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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
http://iol.unh.edu/http://iol.unh.edu/


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The University of New Hampshire


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.

VoIP Consortium SIP Trunking Interoperability Test Suite

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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.


6. Phone A goes on-hook to terminate the call.


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


SIP Server

PSTN Gateway

PSTN Simulator


Test Layout:

Test Setup 1

Procedure:

Part A: PSTN originates and terminates call to SIP Server




4. Place Phone A on-hook before Phone X answers.


Observable Results:

Part A

Step 3: PSTN gateway must transmit an INVITE request to the SIP Server. The SIP Server must


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


SIP Server

PSTN Gateway

PSTN phone


Test Layout:

Test Setup 1

Procedure:

Part A: PSTN originates and call is established, SIP Server terminates



3. Phone X answers call from Phone A.

4. Phone X terminates call with Phone A.


Observable Results:

Part A

Step 2: The PSTN gateway must transmit an INVITE request to the SIP Server. The SIP Server must


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


SIP Server

PSTN Gateway

PSTN phone


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.



4. The SIP Server terminates call before it is answered.


Observable Results:

Part A




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:




SIP Server

PSTN Gateway

PSTN Simulator


Test Layout:

Test Setup 1

Procedure:

Part A: SIP Server originates call to PSTN Gateway, PSTN Gateway terminates


2. Phone X calls Phone A.


4. Phone A answers the call.


6. Place Phone X on-hook to terminate the call.


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


SIP Server

PSTN Gateway

PSTN Simulator


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.


4. Place Phone X on-hook after 30 seconds to cancel the call.


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:



SIP Server

PSTN Gateway

PSTN Simulator


Test Layout:

Test Setup 1

Procedure:

Part A: The SIP Server originates call to PSTN, PSTN terminates






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


SIP Server

PSTN Gateway

PSTN Simulator


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.



4. The PSTN Gateway rejects the call.


Observable Results:

Part A




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:



SIP Server

PSTN Gateway

PSTN Simulator


Test Layout:

Test Setup 1

Procedure:

Part A: SIP Server fail-over.


2. Phone X calls Phone A using trunk #1.




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.




12. Observe traffic on all networks.13. Place Phone X on-hook.


Part B: PSTN Gateway fail-over.


2. Phone A calls Phone X using trunk #1.




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.



11. Phone X answers the call.12. Observe traffic on all networks.

13. Place Phone A on-hook.


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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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:



SIP Server

PSTN Gateway

PSTN Simulator


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.







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.


3. Observe traffic on all networks.4. Phone A answers the call.


6. Phone X goes on-hook to terminate the call.


Observable Results:

Part A




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





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respond with an ACK request to the PSTN Gateway. A 2-way RTP stream must be established.



respond with a 200 OK response to the SIP Server. Multimedia traffic must not be flowing between the


Possible Problems:

None


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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:



SIP Server

PSTN Gateway

PSTN Simulator


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.







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.






Observable Results:

Part A










Part B





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Possible Problems:

None


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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:



SIP Server

PSTN Gateway

PSTN Simulator


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.







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.






Observable Results:

Part A










Part B





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Possible Problems:

None


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References:



SIP Server

PSTN Gateway

PSTN Simulator


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.








1. Configure G.726-32 as the primary Codec, Packet Size = 20(ms) and DTMF method RFC 2833 in the SIP

Server.






Observable Results:

Part A










Part B





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Possible Problems:

None


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References:



SIP Server

PSTN Gateway

PSTN Simulator


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.








1. Configure G.729 as the primary Codec, Packet Size = 20(ms) and DTMF method RFC 2833 in the SIP

Server.






Observable Results:

Part A










Part B





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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:



SIP Server

PSTN Gateway

PSTN Simulator

PSTN call Generator

SIP Call Generator


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.



6. Phone X answers the call from Phone A.


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.



6. Phone A answers the call from Phone X.


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

Documents

Sip Trunking Interoperability Ts v1 0