What is VoIP?
VoIP is short for “Voice over IP”.
When we are talking to each other using a communication platform (e.g., Alcatel-Lucent Rainbow, MS Teams, Zoom …), the audio is sent over the IP network. Of course, the same goes for dedicated IP phones.
VoIP has two key characteristics:
- Packet-Switched (vs. Circuit-Switched)
- UDP (vs. TCP)
Packet-Switched (vs. Circuit-Switched)
The primary hardware for a circuit-switched network is the private branch exchange (PBX) system. Our voice is transformed into electronic signals, that pass through several switches before a connection is established. And during a call, no other network traffic can use those switches. So there is a dedicated connection (and bandwidth) between sender and receiver.
In packet-based networks, however, our voice is broken into small data packets that seek out the most efficient route as circuits become available. Each packet may go a different route (and arrive at a different time interval). Furthermore, the packets are sent along with any other IP traffic, there is no dedicated connection (nor bandwidth) between sender and receiver.
UDP (vs. TCP)
In 99% of the cases, VoIP packets are sent using the UDP transport protocol. UDP is much faster than TCP, because it lacks the ‘acknowledgement’ mechanism. Continuous packet stream. Since TCP connection always acknowledges a set of packets (whether the connection is totally reliable), a retransmission must occur for every negative acknowledgement where a data packet has been lost.
But, on the other hand, UDP doesn’t support retransmission of packets, because the sender will not know IF a packet has arrived at its destination. Nor will the sender know WHEN a packet has arrived at its destination.
Common VoIP quality issues
Packet loss / Jitter
Symptoms: choppy audio or even calls being dropped.
Possible causes: an unstable network connection or lack of bandwidth.
As already mentioned above, every IP packet may go on a different route to reach its destination. And it is possible that some packets fail to reach their destination, meaning that the audio represented by this packet cannot be heard by the receiver.
Jitter is the delay that arises when the packets that are sent arrive in different order at its destination. If there’s only a tiny amount of jitter, a jitter buffer can fix the issue. This is where media packets are shuffled into the correct order to reach their endpoint at the right time.
If a fix doesn’t occur in a reasonable timeframe, jitter-induced packet loss occurs, resulting in choppy audio.
Symptoms: echo, delayed audio.
Possible causes: malfunctioning headsets, unstable network connection or lack of bandwidth.
Latency is the time it takes for audio to move from a phone or computer to your headphones. The amount of lag varies depending on the device itself, the network, and even your headset.
The cause of VoIP quality problems can often, but not always, be found in the underlying network connection. Sometimes the peripherals (eg: headset) can also cause problems, or even the VoIP application or the operating system itself. It is always useful to have these problems thoroughly investigated by your own IT service or the IT service provider.
A more detailed article on VoIP issues and troubleshooting can be found here: