and the PSTN
Gateway.
Possible Problems:
None
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SIP_Trunking_IO_1.4: Call Rejection via SIP Server
Purpose: This test verifies that a PSTN Gateway can properly handle calls that are rejected by a SIP Server.
References:
[RFC3261] Section 13.3.1.3
Resource Requirements:
SIP Server
PSTN Gateway
PSTN phone
Last Modification: 01/29/2008
Test Layout:
Test Setup 1
Procedure:
Part A: PSTN originates call to SIP Server, SIP Server rejects1. Configure PSTN gateway to use DTMF method RFC 2833 and G.711U codec.
2. Phone A calls Phone X.
3. Observe traffic on all networks.
4. The SIP Server terminates call before it is answered.
5. Observe traffic on all networks.
Observable Results:
Part A
Step 3: The PSTN gateway must transmit an INVITE request to the SIP Server. The SIP Server must
respond with 180 Ringing response to the PSTN Gateway. Caller and Called Parties phone numbers
must be displayed in the INVITE on the SIP Server trunk side.
Step 5: The SIP Server must transmit a 4xx error response to the PSTN Gateway. The PSTN gatewayresponds with an ACK request. Multimedia traffic must not be flowing between the SIP Server and the
PSTN Gateway.
Possible Problems:
None
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Group 2: SIP Server Call Origination
Scope:
This set of tests is aimed at verifying basic call origination from a SIP Server to a PSTN Gateway over a SIP trunk.
Overview:This section is aimed at testing the interoperability between a PSTN gateway and a SIP Server using a SIP trunk
when calls are originated at a SIP Server and terminated at a PSTN Gateway. These tests involve establishing,
canceling, and terminating a call originated at a SIP Server.
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SIP_Trunking_IO_2.1: Call Setup and Termination via SIP Server
Purpose: This test verifies that a SIP Server can originate a call over a SIP Trunk to a PSTN Gateway.
References:
[RFC3261] Section 13
[RFC3261] Section 15
Resource Requirements:
SIP Server
PSTN Gateway
PSTN Simulator
Last Modification: 01/29/2008
Test Layout:
Test Setup 1
Procedure:
Part A: SIP Server originates call to PSTN Gateway, PSTN Gateway terminates
1. Configure PSTN gateway to use DTMF method RFC 2833 and G.711U codec.
2. Phone X calls Phone A.
3. Observe traffic on all networks.
4. Phone A answers the call.
5. Observe traffic on all networks.
6. Place Phone X on-hook to terminate the call.
7. Observe traffic on all networks.
Observable Results:
Part A
Step 3: The SIP Server must transmit an INVITE request to the PSTN gateway. The PSTN Gatewaymust respond with 183 Session Progress response to the SIP Server. Caller and Called Parties phone
numbers must be displayed in the INVITE on the PSTN Gateway trunk side.
Step 5: The PSTN gateway must transmit a 200 OK to the SIP Server. The SIP Server must respond
with an ACK request to the PSTN Gateway. A 2-way RTP stream must be established between the
SIP Server and the PSTN Gateway.
Step 7: The SIP Server must transmit a BYE request to the PSTN Gateway. The PSTN Gateway must
respond with a 200 OK to SIP Server. Multimedia traffic must not be flowing between the SIP Server
and the PSTN Gateway.
Possible Problems:
None
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SIP_Trunking_IO_2.2: Call Cancellation via SIP Server
Purpose: This test verifies that a SIP Server can cancel a call over a SIP Trunk.
References:
[RFC3261] Section 9
Resource Requirements:
SIP Server
PSTN Gateway
PSTN Simulator
Last Modification: 01/29/2008
Test Layout:
Test Setup 1
Procedure:
Part A: SIP Server originates call to the PSTN Gateway, SIP Server cancels1. Configure the PSTN gateway to use DTMF method RFC 2833 and G.711U codec.
2. Phone X calls Phone A, Phone A does not answer.
3. Observe traffic on all networks.
4. Place Phone X on-hook after 30 seconds to cancel the call.
5. Observe traffic on all networks.
Observable Results:
Part A
Step 3: The SIP Server must transmit an INVITE request to the PSTN Gateway. The PSTN Gateway
must respond with 180 Ringing response to the SIP Server. Caller and Called Parties phone numbers
must be displayed in the INVITE on the PSTN Gateway trunk side.
