Verify that desktop assistant is available to be downbloaded from web interface and at least runs on Windows

Go to https://<xivocc>/install/win64

Check that download starts 

On windows , right click on the installer .exe file  and check in properties that file is signed as Avencall

Application is marked as signed

Install it 

Application is installed, starts and can connect to xucmgt

Test conference from CC Agent

Given agent A1 has Snom phone

A1 logs in to CC Agent 

A2 calls queue

A1 answers

A1 calls A3 via Agent: type number into search box and press Attended transfer button (do not just press Enter)

A3 answers

A2 is on hold

A1 and A2 talk

A1 clicks conference

A1, A2 and A3 talk

Given agent A1 has Snom phone

A1 logs in to CC Agent

A2 calls queue

A1 answers (by button on phone)

A3 calls A1

A1 answers

A2 is on hold

A1 and A2 talk

A1 clicks conference

A1, A2 and A3 talk

Validate webRTC integration on the agent interface

Login with the user with webrtc line

Login success

Make an ougoing call

You have the ringback tone and once answered the callee

Put the call on hold

The call is on hold

Resume the call

The call is resumed

Hangup the call by the button on the interface

The call is terminated

Make a call to a queue where the agent is connected

The call is distributed to the agent.

Reject the call

The call is rejected.

Wait till call arrives once more to the agent, answer the call

The call is answered

Hangup the call by the button

The call is terminated

Login with the user with web rtc line on HTTP protocol (thus so without SSL)

Login is refused with an error

Login agent with a fixed phone

Login success

When Xuc connection is lost while using Assistant with SIP phone , user is redirected to login page. 

When Xuc connection is lost while using Assistant with WebRtc , user is shown error dialog and after clicking logout is user redirected to login page (also terminating any current call).

Log in user U1 into Web Assistant 

User U1 is logged-in to Web Assistant

Make call to another phone via Web Assistant and accept call on target line

User U1 has call in progress (via SIP phone) to another line 

Disable Xuc (but not Xivo) connection (for example shutdown Xuc or block Xuc port via network settings)

User U1 is redirected to login page, call in progress is not interrupted.

The error message on login page is "No response from server".

Terminate call in progress via phone and re-enable Xuc connection

Log in user U2 into Web Assistant 

User U2 is logged-in to Web Assistant

Make call to another phone via Web Assistant and accept call on target line

User U2 has call in progress (via WebRtc) to another line 

Disable Xuc (but not Xivo) connection (for example shutdown Xuc or block Xuc port via network settings)

User U2 sees error dialog with Logout button, call in progress is not interrupted
until U2 presses the Logout button

Click Logout in error dialog

User is redirected to login page, call in progress is interrupted

Test an outgoing WebRTC call

Login with the credentials of the user configured with the WebRTC line

Login OK, Registered in the navigator developer console.

Type "*55" (echo extension) in the search field and press Enter

You can talk and hear yourself back

Open console of your xivo and enter in asterisk CLI

asterisk -r

and activate rtp debug

rtp set debug on

You see RTP packets coming from and going to your UC IP like this

[Jun 30 15:34:58] Got  RTP packet from    10.32.5.1:49923 (type 111, seq 031220, ts 81358945, len 000046)
[Jun 30 15:34:58] Sent RTP packet to      10.32.4.1:38590 (via ICE) (type 111, seq 034570, ts 81358944, len 000046)
[Jun 30 15:34:58] Got  RTP packet from    10.32.5.1:49923 (type 111, seq 031221, ts 81359905, len 000044)
[Jun 30 15:34:58] Sent RTP packet to      10.32.4.1:38590 (via ICE) (type 111, seq 034571, ts 81359904, len 000044)

Press the hold button on the calls screen

• Call is put on hold
• In asterisk CLI, RTP flow is stopped (you don't see any RTP packet going to/coming from your UC)

Press once more the hold button

Call is retrieved

Click hangup button

No more calls on the calls screen and a sound is played to notify that call is over.

1. Call a number internal or external
2. Then repeat 4 and 5

You have a ringback tone and then the callee

The callee hangs up

No more calls on the calls screen

Call an internal user

You can only hear one ringback tone, and not two different ringback tones playing at the same time. 

Click hangup button before user answers

No more calls on the calls screen. 

After logout any current webRtc audio (ringing, icoming call) stops.

U logs into Web Assistant

U is logged in.

U rings another line L.

L starts ringing, U hears ringing sound played by browser.

While ringing, U clicks logout.

U is logged out, runging sound played by browser stop.

U logs into Web Assistant

U is logged in.

L calls U.

U sees incoming call dialog & hears incoming call audio played by browser.

While incoming call is ringing, press "Esc" to dismiss popup and click on logout

U is logged out, runging sound played by browser stop.

