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SIP Terminal Certification Procedures

©Copyright by TADIRAN TELECOM® (TTL) L.P., 2016.

All rights reserved worldwide.

All trademarks contained herein are the property of their respective holders.

SIP Terminal Certification Procedures for Aeonix 2

Table of Contents

1 INTRODUCTION................................................................................................4

2 GENERAL INFORMATION..................................................................................52.1 REFERENCES..........................................................................................5

2.2 LOOK AND FEEL......................................................................................5

2.3 QUALITY AND PERFORMANCE..................................................................5

2.4 ROBUSTNESS.........................................................................................5

2.5 NETWORKING:........................................................................................5

2.6 SIP AND TELEPHONY..............................................................................5

2.7 APPLICATIONS........................................................................................6

3 GENERAL TESTS.............................................................................................8

4 INTEROP TESTING..........................................................................................114.1 AEONIX ON LOCAL LAN......................................................................114.1.1 Local SIP Terminal..................................................................................................12

4.1.2 Phone key /soft key / display..................................................................................19

4.1.3 TLS and SRTP........................................................................................................20

4.2 - AEONIX VIA INGATE SIPARATOR SBC................................................224.2.1 Remote SIP Terminal..............................................................................................23

4.2.2 Phone key /soft key / display..................................................................................30

4.2.3 TLS and SRTP........................................................................................................31


1 IntroductionToday there are thousands of SIP terminals. Since there is no set standard

for SIP only RFC guidelines, it is imperative that comprehensive testing and certification be done

to protect all involved parties. Additionally, many manufacturers have different software

versions between model.

It is going to be impossible to test and certify all SIP terminals. This program was developed with

the intent to allow dealers and customers to test and certify SIP terminal on Aeonix.

Tadiran SIP terminals provide additional function upon normal SIP operations (Extended SIP

Features – ESF). However, some customers have invested and are migrating existing SIP terminals

over to an Aeonix system.

This document provides basic SIP interop testing that a dealer or customer can perform to see how

interoperable different manufacture SIP terminals are with Aeonix.

Tadiran Partners can connect any SIP device or SIP application over SIP to Aeonix at their own risk. If

the non-certified SIP device or trunk is not working properly, it is the Partners’ obligation to

investigate with the 3rd’ party manufacturer or provider to troubleshoot, gather logs and/or resolve the

problem. If the problem is identified with the Aeonix or Aeonix applications, the partner should open a

ticket with all relevant information pointing out where is the problem. Finally, it is up to Tadiran to

decide if, when and how to address the issue.

To make things clear, Tadiran is not under any obligation to resolve integration issues with non-

certified equipment/integrations.


2 General Information

For informational purposes, please fill out the following for your impressions of the SIP terminal (User

Test Terminal = UUT).

2.1 References

2.2 Look and FeelScreenKeys and LEDsHandsetHeadsetColorTonesLocal programming on phone WEB programming Supported Languages English, French, German, Chinese (traditional and simplified),

Italian, Polish, Portuguese, Russian, Spanish, Turkish, Hebrew

2.3 Quality and PerformanceAudio & Video quality Phone boot up speed - fast / average / slow ______ seconds

2.4 RobustnessHang up and unexpected restarts - ______________

2.5 Networking:PC port/switch Y / NPoE support Y / NGigE network Y / N 802.1x Y / N802.1p/q Y / NDIFFSERV Y / N

2.6 SIP and TelephonyNumber of SIP accounts - Survivability (2nd Registrar, 2nd Proxy, automatic back to Primary) - Supported codecs list -


2.7 Applications

1. HTTPS support - Y / N

2. SRTP support – Y / N

3. TLS support – Y / N

4. Can import ringtones (up to 100K size) – Y / N

5. Supports Alert_Info – Y / N

6. Able to handle and to keep good voice quality on jitter and delay conditions. Y / N Conditions:

Random packet loss of 30%Normal distributed latency of average 750 msec

7. SIP header for Auto-Answer, Call-Info: answer-after=0. Y / N

8. Possibility to send a parameter in auto answer that will tell the phone after what time to answer

9. Divert using 302 moved temporary support – Y /N

10. Support of 305 use proxy – Y / N

11. RFC 4235 (SIP BLF) support – Y / N

12. Remote phone book

support, Y / N


Support BLF State Led Status Colorterminatedconfirmedearlyproceeding

13. RFC Support – line through or circle what RFC are not supported by the device being tested.

RFC 2976 SIP INFO Method RFC 3261 SIP: Session Initiation Protocol RFC 3262 Reliability of Provisional Responses in SIP (PRACK) RFC 3264 An Offer/Answer Model with the Session Description Protocol (SDP)RFC 3265 SIP-Specific Event Notification (SUBSCRIBE and NOTIFY) RFC 3311 SIP UPDATE Method RFC 3312 Integration of Resource Management and SIP RFC 3313 Private SIP Extensions for Media Authorization RFC 3323 Privacy Mechanism for SIP RFC 3324 Requirements for Network Asserted Identity RFC 3325 SIP Asserted Identity RFC 3428 SIP Extension for Instant Messaging (MESSAGE) RFC 3455 3GPP SIP P-Header Extensions RFC 3489 STUN - Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs)RFC 3515 SIP REFER Method RFC 3680 SIP Registrations Event RFC 3711 Secure Real-time Transport Protocol (SRTP)RFC 3891 SIP "Replaces" Header RFC 3892 SIP "Referred-By" Mechanism RFC 3959 Early Session Disposition Type for SIP RFC 3960 Early Media and Ringing Tone Generation in SIP RFC 4028 Session Timers in SIP RFC 4235 INVITE-Initiated Dialog Event Package for SIP RFC 5009 P-Early-Media Header RFC 5079 Rejecting Anonymous Requests in SIP


3 General Tests

Test name Test Description Test notes Actual Results1.1 DNS resolving

supportConfigure DNS host with several Call Agents` ip addresses on the DNS server. Restart the UUT and open the sniffer capturing.

