How should I optimise our company network for Cradle?
There are two main areas of concern when it comes to setting up your IT to ensure good voice quality for you and the other party: Your computer or mobile, and your network. For advice on your computer and headset setup, we recommend checking out our article on headset selection. Read on to make sure your network is set up correctly.
Individual Computer Setup
First, ensure that you're using a computer with at least 4GB of ram, a modern processor that can handle the applications that you're running, some free hard drive space, and a firewall setup that you understand. If your network is set up with a firewall, you probably don't need one on your laptop (or it may be managed by your domain).
We recommend that you always use a hard wired ethernet connection. WiFi introduces jitter (or more correctly packet delay variation). This is where internet packets arrive out of order. This will cause voice quality issues if it gets above a certain threshold. WiFi can also get locally congested and result in packets being dropped. Both of these issues will be heard as a digital voice or if they're more serious, as a total loss of voice.
- If your router includes SIP Application Level Gateway (ALG) function or Stateful Packet Inspection (SPI), disable both of these functions.
- Ensure that your network respects DSCP headers. Cradle voice packets are sent with a DSCP header tag of 46 - expedited - so should take highest priority across your network (and the internet depending on your ISP). Please read this if you're using Windows to ensure DSCP headers are respected.
- Avoid buffer bloat. This can be caused either by a particular router Quality of Service (QoS) implementation selected, or particular router vendor defaults. We recommend ensuring your router is configured with a low buffer size. A test for this is here.
- Bandwidth minimums. Allow 50 kbits/s for each concurrent phone call that you intend to make. You need to test your internet connection to ensure this minimum. You may need to set up QoS on your network to ensure that you always allow VoIP to have a minimum of 110 kbits/s per call you want to make at a maximum, otherwise large downloads or similar heavy network events could interrupt your calls. Start by reading through this basic setup page.
- Whitelist our required IP addresses (see below) in order to ensure that traffic can pass through your firewall.
If you have to be on Wifi
- Ensure that you have the correct number of Wifi Access Points for the number of devices connecting to them and the size of your building. Not too many. Not too few.
- Ensure that you have the power settings on the Wifi Access Points set appropriately to ensure that your devices only connect to one AP (and see weak signals from one or two others).
- Make both 2.4 GHz and 5 GHz available so that as many devices as possible can connect of different frequencies (this should reduce congestion on 2.4 GHz which is typically very congested, and suffers from all sorts of other things using the spectrum).
- Turn off Wifi on devices that are connected using Ethernet.
- If you'd like more depth regarding this topic, please read this blog post.
Jitter < 30 ms
Round Trip Time < 150 ms
Bandwidth > 500 kbits/s minimum plus 50 kibts/s per call
Packet Loss < 2%
If you'd like more advice, please get in touch with us. We can analyse calls and help pinpoint where issues may be creeping in.
IP Addresses and Domains to Whitelist for Real Time Traffic
If you can whitelist IP address ranges to send UDP traffic to your machines, whitelist these ranges on your network:
|Region ID||Location||CIDR notation|
|us1||US East Coast (Virginia)||22.214.171.124/23, 126.96.36.199/23|
|us2||US West Coast (Oregon)||188.8.131.52/24|
The server side port used for RTP will be between 10,000 and 20,000, and the client side port will be between 1,024 and 65,535.
Signalling and Call Setup
Whitelist TCP traffic to the following addresses: