Agents may encounter issues making calls from the browser. This article explores common suspected elements that may lead to this trouble.
CallHub allows agents to make calls after joining the campaign at the agent's console. There are three options to make calls web browser, dial-in, and phone.
You might encounter a specific scenario(s) while making the browser calls, like getting kicked out of the campaign or facing issues with call quality. Let's understand the most suspected elements that can affect your web browser calls.
CallHub's calling experience depends on your browser compatibility. It would be best to ensure you have set the recommended values in your browser settings.
- Run a CallHub Browser Check on your Chrome Browser from here: https://app.callhub.io/browser-check/. You need to run a browser check before making calls to understand if you have the latest version of the browser, set up the microphone correctly and if your browser supports the Web RTC plugin. A typical browser test from the agent console will look like the below image.
- Perform a test by clicking on this link WebRTC Test as well. This test will help you check for microphone, camera, and network connectivity issues.
- You can test your connection with networking testing tools here. In addition, we recommend the following conditions for a smoother browser calling experience.
- All agents should have 5-10 Mbps(download/upload) of dedicated internet bandwidth. By dedicated bandwidth, it means you should not share the calling platform with other applications. E.g., If you are on a 100 Mbps connection, it is advised not to have more than 10-12 agents making calls on the same network.
- Reduce the number of devices operating on the same channel(network connection), this way; you can reduce the packet conflicts. We recommend you have a dedicated network channel that agents can use.
- Preferably use Ethernet(LAN connection) and not WiFi, as the wireless connection is susceptible to packet loss and can hamper the call quality.
- Avoid data transfer, downloads of huge files, video streaming, etc., on the same network or internet connection while making calls. Such activity can result in a faint voice from the other side or disturbances while you’re talking.
Latency | <200 ms |
Jitter | <30ms |
Packet Loss | <3% |
Note: Work with your IT administrator to check these settings.
Suppose you have configured your office network with a VPN or Firewall for security and cannot join the campaign on CallHub. In that case, you might need to consider troubleshooting the below scenarios.
- Check if the UDP ports are open in your router/firewall/antivirus software for the Web RTC connection to work.
- HTTP/HTTPS transports: Allow traffic from these source IP ranges to access your HTTP (TCP port 80) and HTTPS (TCP port 443) web applications
- SIP Transport: Allow traffic from these source IP ranges and source ports UDP 5060, TCP 5060, UDP 5080, TCP 5080, UDP 5061, TCP 5061, UDP 5063, TCP 5063
- RTP transport: Allow traffic from these source IP ranges to UDP port range 16384 to 32768
Note: Work with your IT administrator to check these settings.
Browser calls are dependent on media devices such as microphones/headsets for call connection. Therefore, you need to check the following things:
- Ensure the hardware is correctly connected and you have selected the right microphone and speakers in the software settings of your device.
- Use a USB headset instead of a built-in computer microphone. Allow the Chrome browser to access your media devices when prompted.
- To avoid distortion during calls, reduce ambient noise such as nearby electronic devices. Adjust the microphone distance from the mouth (too close can cause audio clutter and increase the disturbance).
What if none of these steps work?
If the Browser connection is still not working, please switch to a Phone or Dial-In connection preference, and we will check the issue in the meanwhile.
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