QoS

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 echos. 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.

VOIP QoS Requirements

Latency

Callers usually notice roundtrip voice delays of 250ms or more. ITU-T G.114 recommends a maximum of a 150 ms 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 150 ms.

Most network SLAs specify maxium latency

• Axiowave SLA 65ms maximum latency
• Internap SLA 45ms maximum latency
• Qwest SLA 50ms maximum latency - Measured Actual for Oct 2004: 40.86ms
• Verio SLA 55ms maximum latency

The SLA numbers above are for backbone providers, the total latency for a VOIP call may also include additional latency in the VOIP provider's and the user's local ISP networks.

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)

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 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 100 ms. Jitter must therefore be minimized.

Whats an acceptable level of jitter in a network? Several network providers now speciify maximum jitter in their SLAs.

• Axiowave SLA 0.5ms maximum jitter
• Internap SLA 0.5ms maximum jitter
• Qwest SLA 2ms maximum jitter - Measured Actual for Oct 2004: 0.10ms
• Verio SLA 0.5ms average, not to exceed 10ms maximum jitter more than 0.1% of time
• Viterla SLA 1ms maximum jitter

The SLA numbers above are for backbone providers, the total jitter for a VOIP call may also include additional jitter in the VOIP provider's and the user's local ISP networks.

Detailed jitter reading

• More detailed overview