Is my network VoIP ready? How much bandwidth do I need?
VoIP calls typically require 24kbps to 90kbps of bandwidth (200kbps is ideal), but video calling and conferencing can require 3 to 20 times more bandwidth. In planning for your network, estimate how many simultaneous calls your organization expects to have at any given time and multiply those calls by 200 kbps.
Our average customer has about 10% of the organization on the phone at any given time. As an example, if you have 20 employees and anticipate 2 simultaneous calls at any given time, the network would ideally have 2*200kbps, or 400kbps of bandwidth dedicated to VoIP calls. Note that if your company heavily utilizes your network for bandwidth-intensive activities such as streaming video, you should account for that traffic as well. No need for a QoS device; we simply suggest you account for your bandwidth needs.
This test is a general approximation of the quality of your VoIP connection. It does not test whether there are any active issues with your router that are impairing the quality of your connection.
Here are the typical issues associated with a network with under-allocated bandwidth:
- Packet Loss generally occurs when there’s not enough bandwidth on your network (usually <24 kps) for an acceptable call to take place, though hardware issues and Internet connectivity problems can also lead to this. Packet loss cause gaps and stutters in your call.
- Latency refers to the amount of time it takes a data packet to reach its destination. The lower the latency, the clearer your call will be.
- Jitter measures how much latency varies within your network, and it is often caused by network congestion and route changes.
Firewalls can complicate communications over a VoIP network. Firewalls sometimes perform Network Address Translation (NAT), a process OnSIP already takes care of with a “server-side” solution. We do not recommend disabling NAT traversal technologies, because of security concerns. But multiple NATs, SIP ALGs, SIP helpers, and any process that inspects packets can disrupt the quality of your call. We also do not recommend securing traffic by port range, but we do encourage users to open the router to our IP blocks for UDP traffic only, since the port for every call for audio will vary.