VoIP QoS Requirements
QoS (Quality of Service) is a major issue in VOIP implementations. The issue is how to guarantee that packet traffic for a voice or other media connection 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 the end user, large delays are burdensome and can cause bad echoes. It’s hard to have a working conversation with too large delays. 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 lots of packet loss will make sound lousy.
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)
But, 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
s) have jitter buffers to compensate for network jitter. 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 100 ms. Jitter must therefore be minimized.