Dev Documentation

Change Logs

Updated Mar 04, 2021

Release Highlights 1.2.1
AIVA Connect Client - Google Chrome

Executive - Assistant Functionality for WebRTC

WebRTC now supports Executive & Assistant Functionality! Executives will be able to select from available assistants that will be able to intercept calls.

Features include:

  • Simultaneous Ring
  • Forward If I Don't Answer
  • Custom Outgoing Caller ID
  • Call Recall
  • Call Rollover
AIVA Connect Client - Google Chrome

Set Disposition Code While on a Call

If users have disposition codes available to them, they are now able to access and set a disposition code to that call. After a call ends, and moves into auto wrap up, it is still possible to change the disposition code.

AIVA Connect Client - Google Chrome

PLEASE NOTE: The WebRTC will accept the last set disposition code before Wrap-Up ends. On the next inbound call, the disposition codes will be cleared and the user will have to select a new one for that specific call.

WebRTC Call Center Queues

For users who are in possession of a Call Center Supervisor license, they will now have access to a Call Center Queue Statistic Wallboard.

WebRTC Sample Admin Dashboard View

Vahram Barseghyan | Microsoft Teams

WebRTC Sample Agent Dashboard View

Vahram Barseghyan | Microsoft Teams
Previous Releases
Release Highlights 1.8
AIVA Connect Client - Google Chrome

Call Waiting Tone

This feature is disabled by default. When enabled, if a user is on a call, their next incoming call will be a low audible beep instead of the standard WebRTC ringtone.

To enable the Call Waiting Tone:

1. Go to Settings

2. Select "Call Settings"

3. Scroll down to "Call Waiting". Select the toggle to enable the alternative call waiting tone.

AIVA Connect Client - Google Chrome

Accessibility (for JAWS) Hot Keys

By default, the WebRTC supports the following hot keys for handling calls.

AIVA Connect Client - Google Chrome

If a user has an accessibility screen reader, such as JAWS, the following alternative hot keys are available.

AIVA Connect Client - Google Chrome

E911 Alerts

Whenever a user opens up the WebRTC client, they will be informed of our E911 policy.

To close the E911 message, select "OK".

AIVA Connect Client - Google Chrome

Bandwidth Test - On Start

Whenever a user opens the WebRTC client, it will run a quick bandwidth speed test. If a user is below a certain threshold, then they will see the following warning message.

To close the Bandwidth Warning, select "OK".

AIVA Connect Client - Google Chrome

Bandwidth Test - Manual

If a user would like to manually re-take the Bandwidth test:

1. Go to Settings

2. At the bottom of the screen, select "Speed Test"

3. This will run the speed test.

AIVA Connect Client - Google Chrome

The most recent speed test result will display on the screen.

To view the historic results of the speed test:

1. Go to Settings

2. At the bottom of the screen, select "Results".

3. This will show your last 20 speed tests. After 20 entries, the earliest entry will automatically be deleted.

AIVA Connect Client - Google Chrome
Release Notes: 1.6

1. Directory Toggle – Group / Enterprise (Settings / Contacts Settings / Enterprise Directory)

Based on your user profile, users will have the ability to toggle between whether their directory displays the entire Enterprise, or just contacts that are shared at the group level.


2. Real Time ACD and Call Status between AIVA Clients – Functionality and Toggle (Settings / Contacts Settings / Personal Directory)

WebRTC now displays the presence status of users. This includes when a user is: online, offline, or in a call.


3. Real Time ACD Sync between AIVA Client and 3rd party software – Functionality and Toggle (Settings / Profile Settings / ACD Status Sync)

Previous versions of WebRTC did not visually synch when 3rd party clients made updates to an agent’s status. This request has been completed, and 3rd party software will now synchronize with the WebRTC client.


4. Extensions and Phone numbers now displayed in directory and personal contacts & Extension takes priority when dialing out using quick buttons.

Previous versions of WebRTC did not include the Extension as a field in the Contacts page. This has been added and users will now see the extension of users in the Contacts page.


5. Auto Answer – Functionality and Toggle (Settings / Call Settings / Auto Answer)

This is an optional toggle for users to enable auto answer. If a user has this enabled for them at the system level or at the user level, calls will be automatically be answered by the WebRTC client if the user is available to take a call. If a user is already on the call, the next incoming call will be offered to the user, but will not be automatically answered.


