The Quality Of Your VoIP Service Is Directly Related To The Quality Of Your Internet Sevice.
SpectrumVoIP's Hosted PBX and SIP Trunking services are delivered across the Public Internet and by definition are "best-effort". While we have gone to great lengths to architect our platforms and interconnections so that the components we control are redundant and perform optimally at all times, our service also depends on the performance of external components such as local power, broadband carriers, and Internet backbones.
Some internet connections may have less than desirable quality. Typically, we've found that packet loss or high latency nearly always occurs during the last mile of delivery. When IP packets are lost or significantly delayed at any point during transit, a momentary loss of audio will occur. Once uncovered and reported to your Internet Service Provider, they're usually able to easily rectify it.
For many of our customers, we install a Mikrotik router to help shape internet traffic to help with Quality Of Service issues. In the case where we do not have a router on site, we would recommend reaching out to their ISP.
Information regarding Firewall Rules that work best with SpectrumVoIP's services. (HERE)
What is VoIP?
VoIP stands for Voice Over Internet Protocol and it allows for phone calls to be made over the internet in lieu of traditional phone service. SpectrumVoIP's hosted telephone service uses VoIP technology to deliver phone calls to its customers.
Instead of sending the data through the copper telephone lines of the PSTN, when a user speaks into their phone, VoIP services convert that sound data into packets. Everything sent over the internet is transmitted as a “packet” of information, or data.
As long as the local internet connection is stable, consistent, and there are no firewall rules blocking or slowing down the packets from being sent or received, then the packets will be sent and received to allow you to have a live conversation over your internet phone.
What Factors Affect VoIP Calls?
Latency, Packet Loss, and Jitter
Latency can be defined as the time it takes for data packets to reach a destination. For VoIP, if Latency is too high, usually above 150ms, then it can cause frustrating calls where there is a delay in voice packets reaching their destination. This can result in callers accidentally speaking over each other.
Packet Loss can be defined as packets of information that do not reach their end destination. In a VoIP call, this could mean dropped audio in the range from a stutter to whole words and phrases. If there is a noticeable amount of packet loss, phone calls made through a VoIP system may appear staticky, jumbled, or stuttery. Even a 1% packet loss can “significantly degrade” phone quality.
Jitter can be defined as a variation in the delay of received packets. At the sending side, packets are sent in a continuous stream with the packets spaced evenly apart. Due to internet network congestion, improper priority queuing, or other configuration errors in network equipment, this steady stream can become lumpy, or the delay between each packet can vary instead of remaining constant.
How Can I Check The Quality Of My Internet?
Knowing what to check for regarding the quality of your internet is one thing, but how do I check the quality of my internet?
One of the most simple ways to check the general quality of your internet is to run a PING TEST.
A ping test is fairly simple to do. You can instruct your device to send a “ping,” a very basic data packet, to another device. The recipient device then sends a “ping” back, and the time it took to do all this is measured, most commonly in milliseconds (ms).
Run A Ping Test From Your Computer
To run a ping test from your own computer, open up the command prompt. This can be done by pressing the Windows Key + R then searching for "cmd" or by bringing up the Terminal window on a Mac. While the PC and Mac versions act slightly differently, they are both able to run this same ping test.
Upon opening a command prompt, you would enter the command:
ping -n 100 <hostname>
With hostname being your own choice of website or server. A common example to test is google.com or 126.96.36.199 (Google's DNS Server Address).
To test the connection to SpectrumVoIP, we would recommend using stratusp.spectrumvoip.com .
The above command will send 100 pings to the host address, and hopefully, return 100 pings. But if you send 100, and only 60 are received, you have discovered a 40% packet loss. Upon completing the ping, you should receive a message similar to this one:
You can ping as many hosts as many times as you would like. It is recommended to run the test multiple times, both with the same and new hosts to gather a large grouping of data.
Run A Speed Test That Tells Ping Speed
There are some speed test sites, such as SpeedTest.net or even Google that will let you run a Speed Test for your internet connection. They usually not only tell you your Maximum Upload and Download thresholds (how much speed you are paying for by your Internet Service Provider) but they often also tell you your ping speeds for the tests that they run.
Acceptable Ping Values For VoIP
While each case will be unique, there are some ballpark numbers that are good to shoot for.
According to Cisco, “One-way (mouth-to-ear) transmit delay should not exceed 150 ms (per G.114 [protocol] recommendation).” This would make the round trip delay be about 300ms which is where the human ear would be able to detect a significant delay in the audio.
What Do I Do Now?
I See Packet Loss - This will also show itself in the form of TimeOuts during a ping test. The best plan of action is to reach out to your ISP (Internet Service Provider) to make sure they are aware of network quality. If it's a small amount, 1-2% packet loss, then some ISP's may want to shrug it off because it doesn't affect 'most' internet activities, such as watching YouTube videos or checking emails. However, with VoIP, even the slightest amount of packet loss will result in bad quality. This is because services like YouTube and Email utilize a TCP connection which doesn't require that all the data packets arrive in order. If a packet gets dropped, then it can re-request the packet and fill the missing data packet when it arrives later. VoIP, however, uses a UDP connection that requires the data packets arrive in order. If a packet is lost
I See High Ping Times - Ping times averaging above or near 150ms. This is also something to get with your ISP (Internet Service Provider) about. Be sure to run your ping test to multiple hosts (try google.com , stratusp.spectrumvoip.com , 188.8.131.52 , 184.108.40.206) to make sure that the high ping times are not coming from only one host site. High Ping Times could have to do with the quality of the internet that your ISP is able to provide you with your current equipment. They might be able to upgrade your equipment or check for physical public cabling that might be causing an issue reaching your location.
I See A Wide Range Of Ping Times - If the ping tests come back with inconsistent results, such as ranging from 5ms to 300ms ping times, it would be a good idea to reach out to your ISP (Internet Service Provider) to confirm that they are not seeing any issues on their end that could be causing the jitter. It is also possible that this jitter can be caused from internal network congestion. If you have too many local network devices that are not routed and configured properly to prioritize your voip traffic, your local network may be congested, similar to having too many cars on the highway, too many devices on a limited network can cause data flow to be inconsistent, and thus jittery.
Other - Check your router's settings. We have some recommended firewall rules that can be put in place on your network router to help with internet traffic. (HERE). For many of our customers, we install a Mikrotik router to help shape internet traffic to help with Quality Of Service issues. In the case where we do not have a router on site, we would recommend reaching out to their ISP.