Service Provider SIP trunk Validation Detailed Test Plan

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Service Provider SIP trunk Validation

Detailed Test Plan

Modifications

TEST

Document Number

EDCS-827327

Based on Template

EDCS-206096 Rev 35

Create By

Cecily Lui

Revision

Name

User Id

Date

Comments

1 Tony Banuelos

tbanuelo

11/2/2009 Initial Draft

2 Tony Banuelos

Tbanuelo

3/17/2010 Modified Hold/Resume test cases to

include MoH and ToH scenarios, as

well as clarify what side must

invoke Hold

Printed September 11, 2002

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1 Introduction/Background ... 4

2 What Will Be Tested ... 5

2.1 Test Coverage ... 6

3 Test Setups ... 7

4 Test Approach ... 7

4.1 Hardware ... 7

4.2 Software ... 7

4.3 Test Equipment and Test Tools ... 7

5 Test Exit Criteria ... 8

6 Detailed Test Cases ... 9

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Table of Figures

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1 Introduction/Background

This document can be used for Cisco Unified Communications Manager (Cisco UCM) or

Cisco Unified Communications Manager-Session Manager Edition with Cisco Unified Border

Element (Cisco UBE). This test plan is meant to be a guide only and can be modified by the

service provider to meet their needs. All questions should be directed to your Cisco account

team.

If posting results to Cisco Interoperability Portal (www.cisco.com/go/interoperability), please

note that the test plan needs to be reviewed and approved by the Cisco SIP trunk team

before testing begins. Also, the service provider must provide the following items to post:

key SIP call traces (see list below), completed application note template (provided by Cisco),

and a test report within the appendix of the application note. If you do not need to post your

results to Cisco.com, then the service provider does not need to provide this information to

Cisco.



Basic call Customer Premises Equipment (CPE) to PSTN with DTMF in both directions,

keep call up for 90 secs (include signaling and rtp traffic in trace capture)



Basic call PSTN to CPE with DTMF in both directions, keep call up for 90 secs (include

signaling and rtp traffic in trace capture)



Call from PSTN to CPE, CPE side user places call on HOLD and RESUMES call

successfully (include signaling and rtp traffic in trace capture)



Call from PSTN to CPE, PSTN side user places call on HOLD and RESUMES call

successfully (include signaling and rtp traffic in trace capture)



Call from PSTN to CPE, CPE side user1 conferences CPE side user2 into a three-way

conference call (include signaling and rtp traffic in trace capture)



Call from PSTN to CPE, CPE side user1 attended-transfers call to CPE user2 (include

signaling and rtp traffic in trace capture)



Call from PSTN to CPE, CPE side user1 attended-transfers call to PSTN2 (include

signaling and rtp traffic in trace capture)



Call from PSTN to CPE, CPE side user1 early-transfers call to CPE user2 (include

signaling and rtp traffic in trace capture)



Call from PSTN to CPE, CPE side user1 early-transfers call to PSTN2 (include signaling

and rtp traffic in trace capture)

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

Call from PSTN to CPE, CPE forwards unconditional to PSTN2 (include signaling and rtp

traffic in trace capture)



Call from PSTN to CPE, CPE forwards on no answer to PSTN2 (include signaling and rtp

traffic in trace capture)



Call from PSTN to CPE, fax call using T.38 (include signaling, t38 and rtp traffic in trace

capture)



Call from CPE to PSTN, fax call using T.38 (include signaling, t38 and rtp traffic in trace

capture)



Call from PSTN to CPE, fax call using G.711 pass-through (include signaling and rtp

traffic in trace capture)



Call from CPE to PSTN, fax call using G.711 pass-through (include signaling and rtp

traffic in trace capture)

2 What Will Be Tested

This document will cover interoperability testing using SIP to test supplementary

services between Cisco Unified Communications Manager, or Cisco Unified

Communications Manager-Session Manager Edition, to Service Provider through Cisco

Unified Border Element.

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2.1 Test Coverage



Basic Calls



Basic Calls with Calling Name and Number as allowed or restricted



DTMF Relay



Call Conference (Intra-site, PSTN)



Call Transfer (Blind, Attended, Early Attended)



Hold and Resume



Voice Mail



T.38 FAX G3/SG3



Simultaneous Calls



Auto Attendant



International calls



G.711 FAX G3/SG3



Call Forwarding - Find Me (Unconditional, Busy, No Reply)



Codec negotiation



Dial Plans

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3 Test Setups

Figure 3.1 Test Setup

4 Test Approach

4.1 Hardware



Cisco 3845 Cisco unified border element



MCS 7835 –I3

4.2 Software



IOS (C3845-ADVENTERPRISEK9_IVS-M), Version 12.4(20)T4



CUCM 8.5

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

Wireshark Protocol Analyzer 1.4.3

5 Test Exit Criteria

No problems were found during testing. Because additional hardware was not available in the

lab at the time of testing, the CPE to CPE test cases were not executed. Skype does not

support Early Media. When early media is send from PSTN, Skype may or may not cut the

audio by sending a 200 OK with SDP. Also, Skype does not support transmission or

reception of Fax in any form.

Special consideration is required with the FROM header. Skype only accepts INVITE

requests with a FROM header that has a SkypeID. If caller id presentation is required, the

information must be included in the P-asserted-identity, and synchronized with information

provisioned on Skype portal.

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6 Detailed Test Cases

Note 1 CPE refers to the Cisco Unified Communications Manager with Cisco Unified

Border Element or Cisco Unified Communications Manager Session Manager Edition with

Cisco Unified Border Element and the respective end-points devices.

Note 2 PSTN refers to the SIP Service Provider network

Note 3 All CPE to CPE test cases are optional and are to be completed only if it is possible

to have two separate CPE environments communicating across the SP SIP core.

Test Case Details

Test

Case

No. Test Case Pass / Fail

Comments:

Note calling name/number and connected name/number display

1.0

CPE to PSTN: codec advertisement shall be g729 and g711ulaw, with g729 codec listed as the preferred codec

1.1 Ringback heard on Caller phone Pass Tc1.pcap

1.2 Two-way voice path on call answerl Pass Tc1.pcap

1.3

Incomplete call. When caller hangs-up before callee answers, callee

phone stops ringing Pass Tc1.3.pcap

1.4 Call duration: 1 hour Pass Tc1.4.pcap

1.5 DTMF relay (both directions) (RFC2833) Pass Tc1.5.pcap

1.6 Callee disconnect; Caller disconnect automatically Pass Tc1.pcap

1.7 Caller disconnect; Callee disconnect automatically Pass Tc1.5.pcap

Test Case Details

Test

Case

Comments:

2.0

PSTN to CPE: codec advertisement shall be g729 and g711ulaw, with g729 codec listed as the preferred codec

2.1 Ringback heard on Caller phone Pass Tc2.pcap

2.2 Two-way voice path Pass Tc2.pcap

2.3

2.4 Call duration: 1 hour Pass Tc2.4.pcap

2.5 DTMF relay (both directions) (RFC2833) Pass Tc.2.5.pcap

2.6 Callee disconnect; Caller disconnect automatically Pass Tc2.5.pcap

2.7 Caller disconnect; Callee disconnect automatically Pass Tc2.pcap

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Test Case Details

Test

Case

Comments:

3.0

CPE to CPE (outbound to the PSTN and back) : codec advertisement shall be g729 and g711ulaw, with g729 codec listed as the preferred codec

3.1 Ringback heard on Caller phone N/T

3.2 Two-way voice path N/T

3.3

phone stops ringing N/T

3.4 Call duration: 1 hour N/T

3.5 DTMF relay (both directions) (RFC2833) N/T

3.6 Callee disconnect; Caller disconnect automatically N/T 3.7 Caller disconnect; Callee disconnect automatically N/T

Test Case Details

Test

Case

Comments:

Note calling number restricted

4.0 Calling number restriction

4.1 CPE to PSTN: Set calling Number restricted Pass Tcp4.1.pcap

4.2 CPE to another CPE: Set calling Number restricted N/T

Test Case Details

Test

Case

No. Test Case Pass / Fail Comments:

5.0 Telephone Number Support

5.1

CPE to PSTN: CPE must translate phone extension to 10 DID calling

number Pass CUCM has 4 digit internal extensions

5.2

PSTN to CPE: CPE must translate 10 digits called number to phone

extension. Pass

CUCM has 4 digit internal extensions receives 10 digits from PSTN

5.3

CPE to CPE: CPE must translate phone extension to 10 DID calling

number N/T

5.4

CPE to CPE: CPE must translate 10 digits called number to phone

extension. N/T

5.5

PSTN to CPE: CPE must translate 4 digits called number to phone extension.

