TalkPro Minimum Network and System Requirements

Workstation Requirements

Minimum Requirements

Processor: 1.4GHz or better
*2.0 Ghz Pentium Dual Core or AMD X2 equivalent (Net book CPU's such as Intel Atom not recommended)
Operating System: Windows 7 (32-bit / 64-bit) and above. Mac 10.2 and above
RAM: 4GB
Free Space: 20GB or 10% of main drive
USB Port: 2.0 (at least 1 free)

Recommended Requirements

Processor: 2.0GHz Dual Core or better
*2.0 Ghz Pentium Quad Core or AMD equivalent (Net book CPU's such as Intel Atom not recommended)
Operating System: Windows 10 (32-bit / 64-bit) and above. Mac 10.2 and above
RAM: 8GB
Free Space: 20GB or 10% of main drive
USB Port: 3.0 (at least 1 free)

Note: Using a faster computer will provide a more reliable connection when non-Convoso programs (like your CRM or other web applications) use system resources heavily. Under-powered computers can exhibit choppy audio (“jitter”) that may result in a terminated phone connection.

Latest stable version of Google Chrome browser. (Google Chrome)

USB cabled headset with built-in echo cancellation technology (We Use: Jabra UC VOICE 750 MS)

Cat5/6 grade ethernet cabled network connection. (Wifi is not supported)

Additional Note: Wireless/personal hotspots operate in a different wireless protocol known as MiFi.  This is different than traditional WiFi and results can vary.  Due to the unpredictable nature of MiFi connectivity with VoIP, mobile/personal hotspots are not supported at this time.

Bandwidth Requirements

You'll need to make sure that you have enough bandwidth to support the number of simultaneous calls you expect your agents to make. The bandwidth used is 64 kbps per phone call (upstream and downstream). It is important to note, this number is only for the audio traffic.

Other actions in TalkPro Contact Center will send/receive data, so more headroom is required. Also, other tabs the agent may have open, such as email or CRM will be consuming bandwidth as well. It's not possible for us to say exactly what you might need, but as a rough rule of thumb, 1Mbps per person sharing the connection is a good start. For example, if there are 100 people in your office, we recommend 100Mbps symmetrical connection. (It is crucial to have adequate
download and upload bandwidth for the number of users). 

• 1 Mbps of Convoso dedicated upload/download bandwidth required per agent.

• Bandwidth line should never exceed 60% its maximum capacity or VoIP quality can be compromised.  In terms of bandwidth, more available bandwidth is always better for your whole network!  

 As additional traffic on the same network can impact audio quality, here are other suggestions you should follow as well:

• We highly recommend a wired network connection over a Wi-Fi connection. This will generally provide a more consistent and better quality network connection.
• Don't run any network-intensive applications on the computers, such as internet radio or streaming video, or run significant uploads or downloads that might compete with your audio. Close unused desktop apps that might also hoard CPU %
• Check with your IT dept to see if higher Quality of Service is possible for your audio connection.
• Open network ports in your router / firewall / antivirus software (advanced info in table below)
• WebRTC (Chrome/Firefox browser)
• TCP: port 80, 8080 and 443
• UDP: Server port: 10,000 – 30,000. Client will select any available port from the ephemeral range: 1,024 to 65,535.
• If your router supports QoS, prioritize the ports mentioned above, or the IP address of the computer(s) making calls.
• If your router includes SIP Application Level Gateway (ALG) function or Stateful Packet Inspection (SPI), disable both these functions.

Additional Requirements

We highly recommend implementing Quality of Service policies and content filtering in order to prevent agents from participating in bandwidth-heavy activities which can impact call quality. 

• Datacenter-grade firewall (we recommend: Fortigate 300A or Juniper SRX240).
  • Researching a brand new office router?  Check the Make/Model against known VoIP Compatibility Lists to avoid big headaches during configuration/troubleshooting.
• Quality of Service (QoS) is required to guarantee best call quality possible for VoIP, and must be given priority over other network traffic
  • Would you rather a youtube video experiences issues or your VoIP phone call? QoS lets you determine the 'priority' of traffic in your network.  For the best voice quality; grant the port range 10,000-20,000 UDP the HIGHEST priority in your routers QoS settings.
    • Every router has a sightly different method of implementing QoS.  Please see the manufactures website or manual for details on setting up QoS on your device.

Convoso Network/VoIP Requirements

Client’s Network Firewall must allow access to the following services

• Port 5060 TCP/UDP
• Port 10k-20k UDP
• Port 8080 TCP
• Port 80 TCP
• Network IP Range 208.78.136.0/22
• Network IP Range 66.85.240.0/21 

• STUN Servers:
• l.google.com:19302
• l.google.com:19302
• l.google.com:19302
• l.google.com:19302
• l.google.com:19302

• If your router includes these function, please have these disabled.
• Disable SPI (Stateful Packet Inspection) = Causes delay in Real-time Traffic such as VoIP.
• Disable DPI (Deep Packet Inspection) = Causes delay in Real-time Traffic such as VoIP.
• Disable SIP ALG = Can result in SIP Traffic routing issues.
  
  

Further Networking Info:

WebRTC is supported natively in most modern browsers, however, Omni Contact Center only officially supports Google Chrome. (Edge and Firefox are unsupported).

WebRTC usually works without a problem using inbuilt networking technologies (STUN and TURN). However, environments with very restrictive firewalls may require some setup, the details below have further information for your IT-networking department.

WebRTC client connects using the following details:

† The client will select any available port from the ephemeral range. On most machines, these means the port range 1,024 to 65,535. 

We highly recommend implementing content filtering in order to prevent agents from participating in bandwidth-heavy activities which can impact call quality.