Scenario 1:

1. On UUT account configuration, SIP server is defined with host name on DNS and not with ip address.

2. DNS server sends response with the list of several different ip addresses for sip server hostname.

3. 1st host is not available on the network (for example, disconnected); the 2d and the 3d are available.

Scenario 2:1. On UUT account configuration, SIP server is

defined with host name on DNS and not with ip address.

2. DNS server sends response with the list of 2 different ip addresses for sip server hostname

3. Both of 2 DNS hosts are available on the network, SIP service is UP, UUT registers on the 1st host on DNS response list.

4. We disconnect the 1st host from the network, the 2d host remains available.

Scenario 3:

1. On UUT account configuration, SIP server is defined with host name on DNS and not with ip address

2. DNS server sends response with the list of different ip addresses for sip server hostname

3. The 1st host is available, but SIP service is down.

Scenario 4:1. On UUT account configuration, SIP server is

defined with host name on DNS and not with ip address

2. Configure 2 or more ip addresses on DHCP option 6 (DNS Servers); or configure manually primary and secondary DNS on UUT's network configuration

3. Primary DNS server is not available, secondary DNS server is available

1. UUT keeps sending ARP to the 1st ip address only in list, even it does not respond

2. UUT keeps sending ARP to the 1st ip address only in list, even it does not respond

3. UUT sends ARP to the 1st ip address only in list, and receives ARP response from it.

4. UUT sends ARP to the primary DNS, if not replied, it sends ARP to the secondary DNS


Hebrew/Chinese language support testsTest name Test Description Test notes Actual Results

1.1 Hebrew/Chinese MMI

Enter the UUT's menu, Basic settings > choose Hebrew/Chinese language

UUT menu syntax check.1. Status2. Features3. Settings4. Messages5. History6. Soft key7. Date 8. LED DSS button.9. menu include all the sub branch is display at

Hebrew/Chinese 10. Address book11. Add contact name in Hebrew/Chinese 12. Check the existing contact names in

Hebrew/Chinese 1.2 Hebrew/Chinese

Web GUI interface Enter the UUT's Web GUI interface, go to Phone menu, choose Hebrew/Chinese on Language list

UUT Web GUI interface menu syntax check:1. Status2. Account

a. Type name in Hebrew/Chinese on "Label"b. See whether the name is shown properly on

the UUT display3. Network4. Phone5. Contacts6. Upgrade7. Security8. Check notes on right side of the page

The Hebrew / Chinese names should be displayed in correct spelling

1.3 CID in Hebrew/Chinese , configured locally

1. Enter the UUT's Web GUI interface, go to Account and type name in Hebrew/Chinese on "Display name"

2. Call from the UUT to another UUT and check if the CID displayed properly in Hebrew/Chinese


1.4 CID in Hebrew/Chinese , configured on CA

Coral1. Enter name in Hebrew/Chinese on Aeonix. 2. Call from Hebrew/Chinese ext. to UUT

AEONIX1. On AEONIX web interface, go to user list,

choose a desired user and type name in Hebrew/Chinese on "user name" fiend

2. Call from phone assigned to Hebrew/Chinese user to UUT



Test name Test Description Test notes Actual ResultsAuto provisioning with Hebrew/Chinese values

Implement Auto provisioning for UUT.1. Change value of "Display name" on Account on

MAC.cfg file to Hebrew/Chinese name2. Do the auto provisioning now



4 Interop testing

4.1 AEONIX on Local LAN

Versions and phone type

1. Phone type: ___________

2. Phone version: _____________

3. Aeonix version: _____________

4. Tester name :________________ Date: ___________________

Test topologySIP phone connected on local network and registered on Aeonix softswitch

Test description Simple voice call, voice codes, Hook Flash, Call ID, DTMF, registration, DiffServ, transfer, 3-

way conference, redial

SIP Terminal Certification Procedures for Aeonix

T208M

Tadiran T4x

Tadiran T3xx

DHCP/DNS server

Aeonix

Access Point

UUT Device

11

4.1.1 Local SIP Terminal Test name Test Description Test notes Actual Results

1. 1. Simple call to various media types.

Make 4 calls: place two calls from one side:First call disconnects from the UUT side, second call disconnect from the remote side, then place calls from the remote side.

1.Make a call between the UUT to MGCP2. Make a call between the UUT to local SIP

phone

1. Two way voice2. Two way voice

1. 2.

2. 2 Simple call to various media types, using various codecs.

Configure the phone using various codecs on AEONIX side. First call from the UUT side, second call from the remote side. .(the scenario need to be tested without provisioning setting )

1. Reply previous tests with G.711a2. Reply previous tests with G.711u3. Reply previous tests with G.729A4. Reply previous tests with GSM (transparent

SDP defined)—(only if it support)

The UUT uses the codec, which was configured.

1. Two way voice2. Two way voice3. Two way voice4. Two way voice

1,2. 3. 4.