Test incoming WebRTC call

Call the WebRTC line

The assistant pops-up a call notification with caller id and answer/reject button, a ring sound is played

Reject the call

The call is rejected

Call the WebRTC line

Answer the call

The call is established, the level of the left progress bar follows the sound you hear, the level of the right one follows the sound you make

Put the call on hold

The call is on hold, both progress bars disappear

Hang up on phone side without retrieving the call

Call is terminated

Call once more, answer the call and put on hold

The call is on hold

Validates that the number of incoming calls for webrtc user is limited to two

The next caller should hear "user is busy" message  (#1567).

Using the standard phones call the WebRTC user and answer the call twice (once on Five)

Two calls - one established and one held (one call on Five)

Call the WebRTC user from another phone

The call is rejected

Caller hears that "user is busy"

Hangup current call

No active calls

WebRTC endpoint can have only two calls

Using the standard phone call the WebRTC user and answer the call 

The call is established

Start second call by typing number into search box

As other user accept the call

1. First call is in Hold
2. Second call is established

Try to start third call from the WebRTC line using the desktop assistant:

- type number in the search field

- try to call from favorites

- try to call from directory search result

Third call must be impossible

Cancel both calls

No current call

Call from the WebRTC user and answer the call at the destination

The is established

Start second call by typing number into search box

As other user accept the call

1. First call is in Hold
2. Second call is established

Try to start another call from the WebRTC line using the desktop assistant:

- type number in the search field

- try to call from favorites

- try to call from directory search result

Third call must be impossible

Hangup all calls

No current calls

Start a first outgoing call with webrtc

Call is established

Call the webrtc user from a standard phone

A bip tone every 10 second is heard to notify that a new call is incoming

Asterisk should disconnect webrtc call if there are no more RTP packets from webrtc.

rtptimeout = 20 seconds. To check if the value is up tp date, open sip_user.py file on Xivo or run: /usr/bin/xivo-confgen asterisk/sip.conf. All webrtc users should have rtptimeout defined.

(Issue https://projects.xivo.solutions/issues/718)

Open asterisk and run rtp set debug on

Make a call between U1 and W1

RTP packets are sent from both sides (callers can hear each other)

Close Web Assistant window without logging out

(only for Electra and before)

Electra, Freya:

• Asterisk disconnects the call "for lack of RTP activity" after rtptimeout

>= Gaia

• Call is hangup by xuc

Make a new call, check RTP activity (callers can hear each other)

Hold the call from phone

RTP packets are sent from one side

But call is NOT disconnected after rtptimeout interval

Resume the call

Hold the call from Web Assistant

RTP packets are sent from one side

But call is NOT disconnected after rtptimeout interval

Resume the call

Disable UDP on Xivo by: iptables -A INPUT -p udp --dport 10000:20000 -j DROP

RTP packet are not sent

Asterisk disconnects the call after rtptimeout

Enable UDP on Xivo by: iptables -D INPUT -p udp --dport 10000:20000 -j DROP

Make a new call, check RTP activity

Disable UDP on Xivo for 10 seconds, check RTP activity, and re-enable RTP on XiVO:

iptables -A INPUT -p udp --dport 10000:20000 -j DROP && sleep 10 && iptables -D INPUT -p udp --dport 10000:20000 -j DROP

RTP packets are sent from both sides

Call is NOT disconnected after rtptimeout interval

On Xivo disable VLAN interface

ip link set down eth1 (?)

RTP packets are sent only from W1

Call is NOT disconnected after rtptimeout interval

Open VirtualBox console of the Xivo

Run tmux

Open asterisk and run rtp set debug on

Disconnect Xivo from network from new tab:

service networking stop

Asterisk halts call due to RTP timeout

Message check_rtp_timeout: Disconnecting call appears

Restore network connection on Xivo, make new call

Restart XUC

RTP packets are sent from both sides

Call is NOT disconnected by XUC restart.

Web Assistant shows "Erreur fatale" window

Log out from Web Assistant

Call end

Test dtmf keypad to close when called user hangs-up.

Login with the CTI credentials of the user configured with the WebRTC line

Login OK, Registered in the navigator developer console.

WebRTC user calls the Phone user

Phone user answers
WebRTC and phone users can talk

WebRTC user opens the DTMF keypad

Keypad appears

Phone user hangs up

DTMF Keypad closes after call ends on the WebRTC user interface.

1. U1 logs into Web Assistant
2. U1 calls U3
3. U3 accepts

1. U1 is logged into Web Assistant
2. There is ongoing call between U1 and U3

U1 calls U2 by typing his number+Enter to search box

U2 rings

U2 accepts

Calls :

1. held call between U1 and U3 
2. ongoing call between U1 and U2

U1 clicks "attended transfer" button on held call between U1 and U3.

