Boomea Webphone

This is a WebRTC phone built right in your Boomea desktop application or web browser! Never worry again about mobility. If you have access to a computer, you have access to your office phone system!

Features:

• Multi-line support
  • Allows for multiple simultaneous calls
• SRTP
  • The voice and SIP signaling are encrypted with TLS
• Hold / Unhold
• Mute / Unmute
• Call Park
• Transfers
• Input and output settings
  • Allows you to select the microphone input and speaker output to be used for your calls
• Integrates with your contacts
  • If you have integrated your contacts into Boomea, then when making or receiving a call from a contact, the webphone displays the latest communication history with that contact
  • If this is a new contact or number, you can add or edit a contact right from the webphone
• Integrates with your tasks and notes
  • If you have the Boomea productivity suite with notes and tasks, the webphone allows you to create or link the current call to a task or note.

Transfer Options

There are two transfer options available on the webphone: Blind transfer and Attended transfer. Here’s how to use them.

BLIND TRANSFER

Purpose: Blind transfer will forward a call to another number without any interaction between the operator and the number.

• Step 1: As the operator, start a “Blind Transfer” by clicking the dual arrow icon (see #1 on screenshot below) next to the call duration timer on the webphone’s active call page
• Step 2: Select the “Blind” option in the Transfer menu  (#2)
• Step 3: Enter the extension or phone number in the Transfer field  (#3) and click the “Go” icon  (#4)
• Step 4: The call will be forwarded and will ring the number you entered. The blind transfer is complete at this point.

ATTENDED TRANSFER 

Purpose: Attended transfer allows the operator to speak to the person who the call will be transferred to before the call is transferred.

• Step 1: As the operator, start an Attended Transfer by clicking the dual arrow icon (see #1 on screenshot below) next to the call duration timer on the webphone’s active call page.
• Step 2: Select the “Attended” option in the Transfer menu (#2)
• Step 3: Enter the number in the “Attended Call” field (#3) and click the “Go” icon (#4)
• Step 4: The initial caller will be put on hold at this point (#5) and your webphone will call the entered extension / phone number.
• Step 5: While ringing the destination number and while speaking to the person at the destination number, there are “Attended Transfer” options available to you.
  • Click “Complete Transfer”  (#6) to join the initial caller with the destination caller. As the operator, you will be dropped from the call and the transfer process is complete.
  • Click “End & Return”  (#7) to end the call with the destination number, cancel the transfer option, and pick up the initial call.
  • Click the “end call” icon  (#8) to end the call with the destination number and keep the initial caller on hold. In this state, you will need to pick up the call (by clicking the on-hold box (#5) at the top of the webphone page) before being able to transfer the call again (unless you use the Operator Console approach)

Network Panel and Call Quality

• Clicking the “Network” icon on the active call page will reveal a panel displaying dynamic values for latency, jitter, and packet loss. These metrics are meant to provide greater insight into network factors that can impact call quality.
• The metrics provided are:
  • Latency: This measures the delay in data transmission between the user’s machine and the server that is connecting the call.
  • Jitter: Congestion on the webphone’s network and packet loss cause packets to arrive at the receiver with varying amounts of delay. That delay is called “jitter” and it is measured in ping.
  • Packet Loss: Occasionally, transmitted packets of information will fail to arrive at their destination. The loss of packets is measured in percentage (lost packets/total packets). The higher the loss, the choppier the call quality will be.

• To decipher the values associated with the metrics, dynamic “grades” are provided in icon form. Green checkmarks signify a healthy connection. Yellow exclamations suggest degraded performance that may cause reduced call quality. Red blocks identify reasons for poor call experience.  The ranges for each metric are:
  • Ranges for latency are: 0-100 ms (green), 101-200 ms (yellow), and over 200 ms (red)
  • Ranges for Jitter are 0-15 ms (green), 16-30 ms (yellow), and over 30 ms (red)
  • Ranges for Packet Loss: 0% (green), 1% (yellow), and over 1% (red).
• It is quite possible to experience stable and adequate call quality on the webphone with one of the metrics showing red and the other two metrics showing as yellow. However, if call quality is degraded, these metrics can be a first step in identifying areas of concern.

Troubleshooting:

Sometimes technology lets us down. How do we figure out what’s wrong? Here are a couple simple steps to take to resolve issues with the webphone.

• Check your internet connection.
  • This is the most common cause. The webphone requires a stable internet connection in order to maintain exceptional call quality and ability to make and receive calls. Without an internet connection, the webphone will not function.
• Make sure you’ve selected the appropriate input and output devices.
  • If you’ve selected your PC microphone to be your input device but you’re talking into your headset, parties will find it hard to hear you!
• Double-check you’ve dialed the correct telephone number.
• Reload the Boomea app and try again.