QoS (Quality of Service) is a major issue in VOIP implementations. The issue is how to guarantee that packet traffic for voice or other real-time media will not be delayed or dropped due interference from other lower priority traffic.
Things to consider are:
Latency: Delay for packet delivery
Jitter: Variations in delay of packet delivery
Packet loss: Too much traffic in the network causes the network to drop packets
Burstiness of Loss and Jitter: Loss and Discards (due to jitter) tend to occur in bursts
For an end user, large delays are burdensome and can cause a bad echo degrading the quality of a conversation. It’s hard to have a working conversation with too large delays in packet delivery. You keep interrupting each other. Jitter causes strange sound effects, but can be handled to some degree with “jitter buffers” in the software. Packet loss causes interrupts. Some degree of packet loss won’t be noticeable, but at some point, packet loss can & will negatively impact call quality.
Latency
VoIP Calls usually tolerate roundtrip voice delays of 250ms or more. ITU-T G.114 recommends a maximum of a 150ms (milliseconds) one-way latency. Since this includes the entire voice path, part of which may be on the public Internet, your own network should have transit latencies of considerably less than 150ms.
Jitter
Jitter can be measured in several ways. There are jitter measurement calculations defined in:
IETF RFC 3550 RTP: A Transport Protocol for Real-Time Applications
IETF RFC 3611 RTP Control Protocol Extended Reports (RTCP XR)
Equipment and network vendors often don’t detail exactly how they are calculating the values they report for measured jitter. Most VOIP endpoint devices (e.g. VOIP phones and ATAs) have jitter buffers to compensate for network jitter.
Quoting from Cisco:
Jitter buffers (used to compensate for varying delay) further add to the end-to-end delay, and are usually only effective on delay variations less than 100ms. Jitter must therefore be minimized.
What’s an acceptable level of jitter in a network? 0.5ms – 2ms
Packet Loss
VOIP is not tolerant of packet loss. Even 1% packet loss can “significantly degrade” a VOIP call using a G.711 codec. Other codecs, using compression tolerate even less packet loss without significant degradation of call quality. The default G.729 codec requires packet loss far less than 1% to avoid audible errors. Ideally, there should be no packet loss for VoIP