Step 5: The SIP Server must transmit a CANCEL request to the PSTN Gateway. The PSTN Gatewaymust respond with a 200 OK to the CANCEL request. The PSTN Gateway must respond to the
INVITE request with a 487 Request Terminated response. The SIP Server must respond to the 487
Request Terminated response with an ACK request. Multimedia traffic must not be flowing between
the SIP Server and the PSTN Gateway.
Possible Problems:
None
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SIP_Trunking_IO_2.3: Call Termination via PSTN Gateway
Purpose: This test verifies that a SIP Server can properly handle calls that are rejected or terminated by the PSTN
Gateway.
References:
[RFC3261] Section 15
Resource Requirements:
SIP Server
PSTN Gateway
PSTN Simulator
Last Modification: 01/29/2008
Test Layout:
Test Setup 1
Procedure:
Part A: The SIP Server originates call to PSTN, PSTN terminates
1. Configure PSTN gateway to use DTMF method RFC 2833 and G.711U codec.
2. Phone X calls Phone A.
3. Observe traffic on all networks.
4. Phone A answers the call.
5. Observe traffic on all networks.
6. Place Phone A on-hook to cancel the call.
7. Observer traffic on all networks.
Observable Results:
Part A
Step 3: The SIP Server must transmit an INVITE request to the PSTN Gateway. The PSTN Gatewaymust respond with a 180 Ringing response to the INVITE request. Caller and Called Parties phone
numbers must be displayed in the INVITE on the PSTN Gateway trunk side.
Step 5: The PSTN Gateway must transmit a 200 OK response to the SIP Server. The SIP Server must
transmit an ACK request to the PSTN Gateway. A 2-way RTP stream must be established between the
SIP Server and the PSTN Gateway. Audio input to Phone X must be heard at Phone A. Audio input
to Phone A must be heard at Phone X.
Step 7: The PSTN Gateway must transmit a BYE request to the SIP Server. The SIP Server must
transmit a 200 OK response to the BYE request. Multimedia traffic must not be flowing between the
SIP Server and the PSTN Gateway.
Possible Problems:
None
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SIP_Trunking_IO_2.4: Call Rejection via PSTN Gateway
Purpose: This test verifies that a SIP Server can properly handle calls that are rejected by a PSTN Gateway.
References:
[RFC3261] Section 13.3.1.3
Resource Requirements:
SIP Server
PSTN Gateway
PSTN Simulator
Last Modification: 01/29/2008
Test Layout:
Test Setup 1
Procedure:
Part A: The SIP Server originates call to PSTN Gateway, PSTN Gateway rejects1. Configure PSTN gateway to use DTMF method RFC 2833 and G.711U codec.
2. Phone X calls Phone A.
3. Observe traffic on all networks.
4. The PSTN Gateway rejects the call.
5. Observe traffic on all networks.
Observable Results:
Part A
Step 3: The SIP Server must transmit an INVITE request to the PSTN Gateway. The PSTN Gateway
must respond with 180 Ringing response to the SIP Server. Caller and Called Parties phone numbers
must be displayed in the INVITE on the PSTN Gateway trunk side.
Step 5: The PSTN Gateway must transmit a 4xx error response message to the SIP Server. The SIPServer must transmit an ACK request to the PSTN Gateway. Multimedia traffic must not be flowing
between the SIP Server and the PSTN Gateway.
Possible Problems:
None
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Group 3: SIP Call Trunk Failover Testing
Scope:
This group of tests is aimed at verifying SIP trunk failover functionality when a SIP trunk fails.
Overview:This group of tests is aimed at testing the interoperability of a PSTN gateway and a SIP Server that have established
two SIP Trunks and one trunk has failed.
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SIP_Trunking_IO_3.1: SIP Trunk Failover Test
Purpose: This test verifies that a PSTN gateway and SIP Server can fail-over to a second configured SIP trunk if
one in a pair of trunks has failed.
References:
[RFC3261] Section 13
Resource Requirements:
SIP Server
PSTN Gateway
PSTN Simulator
Last Modification: 01/30/2008
Test Layout:
Test Setup 1
Procedure:
Part A: SIP Server fail-over.
1. Configure PSTN gateway to use DTMF method RFC 2833 and G.711U codec.
2. Phone X calls Phone A using trunk #1.
3. Observe traffic on all networks.
4. Phone A answers the call.
5. Observe traffic on all networks.
6. Disconnect the Ethernet cable from the first SIP trunk.
7. Place Phone X and Phone A on-hook.
8. Wait for the session to time out on the SIP Server and the PSTN Gateway.
9. Phone X calls Phone A.
10. Observe traffic on all networks.
11. Phone A answers the call.
12. Observe traffic on all networks.13. Place Phone X on-hook.
14. Observe traffic on all networks.
Part B: PSTN Gateway fail-over.
1. Configure PSTN gateway to use DTMF method RFC 2833 and G.711U codec.
2. Phone A calls Phone X using trunk #1.
3. Observe traffic on all networks.
4. Phone X answers the call.
5. Observe traffic on all networks.
6. Disconnect the Ethernet cable from the first SIP trunk.
7. Place Phone A and Phone X on-hook.
8. Wait for the session to time out on the SIP Server and the PSTN Gateway.
9. Phone A calls Phone X.
10. Observe traffic on all networks.
11. Phone X answers the call.12. Observe traffic on all networks.
13. Place Phone A on-hook.
14. Observe traffic on all networks.
Observable Results:
Part A
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Step 3: The SIP Server must transmit an INVITE request to the PSTN Gateway routed over SIP trunk
#1. The PSTN Gateway must respond with 180 Ringing response to the SIP Server. Caller and Called
Parties phone numbers must be displayed in the INVITE on the PSTN Gateway trunk side.
Step 5: The PSTN Gateway must transmit a 200 OK to the SIP Server. The SIP Server must transmit
an ACK request to the PSTN Gateway. A 2-way RTP stream must be established.
Step 10: The SIP Server must transmit an INVITE request to the PSTN Gateway and be routed over
trunk #2. The PSTN Gateway must respond with 180 Ringing response to the SIP Server.
Step 12: The PSTN Gateway must transmit a 200 OK response to the SIP Server. The SIP Server
must respond with an ACK request. A 2-way RTP stream must be established.
Step 14: The SIP Server must transmit a BYE request to the PSTN Gateway. The PSTN Gateway
must respond with a 200 OK response to the SIP Server. Multimedia traffic must not be flowing
between the SIP Server and the PSTN Gateway.
Part B
Step 3: The PSTN gateway must transmit an INVITE request to the SIP Server and be routed over SIP
trunk #1. The SIP Server must respond with 180 Ringing response to the PSTN Gateway. Caller and
Called Parties phone numbers must be displayed in the INVITE on the SIP Server trunk side.
Step 5: The SIP Server must transmit a 200 OK to the PSTN Gateway. The PSTN gateway must
respond with an ACK request to the SIP Server. A 2-way RTP stream must be established.
Step 10: The PSTN gateway must transmit an INVITE request to the SIP Server and be routed over
SIP trunk #2. The SIP Server must respond with 180 Ringing response to the PSTN Gateway.
Step 12: The SIP Server must transmit a 200 OK response to the PSTN Gateway. The PSTN Gatewaymust respond with an ACK request to the SIP Server. A 2-way RTP stream must be established.
Step 14: The PSTN gateway must transmit a BYE request to the SIP Server. The SIP Server must
respond with a 200 OK to the PSTN Gateway. Multimedia traffic must not be flowing between the SIP
Server and the PSTN Gateway.
Possible Problems:
None
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Group 4: Codec Testing
Scope:
This group of tests is aimed at verifying that the audio codecs supported by a PSTN gateway and a SIP Server
interoperates properly.
Overview:This group of tests is aimed at verifying the interoperability of a PSTN gateway and a SIP Server when multiple
codecs supported by both parties are exercised.
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The University of New Hampshire
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SIP_Trunking_IO_4.1: Call Setup and Termination Using G.711U Codec
Purpose: This test verifies that a basic call can be made over a SIP trunk using the G.711 u-law codec.
References:
[RFC3261] Section 13
Resource Requirements:
SIP Server
PSTN Gateway
PSTN Simulator
Last Modification: 01/31/2008
Test Layout:
Test Setup 1
Procedure:
Part A: PSTN Gateway Calls SIP Server using G.711 u-law codec1. Configure G.711 u-law as the primary Codec, Packet Size = 20(ms) and DTMF method RFC 2833 in PSTN
gateway.
2. Phone A calls Phone X.
3. Observe traffic on all networks.
4. Phone X answers the call.
5. Observe traffic on all networks.
6. Phone A goes on-hook to terminate the call.
7. Observe traffic on all networks.
Part B: SIP Sever Calls PSTN Gateway using G.711 u-law codec
1. Configure G.711 u-law as the primary Codec, Packet Size = 20(ms) and DTMF method RFC 2833 in the
SIP Server.
2. Phone X calls Phone A.
3. Observe traffic on all networks.4. Phone A answers the call.
5. Observe traffic on all networks.
6. Phone X goes on-hook to terminate the call.
7. Observe traffic on all networks.
Observable Results:
Part A
Step 3: The PSTN gateway must transmit an INVITE request to the SIP Server. The SIP Server must
respond with 180 Ringing response to the PSTN Gateway. Caller and Called Parties phone numbers
must be displayed in the INVITE on the SIP Server trunk side.
Step 5: The SIP Server must transmit a 200 OK response to the PSTN Gateway. The PSTN Gateway
must respond with an ACK request to the SIP Server. A 2-way RTP stream must be established.
Audio input to Phone X must be heard at Phone A. Audio input to Phone A must be heard at Phone X.
Step 7: The PSTN gateway must transmit a BYE request to SIP Server. The SIP Server must respond
with a 200 OK response to the PSTN Gateway. Multimedia traffic must not be flowing between the
PSTN Gateway and the SIP Server.
Part B
Step 3: The SIP Server must transmit an INVITE request to the PSTN Gateway. The PSTN Gateway
must respond with 180 Ringing response to the SIP Server. Caller and Called Parties phone numbers
must be displayed in the INVITE on the PSTN Gateway trunk side.
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Step 5: The PSTN Gateway must transmit a 200 OK response to the SIP Server. The SIP Server must
respond with an ACK request to the PSTN Gateway. A 2-way RTP stream must be established.
Audio input to Phone X must be heard at Phone A. Audio input to Phone A must be heard at Phone X.
Step 7: The SIP Server must transmit a BYE request to the PSTN Gateway. The PSTN Gateway must
respond with a 200 OK response to the SIP Server. Multimedia traffic must not be flowing between the
PSTN Gateway and the SIP Server.
Possible Problems:
None
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The University of New Hampshire
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SIP_Trunking_IO_4.2: Call Setup and Termination Using G.711A Codec
Purpose: This test verifies that a basic call can be made over a SIP trunk using the G.711 a-law codec.
References:
[RFC3261] Section 13
Resource Requirements:
SIP Server
PSTN Gateway
PSTN Simulator
Last Modification: 01/31/2008
Test Layout:
Test Setup 1
Procedure:
Part A: PSTN Gateway Calls SIP Server using G.711 a-law codec1. Configure G.711 a-law as the primary Codec, Packet Size = 20(ms) and DTMF method RFC 2833 in PSTN
gateway.
2. Phone A calls Phone X.
3. Observe traffic on all networks.
4. Phone X answers the call.
5. Observe traffic on all networks.
6. Phone A goes on-hook to terminate the call.
7. Observe traffic on all networks.
Part B: SIP Sever Calls PSTN Gateway using G.711 a-law codec
1. Configure G.711 a-law as the primary Codec, Packet Size = 20(ms) and DTMF method RFC 2833 in the
SIP Server.
2. Phone X calls Phone A.
3. Observe traffic on all networks.4. Phone A answers the call.
5. Observe traffic on all networks.
6. Phone X goes on-hook to terminate the call.
7. Observe traffic on all networks.
Observable Results:
Part A
Step 3: The PSTN gateway must transmit an INVITE request to the SIP Server. The SIP Server must
respond with 180 Ringing response to the PSTN Gateway. Caller and Called Parties phone numbers
must be displayed in the INVITE on the SIP Server trunk side.
