What you need to know about latency, jitter and packet loss

As more of our partners start to experiment with SIP trunking and IPCentrex we are often asked what SLAs we offer across our network. The straight answer in relation to jitter, packet loss and latency often leaves the enquirer none the wiser so I thought a layperson's explanation of these terms might be appreciated.

Latency

A measure of the delay in a call. This is calculated by measuring the round-trip delay between when information leaves point A and when a response is returned from point B. The largest contributor to latency is network transmission delay, or congestion. Round-trip latency affects the dynamics of conversation and is the most common contributor to poor voice quality.

With round trip latencies above 300 msec or so, users may experience annoying talk-over effects and call quality can deteriorate at half this delay. Most voice providers will insist on round trip latency guarantees well below this.

Jitter

Jitter refers to how variable latency is in a network. High jitter, greater than approximately 50 msec, can result in both increased latency and packet loss.

When talking to someone it's important that they hear what you say in the same order that you say it. Jitter causes packets to arrive at their destination with different timing and possibly in a different order than they were sent (spoken), with some arriving faster and some slower than they should.

To correct the effects of jitter, VoIP endpoints collect packets in a buffer and put them back together in the proper timing and order before the receiver hears them. Processing the buffer adds delay to the call, so the bigger the buffer, the longer the delay. Also, no matter how big the buffer is, it is finite in size. If voice packets arrive when the buffer is full then packets are dropped and the receiver will never hear them. These are called discarded packets.

Packet Loss

Just as it's important to hear what someone says in the order they say it, it's also important to hear all of what they're saying. If you miss one out of every 10 words or 10 words all at once, chances are you're not going to understand much of the conversation. This is packet loss - some of the voice packets are dropped by network routers or switches that become congested (lost packets), or discarded by the jitter buffer (discarded packets).

Knowing the average packet loss for a call gives you an overall sense for the quality of the call. A call with less than 1 percent average packet loss will always sound better than a call with 10 percent loss. But average loss doesn't tell the whole story. You need to know what type of packet loss you encountered.

There are two kinds of packet loss: "random" and "bursty". Think about two calls each with average 1 percent packet loss. Call A loses one in every 100 packets over the entire call (random loss) while Call B loses 100 packets in two clumps at the beginning and the end of the call (bursty loss). Which call would you rather have? It is important to measure not just the average packet loss but also the type of loss and information on any bursts of packet loss during the call.

Still confused? Give me a call and we can reminisce over the merits of circuit switching and DMS exchanges.