Common VoIP Problems

IP phone systems and unified communications (UC) platforms offer many business benefits, but are frequently subject to call quality problems that can negatively impact the user experience.
These problems occur in enterprise-class IP phone systems such as Cisco and Avaya as well as cloud-based VoIP and UC services such as Vonage and RingCentral.
At a high level, all of these problems are related to Quality of Service (QoS), which involves the measurement and optimization of transmissions over a network. QoS is especially important when it comes to the transmission of interactive applications over IP networks.


Use this list to jump to tips on troubleshooting a particular issue.

Dropped or disconnected calls
One-way audio
Choppy calls
Echo
Static
Packet loss

Types of IP Traffic

The Internet is a “best effort” network, meaning that data packets get delivered when they get delivered. Transmission Control Protocol (TCP) assumes that packets will get lost and sent out of order, and uses error checking, retransmissions and other techniques to ensure that all the data gets there eventually. That’s fine for an email or a web page — no one will notice a few milliseconds of delay. However, delays are very noticeable with voice calls, video conferences and other latency-sensitive apps.
While TCP requires a formal connection between two endpoints, User Datagram Protocol (UDP) is connectionless. Packets can be sent without negotiating a formal connection, and there’s no error control. UDP is often used for VoIP because it keeps the stream of data flowing without the delays caused by error checking and retransmission. If a few packets are lost, it typically won’t degrade call quality very much.

Types of IP Traffic

The Internet is a “best effort” network, meaning that data packets get delivered when they get delivered. Transmission Control Protocol (TCP) assumes that packets will get lost and sent out of order, and uses error checking, retransmissions and other techniques to ensure that all the data gets there eventually. That’s fine for an email or a web page — no one will notice a few milliseconds of delay. However, delays are very noticeable with voice calls, video conferences and other latency-sensitive apps.
While TCP requires a formal connection between two endpoints, User Datagram Protocol (UDP) is connectionless. Packets can be sent without negotiating a formal connection, and there’s no error control. UDP is often used for VoIP because it keeps the stream of data flowing without the delays caused by error checking and retransmission. If a few packets are lost, it typically won’t degrade call quality very much.

QoS and the Internet: A Primer

The biggest problems faced by VoIP are packet loss, latency and jitter

Packet loss, latency and jitter are related to the speed and order of delivery of voice packets. As the name implies, packet loss occurs when packets do not reach their destination at all, while latency and jitter occur when packets arrive late, out of order or at uneven intervals that cannot be offset by buffering.

All of these problems are correctable within internal IP networks. Network engineers use techniques such as packet prioritization, application classification and queueing to ensure quality of service for interactive applications. VoIP packets are marked with headers identifying them as high-priority traffic, and switches and routers are configured to recognize those markings and map them to the appropriate Class of Service (CoS) values.

Your Packets Need to Get Their Priorities in Order

In order to maintain quality from end to end, all devices along the way need to be inspecting the headers and honoring these CoS values that most routers ignore.
When VoIP calls hit the public Internet, all bets are off. Internet service providers (ISPs) generally ignore markings originating from outside their own “trust boundaries,” so the customer’s CoS settings are lost. VoIP data packets traveling across the ISP’s network receive only best-effort service. That’s where most call quality problems occur.

Packet headers are visible only to the devices handling the packets. Users can’t see the headers, but devices that are forwarding the packets—routers—can check the type of service header (i.e. CoS/QoS bits). Unfortunately routers ignore these marks on public Internet.

In order to maintain quality from end to end, all devices along the way need to be inspecting the headers and honoring these CoS values. External networks (WAN) can purchase a private connection from end to end that honors high priority markings. Any point-to-point network can honor packet priorities. The most common such networks are MPLS (Multi-Protocol Label Switch). Both forms of network are quite expensive, MPLS is more than 100 times as costly per bit than the public Internet.

Alternatively, InSpeed’s service creates an overlay network on any network connection which then honors packet priorities. The overlay is created between the InSpeed premise device and InSpeed virtual devices in the cloud. InSpeed uses a variety of proprietary techniques to maintain quality—latency, jitter, and loss—and doesn’t rely on packet markings.

Common VoIP Problems in Detail

Expand or close sections below as needed, to go directly to the VoIP problem you’re fixing, or see the table of contents at the top.

Identifying the Problem: Dropped Calls

A dropped call is a call that ends before you intentionally disconnect. You may hear a fast-busy signal or no sound at all. Troubleshooting this problem is difficult given the number possibilities involved. The issue could originate in your local network or devices, or your connection to the Internet.

If you’re not losing the whole call, just portions of the conversation are dropping out, see packet loss below.

Potential Causes of Dropped Calls

Do you get a fast-busy signal?

If the dropped call goes immediately to a fast-busy signal, it could be:

  • another device on your local network taking control of the line or introducing interference.
  • a problem with the signaling between two IP phone systems.
  • a firewall configuration issue.

Does the Dropped Call Result in Silence?

If the dropped call results in silence, however, it may be due to heavy traffic on your Internet connection.

Does the Dropped Call Eventually Come Back?

If you stay on the line and the call eventually comes back, you can be confident that latency is the problem, see below. Heavy traffic on your Internet connection has pushed the call data to the back of the queue.

Solutions to Fix VoIP Networks Dropping Calls

One way to reduce dropped calls is to use a “soft phone” rather than a physical one. Computer and smart phone applications such as Skype, Zoom, and Google Hangouts use much smarter codecs that do a much better at working around network difficulties. Of course, everyone on the call needs to be using the same application, which is a disadvantage.

Adding bandwidth can resolve dropped calls—at least until your Internet traffic volumes increase again. Or get InSpeed to solve dropped calls once and for all.

Identifying the Problem: One-Way Audio

One-way audio occurs when only one party can hear the other. The connection is good, so you have to figure out what’s preventing the sound from reaching the party who can’t hear.

Causes for One-Way Audio in VoIP Calls

First, check to make sure that the desk phone or softphone that’s experiencing the issue is working properly.

If you eliminate the possibility of a faulty phone, the issue is probably due to:

  • network address translation (NAT) or
  • firewall settings that block traffic from the Internet.

Fixing One-Way Audio

  1. Simplify the connection by plugging the phone directly into the router and make a test call.
  2. If you still get one-way audio, determine if the router is giving the phone a public or private IP address.
  3. If it’s a public IP address, the problem is with your VoIP provider.
  4. If it’s a private IP address, disable DHCP on the router or configure your router to enable port forwarding.

Identifying the Problem: Choppy Calls, Jitter and the Robot

Choppy calls are one of the most common VoIP quality issues. They are the result of jitter, which technically refers to the variability of latency across transmissions. In layman’s terms that simply means that voice data packets arrive out of order because they followed different paths and/or experienced differences in delays.

Jitter occurs due to inadequate bandwidth or a lack of prioritization of voice data packets.

Muffled Voice and Robot Voice

Both muffled voice and robot voice are symptoms of packet loss concealment.

Robot voice can happen in any audio device (phone/softphone/computer/smart phone) dealing with packet loss and repeating packets to replace the lost packets.

On some devices the voice is muffled, such that it is so soft it can be difficult to understand. This is the same problem as robot voice, handled differently by the device. Soft volume is dealing with a low rate of packet loss and the device is turning the gain (volume) down in hopes the listener won’t notice the device’s attempt to cover up the loss.

Fixing Jitter (Choppy Calls)

  • Adding bandwidth may or may not resolve this problem. One way to do this is to have additional routers, as this will increase the data packet transfer rate that may be slowing  your network.
  • Another way to add bandwidth is to get a gaming router.
  • One trick for fixing jitter is to use a cloud service, then make your phone calls using a VoIP app on a smart phone. You’ll want to use Wi-Fi for the smart phone calls, which means you may need to upgrade your Wi-Fi router to one that handles voice.
  • An MPLS link will provide prioritization for VoIP traffic, but that can be expensive. MPLS is more than 100 times as costly per bit than the public Internet.
  • We humbly suggest InSpeed QoS can ensure that interactive applications such as VoIP and UC are given priority over other data transmissions.

Identifying the Problem: Echo and Latency

Echo is typically caused by latency. VoIP data packets are delayed, causing you to hear the sound of your voice repeated after you’ve spoken. Latency occurs even in traditional phone connections because of the time it takes for the sound to physically move from the phone set across various networks and switching systems and reach the phone set on the other end. However, it’s seldom noticeable because of the extremely low latency of traditional telephone networks.