6. Keyboard Shortcuts for main phone functions (Settings / Information)

WebRTC now supports Hot-Key shortcuts for basic call controls. Please refer to the WebRTC 1.6 for more information.


7. Auto Hold Switch (Settings / Call Settings / Auto Hold).

WebRTC by default, uses Auto Hold. For example, when switching between lines, calls are automatically placed on hold. However, this has now become an optional toggle, and users may turn off Auto Hold from their Settings.


8. New QOS Test (Settings / QOS Test).

The QoS Test in WebRTC now includes advanced reporting. These metrics will be available within the email the user receives regarding their VoIP test. For more information, please refer to the WebRTC 1.6 user guide.


9. Disposition Codes display correctly per queue now.

Previous versions of WebRTC displayed all disposition codes that were assigned to queues. WebRTC now intelligently maps disposition codes based on the incoming call. For example, if a call comes into Queue A, the user will only be offered disposition codes associated with Queue A.


10. Some special characters no longer filtered out when making calls – Namely “*” and “#”

A bug in earlier versions of WebRTC did not support asterisks and pound signs. This has been resolved, and WebRTC now supports these keys.


11. Audible tone when making a call transfer

When a user transfers a call, they will hear an audible tone.


12.Call center type specific feature adjustments

Based on a users license type and seat, they will only see the features that are available to them. For example, calling out of queue options will be displayed to Premium seat holders, rather than all agent license types.


13. Visual bug fixes and Performance Improvements

Release Notes 1.3

1. Feature  Change Log

Added “About App” in “Information” tab under settings, to highlight new features and provide version information for the app.

WebRTC 1.1.3 Release Highlights  - Word

To access the WebRTC Change Logs, users can navigate to the “Information” section within the “Settings” tab. Release highlights will be stored here along with a link to the associated documentation.

2. Feature/Fix  Call Timer

Several users had reported that their call timer was freezing or displaying a delay. This issue has been corrected and now displays the call time reflected in BroadSoft.

3. Feature  Notification & Notification Settings

Notification ribbons appear at the bottom of the WebRTC screen. These will fade after a few seconds, but users can also manually dismiss a notification by selecting the “x” on the far-right corner of a notification.

WebRTC 1.1.3 Release Highlights  - Word

Notifications are enabled by default, but if you would like to disable notifications, navigate to “Notification Settings” within the “Settings” tab. From here, you can toggle this setting on or off.

WebRTC 1.1.3 Release Highlights  - Word

4. Call Merge / Split / Conferencing

Users can now start a conference/merge calls while on an active call by selecting the "Merge Calls" button. For more details on how to enable conferencing, please refer to our WebRTC User Guide, located here:

https://docs.aivaconnect.ai/a/1284031-webrtc-1-1-3-quick-start-guide

WebRTC 1.1.3 Release Highlights  - Word

5. Attended Transfers

Attended (Warm) Transfers are now supported within the WebRTC phone. When performing an attended transfer, the initial call will be placed on hold, and the user will be able to call a consultant. When ready, they will be able to select “Complete Transfer” to transfer the call.

For more details on how to perform an Attended Transfer, please refer to our WebRTC User Guide, located here:

https://docs.aivaconnect.ai/a/1284031-webrtc-1-1-3-quick-start-guide

6. Sign into Queue

Users can now sign into individual queues from the WebRTC client. Simply navigate to the ACD Status drop down and select which queue you would like to sign in/out of.

WebRTC 1.1.3 Release Highlights  - Word

7. Queue Calling

Users now have the option to call out of queues that they are skilled for. If their BroadSoft configuration allows them to call out of a particular queue, then the user has the option to change the outbound number from the client to display as either their personal DID or their specific queue’s DID.

For more details on how to change your “Outgoing Call” number, please refer to our WebRTC User Guide, located here:

https://docs.aivaconnect.ai/a/1284031-webrtc-1-1-3-quick-start-guide

WebRTC 1.1.3 Release Highlights  - Word

8. System Performance Enhancements

9. UI and Theme modifications for both Light and Dark modes

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