3. 3 Hold 1.Make a call from the UUT to other extension.2.After answer put it on hold by clicking on the

line the call initiated from. 3.Make a call to other extension from 2d line.4.Return to the first call and vice versa.

The 1st extension on the far side is on hold. The UUT sends dial tone on 2d line and dials another extension.

a. Phone - SIPb. Phone - MGCP

4. 4 Call waiting 1. Make a call from the phone to other extension.

2. Make another call on the same time from other extension to the tested phone.

Scenario 1: The remote phone is calling to UUT, the UUT is not answering. During this time 2d remote phone is calling the UUT. Result: The 2d remote phone receives call waiting tone. On the sniffer seen that the SP sends INVITE once from the 1st remote phone, and then from the 2d remote phone.

Scenario 2: The remote phone is calling to UUT, the UUT is answering. During this time 2d remote phone is calling the UUT. Result: The 2d remote phone receives call waiting tone. It`s CID shown on the UUT display. The second call is answered and switch between calls is successful.

Scenario 1 -

Scenario 2 -

5. 5 DND UUT feature:1. On the phone Screen Press DND SoftKey

when phone in Idle state

AEONIX feature:1. Open a AEONIX web page, enter with the

user profile. 2. On presence screen choose "DND"

Get DND enabled on the SIP phone screen and missed call when getting incoming call


Test name Test Description Test notes Actual Results6. 4.2Follow me

(forward all)UUT feature:

1. On UUT press on "Menu" soft key and go to "Features"

2. Choose "Forward" and enter the target number to forward to calls to.

3. Press "Save" softkey, then press "Back" softkey

4. To disable the forwarding choose option "Cancel Forwarding" on Forward menu, then press "Back" softkey

AEONIX feature:1.On AEONIX web admin page, login with the user profile.2.On user menu go to "My incoming routing"3.Choose "Forward Unconditional"

The call is forwarded to the defined number

7. 4.3Follow if busy UUT feature:

AEONIX feature:1. On AEONIX web admin page, login with the user profile.2.On user menu go to "My incoming routing"3.Choose "Forward Busy"


8. 4.4Follow no answer

UUT feature:

AEONIX feature:1.On AEONIX web admin page, login with the user profile.2.On user menu go to "My incoming routing"3.Choose " Forward no Answer to"

The unanswered call is forwarded to the defined number

9. 5 Pick up (direct ) 1.Place a call between two remote phones.Don’t answer the call.2.Dial from the UUT peak up feature code +destination extension (77 + called dial number).

The UUT picks up the call successfully

10. Auto answer Set the UUT as auto answer by dialing from the UUT 138 11 , and then make a call to the UUT from other phone .

Need that the UUT will answer the call automatically and will have 2 way audio After the caller will disconnect the call the UUT need to be in idle state.

11. Zone pageOn idle

Add the UUT to a zone page group Dial to the zone page need that the UUT will ring and immediately the phone will answer to the zone page and the announcement will be heard in the speaker .


Test name Test Description Test notes Actual Results12. Zone page

On busyAdd the UUT to a zone page group the UUT need to be busy in previous call

Dial to the zone page need that the UUT will get call waiting for the zone page call, after putting the previous call on holed the phone will get the zone page call.

13. Wakeup Defined a weak up via the UUT (set the wake up time 2 minutes forward from the system clock .

For example : set wakeup at 09:30 am need to dial from the UUT 151 11 0930 2151 is the weak up code

1. The setup action need to succeed and need to get confirmation tone and show “Wake UP at 09:30 AM”

2. After 2 minutes the UUT will get the wake up ring.

3. The phone display need to show “Wake Up at 09:30 AM”

4. After answer the call need to hear the weak up announcement.

14. Voice page 1. Dial 191+ number of the UUT The UUT should auto answer1way voice should be from paging phone to UUT

15. Aeonix MOH Make a call between the UUT and to other internal call, put the call on hold ,Ones from the UUT and second one from the other phone .

Need that at the both cases we will hear the Aeonix MOH

16. 22 Side tone Lift the phone receiver and check that you can hear the Side tone (it will come beside the dial tone.

During conversation with other alias check that you can hear the Side tone.

To check the side tone need to lift the receiver and blow into the handset a couple of times with your mouth, you should hear the sound in the handset receiver

17. Headset Remove the receiver from the sip phone.Connect the headset to the socket.In the phone Press on the headset button.Make an incoming call to the sip phone.Make an outgoing call from the sip phone

Incoming call the phone will ring by pressing on answer (soft key) you will answer the call need to get 2 way audio), pressing on cancel the call will disconnect

Note: if you don’t have headset the test can be done with 2 handset


Test name Test Description Test notes Actual Results18. Zip tone Set in the phone web:

Features/audio/call waiting tone- to an “enable”

Features/audio/ringer device for headset –to “use headset”

Connect headset to the phone Disconnect the handset In the phone Press on the headset

button.Call waiting is activate in the

Aeonix .

Test:1. Make a call to the phone (the phone is in idle state )

2. Make a call to the phone (the phone is busy with other call)

1. You need to hear the ring in the headset and not in the phone speaker.

2. You need to hear a call waiting tone during the conversion.


19. Audio Codes M800 SAS test

Enable the SAS on the Audio Codes M800 or MP-118 box

Stop the Aeonix server make sure that the UUT is succeed to register on the M800 or MP-118 and succeed to get a call and execute a call via the M800 or MP-118.