N/A

Test configuration was full NANP numbers received at CUBE and CUCM, translated to 4

digits at CUCM.

5.6

CPE to CPE: CPE must translate 4 digits called number to phone

extension. N/T

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Test Case Details

Test

Case

Comments:

Note calling name and connected name display

6.0 Calling Name Presentation

6.1 CPE to CPE: display name presentation allowed N/T

Test Case Details

Test

Case

Comments:

Note calling name/number and connected name/number update

7.0 PSTN: Call Conference

7.1

PSTN phone A to CPE phone A, CPE phone A conferences PSTN

phone B Pass Tc7.1.pcap

7.2

CPE phone A to PSTN phone A, CPE phone A conferences PSTN

7.3

CPE phone A to CPE phone B, CPE phone A conferences PSTN

Test Case Details

Test

Case

Comments:

Note calling name/number and connected name/number update

8.0 Intra-Site: Call Conference

8.1 Phone A to Phone B. Phone A conferences Phone C at 2nd CPE site N/T 8.2 Phone A to 2nd CPE Phone B. Phone A conferences PSTN N/T

8.3

Phone A to Phone A at 2nd CPE site. Phone A at 2nd CPE

conferences Phone B at Phone A site N/T

8.4 Phone A to PSTN. Phone A conferences 2nd CPE Phone B N/T 8.5 Phone A at 2nd CPE site to Phone A. Phone A conferences Phone B N/T

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Test Case Details

Test

Case

Comments:

9.0 Attended Call Transfer

9.1

PSTN phone A to CPE phone A, CPE phone A transfers to PSTN

phone B (does caller ID update on PSTN phone B?) Pass

TC9.1.pcap. Skype does not support UPDATE method and calling number/name does not get

refresh

9.2

CPE phone A to PSTN phone A, CPE phone A transfers to PSTN

Tc9.2.pcap. Skype does not support UPDATE method and calling number/name does not get

refresh

9.3

CPE Phone A to CPE Phone B. Phone A transfers to Phone A at 2nd

CPE site N/T

9.4 CPE Phone A to CPE Phone B. CPE Phone A transfers to PSTN Pass

Tc9.4.pcap. Skype does not support UPDATE method and calling number/name does not get

refresh

9.5

CPE Phone A to Phone A at 2nd CPE site. Phone A and 2nd CPE

transfers to Phone B at Phone A site N/T

9.6 CPE Phone A to PSTN. CPE Phone A transfers to CPE Phone B Pass

The caller ID changed on CPE Phone B after the transfer was completed.

Tc9.6.pcap

9.7

Phone A at 2nd CPE site to CPE Phone A. CPE Phone A transfers to

Phone B N/T

9.8 PSTN to CPE Phone A. CPE Phone A transfers to CPE Phone B Pass

Tc9.8.pcap

Test Case Details

Test Case

Comments:

10.0 Unattended Call Transfer

10.1

Tc10.1.pcap Skype does not support UPDATE method and calling number/name does not get

refresh

10.2

refresh

10.3

CPE site N/T

10.4 CPE Phone A to CPE Phone B. CPE Phone A transfers to PSTN Pass

refresh

10.5

10.6 CPE Phone A to PSTN. CPE Phone A transfers to CPE Phone B Pass

Tc10.6.pcap

10.7

Phone B N/T

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10.8 PSTN to CPE Phone A. CPE Phone A transfers to CPE Phone B Pass

Tc10.8.pcap

Test Case Details

Test

Case

Comments:

11.0 Blind Call Transfer (applies if SIP blind transfer is supported)

11.1

phone B (does caller ID update on PSTN phone B?) Pass Tc11.1.pcap

11.2

phone B (does caller ID update on PSTN phone B?) Pass Tc11.2.pcap

11.3

CPE site N/T

11.4 CPE Phone A to CPE Phone B. CPE Phone A transfers to PSTN Pass Tc11.4.pcap

11.5

11.6 CPE Phone A to PSTN. CPE Phone A transfers to CPE Phone B Pass Tc11.6.pcap

11.7

Phone B N/T

11.8 PSTN to CPE Phone A. CPE Phone A transfers to CPE Phone B Pass Tc11.8.pcap

Test Case Details

Test

Case

12.0 Call Hold and Resume

Comments: call on hold is always perform on the CPE side. Note as of today, multicast MoH is not

supported with CUBE.

12.1 CPE calls to PSTN (unicast MoH is enabled). Was MoH heard? Pass Tc12.1.pcap

12.2 CPE calls to CPE (unicast MoH is enabled). Was MoH heard? N/T

12.3 PSTN calls to CPE (unicast MoH is enabled). Was MoH heard? Pass Tc12.3.pcap

12.4 CPE calls to PSTN (unicast MoH is disabled). Was ToH heard? Pass Tc12.4.pcap

12.5 CPE calls to CPE (unicast MoH is disabled). Was ToH heard? N/T

12.6 PSTN calls to CPE (unicast MoH is disabled). Was ToH heard?? Pass Tc12.6.pcap

Test Case Details

Test

Case

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14 13.0 Voice Mail (e.g. using Unity or Unity Connection)

13.1 PSTN to CPE: leave voice mail Pass Tc13.1.pcap

13.2 PSTN to CPE: retrieve voice mail Pass Tc13.2.pcap

Test Case Details

Test

Case

14.0 PSTN Voice Mail (e.g. using mobile phone voicemail)

14.1 CPE to PSTN (mobile VM): leave voice mail Pass Tc14.1.pcap

14.2 CPE to PSTN (mobile VM): retrieve voice mail Pass Tc14.2.pcap

Test Case Details

Test

Case

Comments:

15.0 Find Me (Call Forward Unconditionally)

15.1 PSTN to CPE enabled call forward unconditionally feature Pass Tc15.1.pcap

15.2 CPE to CPE enabled call forward unconditionally feature N/T

15.3

PSTN to CPE phone A enabled call forward unconditionally feature to

PSTN Pass Tc15.3.pcap

15.4

CPE to CPE phone A enabled call forward unconditionally feature to

Test Case Details

Test

Case

Comments:

16.0 T.38 FAX G3: Priority codec - G.729

16.1 CPE FAX to PSTN FAX - G3-G3 N/S Skype does not support fax

16.2 CPE FAX to PSTN FAX - G3-SG3 N/S Skype does not support fax

16.3 CPE FAX from PSTN FAX - G3-G3 N/S Skype does not support fax

16.4 CPE FAX from PSTN FAX - SG3-G3 N/S Skype does not support fax

16.5 CPE FAX to CPE FAX - G3-G3 N/S Skype does not support fax

16.6 CPE FAX to CPE FAX - G3-SG3 N/S Skype does not support fax

Test Case Details

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Test Case

Comments:

17.0 T.38 FAX SG3 (G.729 is offered first)

17.1 CPE FAX to PSTN FAX - SG3-G3 N/S Skype does not support fax

17.2 CPE FAX to PSTN FAX - SG3-SG3 N/S Skype does not support fax

17.3 CPE FAX from PSTN FAX - G3-SG3 N/S Skype does not support fax

17.4 CPE FAX from PSTN FAX -SG3-SG3 N/S Skype does not support fax

17.5 CPE FAX to CPE FAX - SG3-G3 N/S Skype does not support fax

17.6 CPE FAX to CPE FAX - SG3-SG3 N/S Skype does not support fax

Test Case Details

Test

Case

18.0 Simultaneous Calls (Minimum 2)- Duration 90 secs

18.1 CPE to PSTN gateway Pass Tc18.1.pcap

18.2 PSTN gateway inbound to CPE Pass Tc18.2.pcap

18.3 CPE to CPE N/T

Test Case Details

Test

Case

19.0

Auto Attendant (e.g. using Unity or Unity Connection AA services)

19.1 PSTN to CPE: navigate AA menus Pass Tc19.1.pcap

19.2 PSTN to CPE: navigate AA menu to transfer to a user Pass Tc19.2.pcap

Test Case Details

Test

Case

Comments:

20.0 CPE to PSTN gateway international call

20.1 Ringback heard on Caller phone Pass Tc20.1.pcap

20.2 Two-way voice path on call answerl Pass Tc20.2.pcap

20.3

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20.4 DTMF relay (both directions) (RFC2833) Pass

Tc20.2.pcap

DTMF relay can only be verified on one direction.

20.5 Callee disconnect; Caller disconnect automatically Pass Tc20.2.pcap

20.6 Caller disconnect; Callee disconnect automatically Pass TC20.1.pcap

Note 4 Cisco UCM does not support FAX pass-through (mid-call codec change to G711,

upspeed) (Support is available on Cisco UCM 7.1(5) and above using SIP devices/gw’s

only). CUCM only supports Fax over G.711, where the initial call must begin as a G711

media stream. Specific configurations on Cisco gateways and CUBE are required. See

appendix A for details

Test Case Details

Test

Case

Comments:

21.0 G.711 FAX G3

21.1 G3 CPE to PSTN FAX - G3-G3 N/S Skype does not support fax

21.2 G3 CPE to PSTN FAX - G3-SG3 N/S Skype does not support fax

21.3 G3 CPE from PSTN FAX - G3-G3 N/S Skype does not support fax

21.4 G3 CPE from PSTN FAX - SG3-G3 N/S Skype does not support fax

21.5 G3 CPE to CPE FAX - G3-G3 N/S Skype does not support fax

21.6 G3 CPE to CPE FAX - G3-SG3 N/S Skype does not support fax

Test Case Details

Test

Case

Comments:

22.0 G.711 FAX SG3

22.1 FAX SG3 to PSTN FAX - SG3-G3 N/S Skype does not support fax

22.2 FAX SG3 to PSTN FAX - SG3-SG3 N/S Skype does not support fax

22.3 FAX SG3 from PSTN FAX - G3-SG3 N/S Skype does not support fax

22.4 FAX SG3 from PSTN FAX -SG3-SG3 N/S Skype does not support fax

22.5 FAX SG3 to CPE FAX - SG3-G3 N/S Skype does not support fax

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22.6 FAX SG3 to CPE FAX - SG3-SG3 N/S Skype does not support fax

Test Case Details

Test

Case

Comments:

23.0 Find Me (Call Forward On Busy)

23.1

PSTN to CPE phone A enabled Call Forward on Busy feature to

23.2 PSTN to CPE phone enabled Call Forward on Busy feature to PSTN Pass Tc23.2.pcap

23.3

CPE to CPE phone A enabled Call Forward on Busy feature to phone

B N/T

23.4 CPE to CPE phone A enabled Call Forward on Busy feature to PSTN Pass Tc23.4.pcap

Test Case Details

Test

Case

Comments:

24.0 Find Me (Call Forward No Reply)

24.1

PSTN to CPE phone A enabled Call Forward No Reply feature to

24.2

PSTN to CPE phone A enabled Call Forward No Reply feature to

24.3

CPE to CPE phone A enabled Call Forward No Reply feature to

phone B N/T

24.4

CPE to CPE phone A enabled Call Forward No Reply feature to

Test Case Details

Test

Case

25.0 Codec mid-call negotiation (without transcoder)

25.1

PSTN calls CPE phone A (G729), phone A transfers to gateway (g711u). PSTN and CPE gateway negotiate codec and call is

transferred. N/T

SKIP test case (Cisco UCM does not support mid-call codec renegotiation) Unless you are testing Cisco UCM 7.1(5) or above where mid-call codec

renegotiation is supported with SIP end-points only

25.2

CPE phone A calls PSTN (G711), PSTN transfers call to CPE phone

B (G729), calls set up between CPE phone A and CPE phone B N/T

SKIP test case (Cisco UCM does not support mid-call codec renegotiation) Unless you are testing Cisco UCM 7.1(5) or above where mid-call codec

renegotiation is supported with SIP end-points only

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Test Case Details

Test

Case

Comments:

27.0

PRACK with SDP (early-media cut-through with DTMF (RFC2833)

navigation before 200OK)) - call 800-864-8331 - United Airlines .

27.1

CPE phone A call 800 number, phone user navigates through AA to

reach correct menu option

.

N/S

The united airlines number is not early-media anymore. Skype does not support early-media

Service Provider SIP trunk Validation Detailed Test Plan