Step 5: The SIP Server must transmit a 200 OK response to the PSTN Gateway. The PSTN Gateway
must respond with an ACK request to the SIP Server. A 2-way RTP stream must be established.
Audio input to Phone X must be heard at Phone A. Audio input to Phone A must be heard at Phone X.
Step 7: The PSTN gateway must transmit a BYE request to SIP Server. The SIP Server must respond
with a 200 OK response to the PSTN Gateway. Multimedia traffic must not be flowing between the
PSTN Gateway and the SIP Server.
Part B
Step 3: The SIP Server must transmit an INVITE request to the PSTN Gateway. The PSTN Gateway
must respond with 180 Ringing response to the SIP Server. Caller and Called Parties phone numbers
must be displayed in the INVITE on the PSTN Gateway trunk side.
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Step 5: The PSTN Gateway must transmit a 200 OK response to the SIP Server. The SIP Server must
respond with an ACK request to the PSTN Gateway. A 2-way RTP stream must be established.
Audio input to Phone X must be heard at Phone A. Audio input to Phone A must be heard at Phone X.
Step 7: The SIP Server must transmit a BYE request to the PSTN Gateway. The PSTN Gateway must
respond with a 200 OK response to the SIP Server. Multimedia traffic must not be flowing between the
PSTN Gateway and the SIP Server.
Possible Problems:
None
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The University of New Hampshire
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SIP_Trunking_IO_4.3: Call Setup and Termination Using G.723 Codec
Purpose: This test verifies that a basic call can be made over a SIP trunk using the G.723 codec.
References:
[RFC3261] Section 13
Resource Requirements:
SIP Server
PSTN Gateway
PSTN Simulator
Last Modification: 01/31/2008
Test Layout:
Test Setup 1
Procedure:
Part A: PSTN Gateway Calls SIP Server using G.723 codec1. Configure G.723 as the primary Codec, Packet Size = 30(ms) and DTMF method RFC 2833 in PSTN
gateway.
2. Phone A calls Phone X.
3. Observe traffic on all networks.
4. Phone X answers the call.
5. Observe traffic on all networks.
6. Phone A goes on-hook to terminate the call.
7. Observe traffic on all networks.
Part B: SIP Sever Calls PSTN Gateway using G.723 codec
1. Configure G.723 as the primary Codec, Packet Size = 20(ms) and DTMF method RFC 2833 in the SIP
Server.
2. Phone X calls Phone A.
3. Observe traffic on all networks.4. Phone A answers the call.
5. Observe traffic on all networks.
6. Phone X goes on-hook to terminate the call.
7. Observe traffic on all networks.
Observable Results:
Part A
Step 3: The PSTN gateway must transmit an INVITE request to the SIP Server. The SIP Server must
respond with 180 Ringing response to the PSTN Gateway. Caller and Called Parties phone numbers
must be displayed in the INVITE on the SIP Server trunk side.
Step 5: The SIP Server must transmit a 200 OK response to the PSTN Gateway. The PSTN Gateway
must respond with an ACK request to the SIP Server. A 2-way RTP stream must be established.
Audio input to Phone X must be heard at Phone A. Audio input to Phone A must be heard at Phone X.
Step 7: The PSTN gateway must transmit a BYE request to SIP Server. The SIP Server must respond
with a 200 OK response to the PSTN Gateway. Multimedia traffic must not be flowing between the
PSTN Gateway and the SIP Server.
Part B
Step 3: The SIP Server must transmit an INVITE request to the PSTN Gateway. The PSTN Gateway
must respond with 180 Ringing response to the SIP Server. Caller and Called Parties phone numbers
must be displayed in the INVITE on the PSTN Gateway trunk side.
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Step 5: The PSTN Gateway must transmit a 200 OK response to the SIP Server. The SIP Server must
respond with an ACK request to the PSTN Gateway. A 2-way RTP stream must be established.
Audio input to Phone X must be heard at Phone A. Audio input to Phone A must be heard at Phone X.
