The desk phone was once an omnipresent in any work environment. But now, more and more companies are making the switch to VoIP. There can be several advantages to using browser-based telephony, including eliminating hardware costs and reducing telephony costs.
One of the most frequent questions that our customers ask us is whether their networks can support VoIP. We therefore thought we’d offer some ways to diagnose whether your network can support VoIP calls. We’ll also recommend some tools that you can use to test and optimize your network for VoIP calls.
Thanks to our great partner, Twilio, for many of these effective tips.
Network Per-Rep Bandwidth Standards
One of the first things you’ll need to take into consideration is whether there is enough bandwidth available for your reps to make and take VoIP calls. When determining the optimal network bandwidth required for your reps, you should take three things into consideration:
- The amount of bandwidth being consumed per rep without using a VoIp solution during peak network hours (in mb)
- The amount of bandwidth required by the VoIP solution
- The amount of “headroom” required (X 1.4 total)
As an example, let’s say that a group of sales reps is already using .5mb upload/download each during peak network usage hours. They would then also need to have at least .1mb upload/download for VoIP calls. The total would be multiplied by 1.4 (140%). In this scenario, each rep in this company would then need (.1mb + .5mb) x 1.4 = .84mb of bandwidth. If there were 10 reps (10 x .84), then 8.4mb upload/download would be required.
Upload/download speeds can be measured using a free online tool, Speedtest.net.
To measure per-rep/per-IP bandwidth usage, try Wireshark.
Tips for Optimizing Network Quality and Speed for VoIP
Packet loss occurs when one or more packets of data travelling across a network fail to reach a destination. The closer to zero, the better. Do not exceed 5%.
Latency (aka Ping)
A measurement of how much time it takes for a packet of data to get from one designated point to another. For best results, keep network latency under 50ms. Exceeding 200ms will noticeably affect quality.
Jitter is the amount of variation in latency/response time, in milliseconds. For best results, keep below 5ms. Jitter in excess of 10ms will likely affect quality.
To test for packet loss, latency and jitter use PingTest.
Related to jitter, bufferbloat occurs when your router is unable to transmit all the packets required and begins storing them in a queue. This causes latency and bursts and jitter which can noticeably affect conversation. To avoid bufferbloat, we recommend ensuring your router is configured with a low buffer size.
Using this tool, test to see if your buffer size is 100ms or less. If your buffer size is 100ms or more, see if your router can allow for adjustable buffer settings. Open source routers, enterprise grade routers and gamer-oriented routers may be able to provide the right configuration options and defaults.
By using these diagnostic tools you should be able to accurately determine whether your network can support using RingDNA’s VoIP solution. If not, RingDNA can easily be configured to work in tandem with your existing phone system.
For more IT tips that will help you maximize your Salesforce investment check out our free eBook.