Restart the Aeonix make sure that the UUT is succeed to register on the Aeonix and succeed to get a call and execute a call via the M800 or MP-118, and to make and get internal call from the Aeonix.

20. Redundancy -1 Precondition:The UUT is registered to server A The second phone is registered to server B

HSB is not set in the system.

Make a call from the UUT to the second phone

Stop the Aeonix service in server A

The call may not to get disconnect


Test name Test Description Test notes Actual Results21. Redundancy -2 Set the UUT to be able to register to tow

servers

Check that the UUT is register on server A

Stop the Aeonix service on server A

Check that the UUT will register in 3 minutes to server B


Test name Test Description Test notes Actual Results22. 6.1Consultant

Transfer

B – transferee - the user agent to be transferred to the third party.

A – transferor - the user agent initiating the transfer

C – transfer target - the third party to which the transferee is to be connected.

Original call - the dialog between the transferor and the transferee which is setup before initiating the transfer.

Consultative call - the dialog between the transferor and the transfer target which is setup before completing the transfer.

Target call - the dialog between the transferee and the transfer target which is the final outcome of the transfer.

Scenario 1: UUT is AB – Tadiran SIP, C – Tadiran SIP phonesB – Tadiran MGCP, C – Tadiran MGCP phones

1. Place a call from the UUT to a remote phone (A call B).

2. Remote phone B answers and puts the conversation on HOLD

3. Make a call from B to another remote phone C, the phone is answering (B calls C)

4. Press the Transfer button in B and disconnect.

Scenario 2: UUT is BA – Tadiran SIP, C – Tadiran SIP phonesA – Tadiran MGCP, C – Tadiran MGCP phones

1. Place a call from a remote phone to UUT (A calls B).

2. UUT is answering the call3. Press "Tran" key and make a call from

UUT to another remote phone, the phone (C) is answering (B calls C)

4. Hang up the UUT (B)Scenario 3: UUT is CA – Tadiran SIP, B – Tadiran SIP phonesA – Tadiran MGCP, B – Tadiran MGCP phones

1. Place a call from a 1st remote phone to the 2d remote phone (A calls B).

2. The 2d remote phone is answering and puts the conversation on HOLD (B)

3. Make a call from 2d remote phone to UUT, the UUT is answering (B calls C)


Scenario 1 B should talk to CScenario 2 A should talk to CScenario 3 B should talk to C

Scenario 1 -

Scenario 2 -

Scenario 3 -


Test name Test Description Test notes Actual Results23. 6.2Transfer on ring








1. Place a call from the UUT to a remote phone (B call A).

2. Remote phone A answers and puts the conversation on HOLD

3. Make a call from A to another remote phone C, the phone is ringing (A calls C)

4. Press the Transfer button in A and disconnect.



2. UUT is answering the call3. Press "Tran" key and make a call from

UUT to another remote phone, the phone (C) is ringing (B calls C)


1. Place a call from a 1st remote phone to the 2d remote phone (B calls A).

2. The 2d remote phone is answering and puts the conversation on HOLD (A)

3. Make a call from 2d remote phone to UUT, the UUT is ringing (A calls C)



Scenario 1-

Scenario 2 -

Scenario 3 -

24. 7 3-way conference

1. Call the 1st 3-way participant extension2. When on conversation with the 1st

participant, press "conf" button and call the 2d way participant extension

3. The 3-way starts when the 2d way participant answers the call

3-way conference call, based at the UUT with clear voice for every participant.a. B – Tadiran SIP, C – Tadiran

SIP phonesb. B –MGCP, C – DKT phones


Test name Test Description Test notes Actual Results25. 8 Redial button 1. Use the redial button to place a call to a

number used before to place a call.2. Click on missed/received calls from "Calls"

column to place a call to a number from which a call was received before.

3. Dial the redial Aeonix code (by press “*”)

1.Find the number and place the call.

2.Find the number and place a call.3. Find the number and

place a call.

26. 9 Mute 1.Make a call. Verify two way clear voice.2.Press the “mute” button.

Verify only one-way voice (the SIP phone mute the voice from it side).

27. 10 Caller ID Enable Call ID on the AEONIX. Configure in station definition to send Call ID.

1.Make a call to this extension.2.Configure the Coral with numbers and names

in different length. Repeat the test.

1.Get correct Call ID2.Get correct Call ID.

28. 11 DTMF Open sniffer application. Make two voice calls: first emitted from SIP phone to SLS extension, second emitted from the SLS side.1. During the voice call, press keypad (DTMF)

on UUT2. Verify that you hear DTMF in the

destination.Verify with the Sniffer that the DTMF are transferred using in band RTP, with different payload type (Info, RFC 2833, Inband)

See the sniffer output, DTMF should be transferred using different payload type (Info, RFC 2833, Inband)

29. 12 Voice mail and MWI

1. Set CF ALL from the UUT to the SeaMail2. Dial from other extension to the UUT 3. After getting the sea mail prompt leave a

message in the mail box 4. Check that MWI led is on , and in some of

the phone need to appear an envelope sign 5. restart the phone after restart check that:

MWI led is on and also envelope sign6. from the UUT press on the message button ,

need to get the sea mail box 7. Type the mail box pass word listen the

message and delete it.8. After deleting and if you had only one

messageIn the mail box the MWI need to turn off, and the envelope sign clear. (If you have more than one message need to delete all the message )

1.CF ALL will succeeded.

2.Message will be saved successfully

3.MWI led is on The message envelop appears.

4.Getting to the mailbox 5.Getting the message 6.MWI is turn off and the

envelope sign clear.