• there no ongoing call displayed in U1's Assistant
• U2 and U3 are connected by call (transfer was succesfull)

1. End all calls everythiong, logout from Assistant
2. U1 logs into Web Assistant
3. U1 calls U3
4. U3 accepts

1. U1 is logged into Web Assistant
2. There is ongoing call between U1 and U3

U1 calls U2 by typing his number+Enter to search box

U2 Rings

U2 refuses call

There is held call between U1 and U3

1. End all calls everythiong, logout from Assistant
2. U1 logs into Web Assistant
3. U1 calls U3
4. U3 accepts

1. U1 is logged into Web Assistant
2. There is ongoing call between U1 and U3

U1 calls U2 by typing his number+Enter to search box

U2 Rings

U2 accepts

Calls :

1. held call between U1 and U3 
2. ongoing call between U1 and U2

U2 hangs up

U3 hands up

No ongoing calls

U2 calls U1

U1 accept call

U1 calls U3

U1 transfer call

U1 is ringing

One ongoing call

One call on hold / one call in progress

No call in progress, U2 and U3 in conversation

U2 calls U1

U1 accept call

U1 calls U3

U3 answer

U1 transfer call

U1 is ringing

One ongoing call

One call on hold / one call in progress

One call on hold / one ongoing call

No call in progress, U2 and U3 in conversation

After direct pickup (*8) call is terminated and does not appear as ongoing/active in Web Assistant - "No current call."

Login to Web Assistant as User 1

User 2 calls User 3

User 3 is ringing
User 3 does not answer

In Web Assistant type User 3's number with *8 prefix (e.g. *81003)
Hit Enter or click the dial button

User 3 stops ringing
User 1 and User 2 are talking
 

User 2 hangs-up

User 1 has no call
User 2 has no call
Web Assistant shows no calls in list of calls - "No current call."

Repeat steps 2-4 and hang-up by User 1 (WebRTC user)

Login with different users to the webAssistent

Logged in

On the first user assistant search for the second user and make a video call from the search result

Calling user:

• Sees the video frame appearing in the call control
• Can see its local video feedback while call is ringing
• Can't see remote video while call is ringing (when call is not answered)
• In video frame, you doin't see the spinner as if the video was loading while call is ringing

Called user:

• receives a video call
• the incoming call notification indicates correctly an incoming video call

Answer the call

The call is established with video

Put the call on hold

The call is held for both participants (the video can be freezed)

Resume the call

The call should be resumed

Repeat once more hold/unhold

should still work

Repeat the test on the other operation system than yours (Linux/Windows)

Should work on both

Validates that the conference and transfer are disabled for video calls

Make a video call between two users (either from search result, either from favorites)

The call is established

Call one of the participants in the video call from a phone

A new incoming call is indicated, when user answers the video call is put on hold and the audio call is established

Check that you can't make conference or transfer call

Should not be possible

Repeat the test with establishing an audio call first and then the video call

The second call is audio only (because it's initiated by XUC which does not support video calls)

Tests the echo tests for video calls

Create a user echo test with mobile number *66

user is created

Login to the webAssitant as the WebRTC user and search for echo, than call the echo user in video (*55)

The call is established, when the introduction announce is terminated you get both audio and video echo, you receive back your video stream.

Test webrtc conference hold/resume 

Given U1 has webrtc line
U1 logs in to XiVO Assistant
U2 calls U1
U1 answers
U1 calls U3 via Assistant
U3 answers

U2 is on hold
U1 and U3 talk

U1 clicks conference

• U1, U2 and U3 talk connected in conference
• Make sure that
  • U1 is heard by U2 and U3
  • U2 is heard by U1 and U3
  • and U3 is heard by U1 and U2
• Conference is displayed in Assistant casll list

U1 clicks hold

U2 and U3 are on hold

U1 clicks conference

• U1-3 are in conference
• Make sure that
  • U1 is heard by U2 and U3
  • U2 is heard by U1 and U3
  • and U3 is heard by U1 and U2

Hangup

Given agent U1 has webrtc line
U1 logs in to CC Agent
U2 calls U1
U1 answers
U3 calls U1
U1 answers

U2 is on hold
U1 and U3 talk

U1 clicks conference

• U1, U2 and U3 talk connected in conference
• Make sure that
  • U1 is heard by U2 and U3
  • U2 is heard by U1 and U3
  • and U3 is heard by U1 and U2
• Conference is displayed in Assistant casll list

U1 clicks hold

U2 and U3 are on hold

U1 clicks resume

• U1-U3 are in conference
• Make sure that
  • U1 is heard by U2 and U3
  • U2 is heard by U1 and U3
  • and U3 is heard by U1 and U2

Test conference from XiVO UC assistant

Given A1 has WebRTC phone
A1 logs in to XiVO UC
U2 calls A1
A1 answers
A1 calls U3 via Assistant
U3 answers

U2 is on hold
A1 and U3 talk

A1 clicks conference

• A1, U2 and U3 talk connected in conference
• Make sure that
  • A1 is heard by U2 and U3
  • U2 is heard by A1 and U1
  • and U1 is heard by A1 and U2
• Conference is displayed in Assistant casll list

Open

https://XIVO_CC:8443/sample

Webrtc sample page opens

Login with W1

Init webrtc

Dial *55

You hear echo test