Step 7: The SIP Server must transmit a BYE request to the PSTN Gateway. The PSTN Gateway must
respond with a 200 OK response to the SIP Server. Multimedia traffic must not be flowing between the
PSTN Gateway and the SIP Server.
Possible Problems:
None
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The University of New Hampshire
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SIP_Trunking_IO_4.4: Call Setup and Termination Using G.726 Codec
Purpose: This test verifies that a basic call can be made over a SIP trunk using the G.726 codec.
References:
[RFC3261] Section 13
Resource Requirements:
SIP Server
PSTN Gateway
PSTN Simulator
Last Modification: 01/31/2008
Test Layout:
Test Setup 1
Procedure:
Part A: PSTN Gateway Calls SIP Server using G.726 codec1. Configure G.726-32 as the primary Codec, Packet Size = 20(ms) and DTMF method RFC 2833 in PSTN
gateway.
2. Phone A calls Phone X.
3. Observe traffic on all networks.
4. Phone X answers the call.
5. Observe traffic on all networks.
6. Phone A goes on-hook to terminate the call.
7. Observe traffic on all networks.
Part B: SIP Sever Calls PSTN Gateway using G.726 codec
1. Configure G.726-32 as the primary Codec, Packet Size = 20(ms) and DTMF method RFC 2833 in the SIP
Server.
2. Phone X calls Phone A.
3. Observe traffic on all networks.4. Phone A answers the call.
5. Observe traffic on all networks.
6. Phone X goes on-hook to terminate the call.
7. Observe traffic on all networks.
Observable Results:
Part A
Step 3: The PSTN gateway must transmit an INVITE request to the SIP Server. The SIP Server must
respond with 180 Ringing response to the PSTN Gateway. Caller and Called Parties phone numbers
must be displayed in the INVITE on the SIP Server trunk side.
Step 5: The SIP Server must transmit a 200 OK response to the PSTN Gateway. The PSTN Gateway
must respond with an ACK request to the SIP Server. A 2-way RTP stream must be established.
Audio input to Phone X must be heard at Phone A. Audio input to Phone A must be heard at Phone X.
Step 7: The PSTN gateway must transmit a BYE request to SIP Server. The SIP Server must respond
with a 200 OK response to the PSTN Gateway. Multimedia traffic must not be flowing between the
PSTN Gateway and the SIP Server.
Part B
Step 3: The SIP Server must transmit an INVITE request to the PSTN Gateway. The PSTN Gateway
must respond with 180 Ringing response to the SIP Server. Caller and Called Parties phone numbers
must be displayed in the INVITE on the PSTN Gateway trunk side.
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Step 5: The PSTN Gateway must transmit a 200 OK response to the SIP Server. The SIP Server must
respond with an ACK request to the PSTN Gateway. A 2-way RTP stream must be established.
Audio input to Phone X must be heard at Phone A. Audio input to Phone A must be heard at Phone X.
Step 7: The SIP Server must transmit a BYE request to the PSTN Gateway. The PSTN Gateway must
respond with a 200 OK response to the SIP Server. Multimedia traffic must not be flowing between the
PSTN Gateway and the SIP Server.
Possible Problems:
None
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The University of New Hampshire
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SIP_Trunking_IO_4.5: Call Setup and Termination Using G.729 Codec
Purpose: This test verifies that a basic call can be made over a SIP trunk using the G.729 codec.
References:
[RFC3261] Section 13
Resource Requirements:
SIP Server
PSTN Gateway
PSTN Simulator
Last Modification: 01/31/2008
Test Layout:
Test Setup 1
Procedure:
Part A: PSTN Gateway Calls SIP Server using G.729 codec1. Configure G.729 as the primary Codec, Packet Size = 20(ms) and DTMF method RFC 2833 in PSTN
gateway.
2. Phone A calls Phone X.
3. Observe traffic on all networks.
4. Phone X answers the call.
5. Observe traffic on all networks.
6. Phone A goes on-hook to terminate the call.
7. Observe traffic on all networks.
Part B: SIP Sever Calls PSTN Gateway using G.729 codec
1. Configure G.729 as the primary Codec, Packet Size = 20(ms) and DTMF method RFC 2833 in the SIP
Server.