30. 13 BLF 1. Configure BLF numbers on the UUT 2. Make a call between 2 BLF numbers

BLF LED’s should light ON when the numbers are in call

31. 14 BLF Configure the line2 button as BLF with number pageQ alias 7060 and then:

1.UUT is busy in a call2.press transfer and line2(pageQ 7060)3.press line2 (pageQ 7060) {to retrieve the call}4.repeat steps 2,3

You should be able to retrieve the call.


Test name Test Description Test notes Actual Results32. 15 Reject call 1. Call to UUT

2. When the UUT is ringing, press Reject softkey

The call should be rejected

33. 16 Presence (if supported by UUT)

Register the SIP phone to the Aeonix. Connect the SIP phone to the Aeonix using a Hub, and open a sniffer.

1.Presence is enabled Peer-to-Peer

Presence status should be seen

34. 17 License Tadiran phone

On Aeonix administration menu go to ”system", then to "License".

1. Register the UUT on Aeonix2. Disconnect the UUT from Aeonix

The value at “Aeonix Tadiran user” will increased by 1The value at Aeonix Tadiran user” will decreased by 1

35. License non Tadiran phone


1. Register the non Tadiran phone on Aeonix2. Disconnect the non Tadiran phone from

Aeonix

The value at “Aeonix non Tadiran user” will increased by 1The value at “Aeonix non Tadiran user” will decreased by 1

36. 18 Video (if supported by UUT)

Make video call with codecs H263 and H264 Check whether H.263 and H.264 codecs are supported and two-way video works on both codecs

37. 19 UUT with VPN client connected (relevant for softphones)

1. Connect to the corporate network with the VPN client

2. Run the UUT and register on corporate CA

UUT should be register and to be able to make and receive calls

38. 20 High availability test

Make a call, disconnect from LAN the server that the UUT is registered on, and wait till it registered to other server. Press HOLD & Retrieve – MOH played and you can retrieve the call.

Precondition in the Aeonix need to set Hsb = y Process time 5 second

Configure in the phone:SIP ServerEnable Outbound Proxy ServerOutbound Proxy ServerBackup Outbound Proxy Server

39. 21 Transfer to ULA 1. A dial to B ( ULA Owner) - ULA member answer ( x2002 )2. A press xfer (Not blind transfer) and dial to B ( ULA Owner) - ULA owner answer 3. B disconnect

Transfer is successfully establishedVoice on both ways.

4.1.2 Phone key /soft key / display Test name Test Description Test notes Actual

Results40. Display

Internal incoming call

Make Internal incoming call to the UUT : Check that the caller name is display in the UUT phone

41. DisplayInternal outgoing call

Make Internal outgoing call from the UUT Check that the destination name is display in the UUT phone


Test name Test Description Test notes ActualResults

42. History call log Make to and from the UUT calls

1. Incoming calls2. Misses Incoming calls3. Outgoing calls

Open the call log and check that :

1. All the Incoming calls that you made are appears in the call log

2. All the Misses Incoming calls that you made are appears in the call log

3. All the Outgoing calls that you made are appears in the call log

43. Combined Audio Enable combined audio in the in the UUT via the UUT web .Make a call to the UUT answer the call by lifting the handset Press on the speaker and put the handset on the cradle

Need that the call will not disconnect and you will be able to continue with the conversion with the speakerphone

4.1.3 TLS and SRTP

Test name Test Description Test notes Actual Results

44.TLS & SRTP Precondition:

Need to open a sniffer.

Aeonix need to be set with provisioning - Need to set in the Aeonix in user/ default general info/ required security level to “best

After setup the Check in the UUT phone web -> account/advance/rtp encryption – change to “optional”

Make a call between the two phones, Check in the sniffer that the RTP is running is SRTP between the two phones.


45.BLF performance – Test#1 Boss Group with 10 members – test UUT BLF and

delay in voice on incoming & outgoing calls

Define a ULA group with 10 members – one of the

members is a UUT

Define a BLF (at the UUT) to each member (9

BLFs) and owner (1 BLF) – Total of 10 BLF (You

must connect an expansion to the UUT in order to

define 10 BLFs)

A call arrives to the ULA owner alias – all 10 BLFs

should light.

Answer from any member (not from the UUT) – 9

BLFs should turn off, and only the BLF of the

member who answered the call should stay lit.

Additionally check for any delay in voice opening.

Repeat the test few times.

46.BLF performance – Test#2

Call group (ULA) with 50 members (use TGW-24

and TGW-96)

Define a call group of 50 members - use (TGW-24

and TGW-96) and the Phone UUT and two other

phones (Total 51 users) (Don't connect SLT phones

physically to the TGW-24 and TGW-96, just define

users and phones.

Attached expansions to the Phone UUT

Define all the buttons at the phone & expansions to

be BLF to all other users

Call to the call group alias – all 50 phones should

ring, make sure all the BLFs are light,

• Answer from the UUT and look for any delay in

voice opening, and all the BLF are turned off.

• Repeat the test few times


4.2 - AEONIX via Ingate SIParator SBCVersions

Versions and phone type

1. Phone type: _T46, T42, T49 and Touch_____________

Phone version: T46 28.80.19.4, T29 46.80.19.2, Touch iOS _____________

2. Aeonix version: __3.0.16___________

3. AudioCodes VE SBC version_7.2___________

4. Tester name :_Lindsay Kintner_______________ Date: _18 Aug 2016______

Test description Simple voice call, voice codes, Hook Flash, Call ID, DTMF, registration, DiffServ, transfer, 3-

way conference, redial

Test topologySIP phone connected on the Internet and registered on Sentinel Pro

SIP Terminal Certification Procedures for Aeonix

Router

UUT Device

AudioCodes VE SBC

T208M

Router

Tadiran T3xx

Aeonix

Tadiran T4x23

4.2.1 Remote SIP TerminalTest name Test Description Test notes Actual Results

1. Simple call to various media types.

Make 4 calls: place two calls from one side:First call disconnects from the UUT side, second call disconnect from the remote side, then place calls from the remote side.

1. Make a call between the UUT to MGCP2. Make a call between the UUT to local SIP

phone

1. Two way voice2. Two way voice

1. Pass2. Pass

2. Simple call to various media types, using various codecs.

Configure the phone using various codecs on Aeonix side. First call from the UUT side, second call from the remote side. .(the scenario need to to test without provisioning setting )

1. Reply previous tests with G.711a2. Reply previous tests with G.711u3. Reply previous tests with G.729A

The UUT uses the codec, which was configured.

1. Two way voice2. Two way voice3. Two way voice

1.Pass2. Pass3. Pass

3. Hold 1. Make a call from the UUT to other extension.

2. After answer put it on hold by clicking on the line the call initiated from.

3. Make a call to other extension from 2d line.

4. Return to the first call and vice versa.

The 1st extension on the far side is on hold. The UUT sends dial tone on 2d line and dials another extension.

c. Phone - SIPd. Phone - MGCP

1. Pass2. Pass

4. Call waiting 1. Make a call from the phone to other extension.

2. Make another call on the same time from other extension to the tested phone.

Scenario 1: The remote phone is calling to UUT, the UUT is not answering. During this time 2d remote phone is calling the UUT. Result: The 2d remote phone receives call waiting tone. On the sniffer seen that the SP sends INVITE once from the 1st remote phone, and then from the 2d remote phone.

Scenario 2: The remote phone is calling to UUT, the UUT is answering. During this time 2d remote phone is calling the UUT. Result: The 2d remote phone receives call waiting tone. It`s CID shown on the UUT display. The second call is answered and switch between calls is successful.

Scenario 1 -Pass

Scenario 2 - Pass

5. DND UUT feature:1. On the phone Screen Press DND

SoftKey when phone in Idle state

AEONIX feature:1. Open a Aeonix web page, enter with the

user profile. 2. On presence screen choose "DND"

Get DND enabled on the SIP phone screen and missed call when getting incoming call Pass


Test name Test Description Test notes Actual Results6. Follow me

(forward all)UUT feature:

1. On UUT press on "Menu" soft key and go to "Features"

2. Choose "Forward" and enter the target number to forward to calls to.

3. Press "Save" softkey, then press "Back" softkey

4. To disable the forwarding choose option "Cancel Forwarding" on Forward menu, then press "Back" softkey

AEONIX feature:1. On Aeonix web admin page, login

with the user profile.2. On user menu go to "My incoming

routing"3. Choose "Forward Unconditional"


Failed

7. Follow if busy UUT feature:

Aeonix feature:4. On Aeonix web admin page, login with the user profile.5.On user menu go to "My incoming routing"6.Choose "Forward Busy"


Failed – Aeonix does not act on packet because of X-Forwarded headers added by ABC

8. Follow no answer UUT feature:

Aeonix feature:4.On Aeonix web admin page, login with the user profile.5.On user menu go to "My incoming routing"6.Choose " Forward no Answer to"

The unanswered call is forwarded to the defined number

Failed – Aeonix does not act on packet because of X-Forwarded headers added by ABC

9. Pick up (direct ) 3.Place a call between two remote phones.Don’t answer the call.4.Dial from the UUT peak up feature code +destination extension (77 + called dial number).

The UUT picks up the call successfully

Passed

10. Auto answer Set the UUT as auto answer by dialing from the UUT 138 11 , and then make a call to the UUT from other phone .

Need that the UUT will answer the call automatically and will have 2 way audio After the caller will disconnect the call the UUT need to be in idle state.

Passed

11. Zone pageOn idle

Add the UUT to a zone page group Dial to the zone page need that the UUT will ring and immediately the phone will answer to the zone page and the announcement will be heard in the speaker.

Passed

12. Zone pageOn busy

Add the UUT to a zone page group the UUT need to be busy in previous call

Dial to the zone page need that the UUT will get call waiting for the zone page call, after putting the previous call on holed the phone will get the zone page call.

Passed


Test name Test Description Test notes Actual Results13. Wakeup Defined a wake up via the UUT

(set the wake up time 2 minutes forward from the system clock.

For example: set wake up at 09:30 am need to dial from the UUT 173 11 0930 2173 is the wake up code

1. The setup action need to succeed and need to get confirmation tone and show “Wake UP at 09:30 AM”

2. After 2 minutes the UUT will get the wake up ring.

3. The phone display need to show “Wake Up at 09:30 AM”

4. After answer the call need to hear the weak up announcement.

14. Voice page 1. Dial 191+ number of the UUT The UUT should auto answer1way voice should be from paging phone to UUT

Pass

15. Aeonix MOH Make a call between the UUT and to other internal call put the call on hold.Ones from the UUT and second one from the other phone .

Need that at the both cases we will hear the Aeonix MOH

Pass

16. Side tone Lift the phone receiver and check that you can hear the Side tone (it will come beside the dial tone.

During conversation with other alias check that you can hear the Side tone.