With VoIP, echo can be very noticeable due to the extended delays that often occur when packets travel over the public Internet. Once latency reaches 300 milliseconds, call quality has noticeably degraded.

Getting Rid of Echo in VoIP Calls

Note that echo can also be caused by sound coming out of the speaker looping back and coming in the microphone from the far end. In the old telephony world, the delay was short—less than 50 ms—so the caller rarely noticed. In VoIP the delay is longer and as it approaches 150ms the echo becomes a nuisance. In that case, use a soft phone as it’s codec may have a better echo canceller. In that case the solution to echo is simple: turn off one of the mics or devices. Otherwise, use a soft phone as its codec may have a better echo canceller.

For latency, there are echo cancellation settings and add-on devices that can help reduce the problem. You may be able to buy a specialized router that will set aside bandwidth for particular uses or applications. An IT professional can adjust your TCP/IP protocol settings, which will be depend on the particular bandwidth traffic issues in your business. Just like the traffic on the freeway, your bandwidth will have particular traffic patterns but will also change based on your particular network and use.

Keep in mind Quality of Service disappears the moment it enters your modem. Using InSpeed will eliminate echo by prioritizing VoIP data packets over the public Internet. It’s a dynamic solution, because your network traffic is always changing.  Just get InSpeed and be done with it.

 

Identifying Causes of Static in VoIP Calls

Static or buzzing on VoIP calls is typically related to electromagnetic interference. This could be caused by the use of the wrong type of power supply, a nearby device that’s introducing electrical voltage, or wireless devices that broadcast on a similar frequency as your phone. Sometimes nearby cell phones create interference.

Handsets on desk phones can create hum. Switch it with another handset in the office. Guess what? Even if there’s nothing wrong with the handset, the problem will frequently go away. Look out for the placebo effect in solving VoIP static!

Fixing Static on VoIP Calls

  1. For cell phones, move into a different environment in case you are experiencing interference.
  2. Do a test call in another area that has no electronics humming.
  3. Check that your power supply provides the correct voltage and amperage, and eliminate any devices that may be causing interference.

Identifying Causes of Packet Loss

Packet loss is the culprit when portions of the phone call get dropped. The human ear is pretty good at filling in very short gaps, but even small amounts of packet loss will degrade the quality of the phone call.

Poor WAN connections that generate a lot of transmission errors are a common source of packet loss, as well as network congestion and a lack of voice packet prioritization.

Fixing Packet Loss

These days, packet loss is caused by congestion. The network itself rarely drops a packet. If you can limit the congestion for traffic from your own site that helps a lot. SD-WAN products solve this by having a second connection, and will automatically avoid the network with packet loss.

  • The home small office user can buy a specialized router that reserves bandwidth for particular users and particular applications, such as gaming. It’s a bit crude but one can roughly send voice packets out first and throttle both upstream and downstream TCP traffic.
  • An MPLS link will minimize packet loss, but that’s an expensive solution.
  • With InSpeed QoS you can eliminate packet loss over any WAN, with a single connection. InSpeed’s service creates an overlay network on any network connection which then honors packet priorities. The overlay is created between the InSpeed premise device and InSpeed virtual devices in the cloud. Rather than relying on packet markings, InSpeed uses a variety of proprietary techniques to maintain quality: latency, jitter, and loss.
 

The Ultimate Solution to VoIP Call Quality Problems

You can put a Band-Aid over call quality problems by adding bandwidth, or spend a small fortune on an MPLS connection. A better option is to resolve VoIP issues once and for all with InSpeed Quality Service (IQS).
IQS is a software-defined WAN (SD-WAN) solution that was purpose-built to ensure high-quality voice calls over any Internet connection, including commodity broadband links. Its patented technology prioritizes traffic from voice, video conferencing and other interactive applications, eliminating dropped calls, jitter, echo, packet loss and other QoS issues.
IQS doesn’t require any networking expertise — simply plug in the InSpeed appliance and it automatically starts prioritizing voice traffic. Sophisticated software in the InSpeed cloud monitors and manages the traffic flow, creating a self-driving WAN that adjusts to changing conditions.
InSpeed’s technology is so simple and cost-efficient that you can deploy it in small remote locations and even home-based offices. Everybody in the organization enjoys high-quality business communications over any WAN connection every time.


InSpeed Quality Service (IQS)™$99/month

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