2. Phone X calls Phone A.
3. Observe traffic on all networks.4. Phone A answers the call.
5. Observe traffic on all networks.
6. Phone X goes on-hook to terminate the call.
7. Observe traffic on all networks.
Observable Results:
Part A
Step 3: The PSTN gateway must transmit an INVITE request to the SIP Server. The SIP Server must
respond with 180 Ringing response to the PSTN Gateway. Caller and Called Parties phone numbers
must be displayed in the INVITE on the SIP Server trunk side.
Step 5: The SIP Server must transmit a 200 OK response to the PSTN Gateway. The PSTN Gateway
must respond with an ACK request to the SIP Server. A 2-way RTP stream must be established.
Audio input to Phone X must be heard at Phone A. Audio input to Phone A must be heard at Phone X.
Step 7: The PSTN gateway must transmit a BYE request to SIP Server. The SIP Server must respond
with a 200 OK response to the PSTN Gateway. Multimedia traffic must not be flowing between the
PSTN Gateway and the SIP Server.
Part B
Step 3: The SIP Server must transmit an INVITE request to the PSTN Gateway. The PSTN Gateway
must respond with 180 Ringing response to the SIP Server. Caller and Called Parties phone numbers
must be displayed in the INVITE on the PSTN Gateway trunk side.
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Step 5: The PSTN Gateway must transmit a 200 OK response to the SIP Server. The SIP Server must
respond with an ACK request to the PSTN Gateway. A 2-way RTP stream must be established.
Audio input to Phone X must be heard at Phone A. Audio input to Phone A must be heard at Phone X.
Step 7: The SIP Server must transmit a BYE request to the PSTN Gateway. The PSTN Gateway must
respond with a 200 OK response to the SIP Server. Multimedia traffic must not be flowing between the
PSTN Gateway and the SIP Server.
Possible Problems:
None
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Group 5: SIP Trunk Busy Testing
Scope:
This group of tests is aimed at testing the ability of a PSTN gateway and a SIP Server to forward calls over a second
SIP Trunk when all ports on the first are in use.
Overview:This group of tests is aimed at testing the interoperability between a PSTN gateway and a SIP Server when two SIP
Trunks are established and one is completely filled. When a call destined for the SIP Server arrives at the PSTN
gateway in this scenario it should route the call over the second SIP trunk. Similarly, when a call destined for the
PSTN Gateway arrives at the SIP Server in this scenario it should route the call over the second SIP Trunk. This
group of tests relies on the testers ability to configure a maximum number of ports per SIP Trunk on both devices.
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SIP_Trunking_IO_5.1: SIP Trunk Capacity
Purpose: This test verifies that a SIP Server and a PSTN Gateway can handle capacity fail-over for two configured
SIP Trunks.
References:
[RFC3261] Section 13
Resource Requirements:
SIP Server
PSTN Gateway
PSTN Simulator
PSTN call Generator
SIP Call Generator
Last Modification: 01/31/2008
Test Layout:
Test Setup 1
Procedure:
Part A: Trunk Failover with PSTN call generator
1. The PSTN call generator calls the SIP call generator.
2. The SIP call generator answers the call from the PSTN call generator.
3. Repeat steps 1-2 until the capacity of SIP trunk #1 has been reached.
4. Observe traffic on all networks.
5. Phone A calls Phone X.
6. Phone X answers the call from Phone A.
7. Observe traffic on all networks.
Part B: Trunk Failover with SIP call generator
1. The SIP call generator calls the PSTN call generator.
2. The PSTN call generator answers the call from the SIP call generator.3. Repeat steps 1-2 until the capacity of SIP trunk #1 has been reached.
4. Observe traffic on all networks.
5. Phone X calls Phone A.
6. Phone A answers the call from Phone X.
7. Observe traffic on all networks.
Observable Results:
Part A
Step 4: All calls are routed and active on SIP trunk #1.
Step 7: Call is routed and active on SIP trunk #2
Part B
Step 4: All calls are routed and active on SIP trunk #1.
Step 7: Call is routed and active on SIP trunk #2
Possible Problems:
None