Call quality is always a primary consideration for anybody looking to utilize a phone system. Whether a call suffers from echo, choppiness, or delay, the disruption of a given call can result in anything from minor annoyances to serious impediments. But oftentimes, network admin can implement solutions on their own networks to ensure that these issues disappear.
While we have limited control over what happens to packets outside our network, OnSIP has engineered its system, network, and Internet connectivity to minimize packet loss, jitter, and latency.
- The fact that we are geographically distributed allows us to chart the shortest path for call routing.
- We peer directly with Tier 1 service providers, allowing us to avoid the network hops that can often degrade call quality.
- We maintain a physically separate network for real time media traffic to avoid congestion.
- We over-provision capacity and take a proactive approach to preventing congestion by utilizing self-limiting and fixed rate protocols
At all times, by design, we operate at 0% packet loss and less than 1ms of latency across our network. However, as Robert Bellovin of VoIP and telecom research company Software Advice says, reviews call quality symptoms that can occur on the last leg of the call— a business' network.
"Many businesses do not have a dedicated IT staff, but there are usually easy fixes to many of the common call quality issues users might experience," said Bellovin. The following diagram illustrates common call-quality issues and the network devices that are known to cause them. (Image courtesy of Software Advice)
Echo, one of the leading disruptions of VoIP calls, occurs when users hear voices repeated (either theirs, or their caller's, or both) that overrun the current conversation. In many cases echo is a symptom of acoustic feedback from the phone of the party you are talking to. Your voice travels across the phone system to the other party, their phone's speaker plays the sound, then their phone microphone picks up on that sound and it is transmitted back to your phone as an echo
Fortunately, this disruption is often easy to fix. By lowering the handset volume of both users, the phone's microphone does not catch conversational snippets from the phone's earpiece, the main cause of repeated voices. When using a softphone, it is normally best to use a headset, so that the computer's speakers do not interfere with the computer's microphone. More info here.
Choppiness, another common call-quality issue, occurs when the call's audio is spliced, so it sounds like the conversation is coming in unconnected clumps. This can severely hamper the flow of a call, and it is mostly caused by packet loss. A usual suspect in this case is the firewall configuration. Generally, we have found that OnSIP works great with all sorts of routers and firewalls so long as:
- Any SIP ALG has been disabled (this is a function on some routers)
- Any SIP firewall has be deactivated (this is a function on some routers)
- Phones are located behind a single NAT server on your internal network
No ports are required to be opened in order for the OnSIP Hosted PBX to work. The phones inside the network make the initial communication out to OnSIP and therefore open the port. All packets coming from OnSIP are in response to packets that originated on your network. More info here.
Packet Loss, Jitter, and Latency
Packet loss, jitter, and latency can act as delaying factors, which can ultimately result in missing or unintelligible portions of the call.
- Packet Loss can be caused by too much traffic on the network resulting in packets being dropped, incorrectly configured routers, hardware failures, as well as a host of other issues. Whatever the cause, lost packets result in gaps and stutters in communication and can greatly affect the perceived quality of communication.
- Jitter is variation in packet latency. The effects of jitter can be mitigated by storing voice packets in a jitter buffer upon arrival and before producing audio, although this can increase delay.
- Latency is the time from the source sending a packet to the destination receiving it. As long as packet loss and jitter are avoided, latency tends not to be an issue unless it is extreme.
For more info on steps you can take on your end to ensure optimum call quality, check out this Knowledgebase article. To quickly measure the quality and performance of the Internet connection between your network and our servers, try our VoIP test (Note: Java required).