To check the side tone need to lift the receiver and blow into the handset a couple of times with your mouth, you should hear the sound in the handset receiver

Pass

17. Headset Eject the receiver from the sip phone.Connect the headset to the socket.In the phone Press on the headset button.Make an incoming call to the sip phone.Make an outgoing call from the sip phone

Incoming call the phone will ring by pressing on answer (soft key) you will answer the call need to get 2 way audio), pressing on cancel the call will disconnect


Pass


Test name Test Description Test notes Actual Results18. Zip tone Set in the phone web:

Features/audio/call waiting tone- to an “enable”

Features/audio/ringer device for headset –to “use headset”

Connect headset to the phone Disconnect the handset In the phone Press on the headset

button.Call waiting is activated in the

Aeonix.

Test:1. Make a call to the phone (the phone is in idle state)

2. Make a call to the phone (the phone is busy with other call)

1. You need to hear the ring in the headset and not in the phone speaker.

2. You need to hear a call waiting tone during the conversion.


Pass

19. Audio Codes M800 SAS test

Enable the SAS on the Audio Codes M800 or MP-118 box

Stop the Aeonix server make sure that the UUT is succeed to register on the M800 or MP-118 and succeed to get a call and execute a call via the M800 or MP-118.

Restart the Aeonix make sure that the UUT is succeed to register on the Aeonix and succeed to get a call and execute a call via the M800 or MP-118,and to make and get internal call from the Aeonix.

N/A

20. Redundancy -1 Precondition:The UUT is registered to server A The second phone is registered to server B

HSB is not set in the system.

Make a call from the UUT to the second phone

Stop the Aeonix service in server A

The call may not to get disconnect


Test name Test Description Test notes Actual Results21. Redundancy -2 Set the UUT to be able to register to tow servers

Check that the UUT is register on server A

Stop the Aeonix service on server A

Check that the UUT will register in 3 minutes to server B

22. Consultant Transfer








5. Place a call from the UUT to a remote phone (A call B).

6. Remote phone B answers and puts the conversation on HOLD

7. Make a call from B to another remote phone C, the phone is answering (B calls C)




2. UUT is answering the call3. Press "Tran" key and make a call from UUT

to another remote phone, the phone (C) is answering (B calls C)


1. Place a call from a 1st remote phone to the 2d remote phone (A calls B).

2. The 2d remote phone is answering and puts the conversation on HOLD (B)

3. Make a call from 2d remote phone to UUT, the UUT is answering (B calls C)



Scenario 1 - Pass

Scenario 2 - Pass

Scenario 3 – Pass

NOTE – T29G does not display CONF softkey when call on hold and placing second call


Test name Test Description Test notes Actual Results23. Transfer on ring








5. Place a call from the UUT to a remote phone (B call A).

6. Remote phone A answers and puts the conversation on HOLD

7. Make a call from A to another remote phone C, the phone is ringing (A calls C)




2. UUT is answering the call3. Press "Tran" key and make a call from UUT

to another remote phone, the phone (C) is ringing (B calls C)


1. Place a call from a 1st remote phone to the 2d remote phone (B calls A).

2. The 2d remote phone is answering and puts the conversation on HOLD (A)

3. Make a call from 2d remote phone to UUT, the UUT is ringing (A calls C)



Scenario 1- Pass

Scenario 2 - Pass

Scenario 3 - Pass

24. 3-way conference 1. Call the 1st 3-way participant extension2. When on conversation with the 1st

participant, press "conf" button and call the 2d way participant extension

3. The 3-way starts when the 2d way participant answers the call

3-way conference call, based at the UUT with clear voice for every participant.

1. B – Tadiran SIP, C – Tadiran SIP phones

2. B –MGCP, C – DKT phones


Test name Test Description Test notes Actual Results25. Redial button 1. Use the redial button to place a call to a

number used before to place a call.2. Click on missed/received calls from

"Calls" column to place a call to a number from which a call was received before.

3. Dial the redial Aeonix code (by press “*”)

1. Find the number and place the call.

2. Find the number and place a call.

3. Find the number and place a call.

Pass

26. Mute 1. Make a call. Verify two way clear voice.

2. Press the “mute” button.

Verify only one-way voice (the SIP phone mute the voice from it side).

Pass

27. Caller ID Enable Call ID on the AEONIX. Configure in station definition to send Call ID.

1. Make a call to this extension.2. Configure the Coral with numbers and

names in different length. Repeat the test.

1. Get correct Call ID2. Get correct Call ID.

28. DTMF Open sniffer application. Make two voice calls: first emitted from SIP phone to SLS extension, second emitted from the SLS side.

1. During the voice call, press keypad (DTMF) on UUT

2. Verify that you hear DTMF in the destination.

3. Verify with the Sniffer that the DTMF are transferred using in band RTP, with different payload type (Info, RFC 2833, Inband)

See the sniffer output, DTMF should be transferred using different payload type (Info, RFC 2833, Inband)

29. Voice mail and MWI

1. Set CF ALL from the UUT to the SeaMail2. Dial from other extension to the UUT 3. After getting the sea mail prompt leave a

message in the mail box 4. Check that MWI led is on , and in some of

the phone need to appear an envelope sign 5. restart the phone after restart check that:

MWI led is on and also envelope sign6. from the UUT press on the message

button , need to get the sea mail box 7. Type the mail box pass word listen the

message and delete it.8. After deleting and if you had only one

message9. In the mail box the MWI need to turn off,

and the envelope sign clear. 10. (If you have more than one message need

to delete all the message )

1. CF ALL will succeeded .

2. Message will be saved successfully

3. MWI led is on The message envelop appears.

4. Getting to the mailbox 5. Getting the message 6. MWI is turn off and the

envelope sign clear.

30. BLF 1. Configure BLF numbers on the UUT 2. Make a call between 2 BLF numbers

BLF LED’s should light ON when the numbers are in call

Pass


Test name Test Description Test notes Actual Results31. BLF Configure the line2 button as BLF with

number pageQ alias 7060 and then:

1.UUT is busy in a call2.press transfer and line2(pageQ 7060)3.press line2 (pageQ 7060) {to retrieve the call}4.repeat steps 2,3

You should be able to retrieve the call.

Pass

32. Reject call 1. Call to UUT2. When the UUT is ringing, press Reject

softkey

The call should be rejected Pass

33. Presence (if supported by UUT)

Register the SIP phone to the Aeonix. Connect the SIP phone to the Aeonix using a Hub, and open a sniffer.

Presence is enabled Peer-to-Peer

Presence status should be seen

NA

34. License Tadiran phone


1. Register the UUT on Aeonix Disconnect the UUT from Aeonix

The value at “Aeonix Tadiran user” will increased by 1The value at Aeonix Tadiran user” will decreased by 1

Pass

35. License non Tadiran phone

On Aeonix administration menu go to” system ", then to "License".

1. Register the non Tadiran phone on AEONIX2. Disconnect the non Tadiran phone from

Aeonix

The value at “Aeonix non Tadiran user” will increased by 1The value at “Aeonix non Tadiran user” will decreased by 1

Pass

36. Video (if supported by UUT)

Make video call with codecs H263 and H264 Check whether H.263 and H.264 codecs are supported and two-way video works on both codecs

Not relevant

37. UUT with VPN client connected (relevant for softphones)

1. Connect to the corporate network with the VPN client

2. Run the UUT and register on corporate CA

UUT should be register and to be able to make and receive calls

Not relevant

38. High availability test

Make a call, disconnect from LAN the server that the UUT is registered on, and wait till it registered to other server. Press HOLD & Retrieve – MOH played and you can retrieve the call.

Precondition in the Aeonix need to set HSB = y Process time 5 second

Configure in the phone:SIP ServerEnable Outbound Proxy ServerOutbound Proxy ServerBackup Outbound Proxy Server

39. Transfer to ULA 1. A dial to B ( ULA Owner) - ULA member answer ( x2002 )2. A press xfer (Not blind transfer) and dial to B ( ULA Owner) - ULA owner answer 3. B disconnect

Transfer is successfully establishedVoice on both ways.

4.2.2 Phone key /soft key / display Test name Test Description Test notes Actual

ResultsDisplayInternal incoming call

Make Internal incoming call to the UUT : Check that the caller name is display in the UUT PHONE


Test name Test Description Test notes ActualResults

DisplayInternal outgoing call

Make Internal outgoing call from the UUT Check that the destination name is display in the UUT PHONE

History call log Make to and from the UUT calls

4. Incoming calls5. Misses Incoming calls6. Outgoing calls

Open the call log and check that :

7. All the Incoming calls that you made are appears in the call log

8. All the Misses Incoming calls

that you made are appears in the call log

9. All the Outgoing calls that you made are appears in the call log

Combined Audio Enable combined audio in the in the UUT via the UUT web .Make a call to the UUT answer the call by lifting the handset Press on the speaker and put the handset on the cradle

Need that the call will not disconnect and you will be able to continue with the conversion with the speakerphone

4.2.3 TLS and SRTP

Test name Test Description Test notes Actual Results

TLS & SRTP Precondition :

Need to open a sniffer.

Aeonix need to be set with provisioning ,

Need to set in the Aeonix in user/ default general info/ required security level to “best

Set in the two sip Tadiran SIP phone :The provisioning setup parameters.

After setup the Check in the phone web -> account/advance/rtp encryption – change to “optional”

Make a call between the two phones, Check in the sniffer that the RTP is

running is SRTP between the two phones.


BLF performance – Test#1 Boss Group with 10 members – test UUT BLF

and delay in voice on incoming & outgoing

calls

Define a ULA group with 10 members – one

of the members is a UUT

Define a BLF (a t the UUT) to each member

(9 BLFs) and owner (1 BLF) – Total of 10

BLF (You must connect an expansion to the

UUT in order to define 10 BLFs)

A call arrives to the ULA owner alias – all 10

BLFs should light.

Answer from any member (not from the UUT)

– 9 BLFs should turn off, and only the BLF of

the member who answered the call should stay

lit.

Additionally check for any delay in voice

opening.

Repeat the test few times.


BLF performance – Test#2

Call group (ULA) with 50 members (use TGW-

24 and TGW-96)

Define a call group of 50 members - use

TGW-24 and TGW96 and the Phone UUT and

two other phones (Total 51 users) (Don't

connect SLT phones physically to the TGW-

24 or TGW-96, just define users and phones).

Attached expansions to the Phone UUT

Define all the buttons at the phone &

expansions to be BLF to all other users

Call to the call group alias – all 50 phones

should ring, make sure all the BLFs are light,

• Answer from the UUT and look for any

delay in voice opening, and all the BLF are

turned off.

• Repeat the test few times


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