Jitter is a more complex problem than latency and packet loss. It's the variation in latency time from one packet to the next. It causes packets to arrive out of order, leaving gaps in the framing sequence of the voice signal. Jitter is at its worst when voice traffic must travel through several routers on the network. The more hops, the worse jitter can get. Different routers, especially those at ISPs, may be configured to queue and forward different kinds of traffic in different ways. Others may be load-balancing, which can contribute to jitter. The chief goal of QoS protocols is to eliminate jitter. Devices called jitter buffers, in endpoints and VoIP servers, can minimize the effect of jitter, too. But, like PLC measures, they do so by increasing latency. (Switching to VoIP By Theodore Wallingford)
In the given example:
ssrc=254186206;themssrc=2024901615;lp=0;rxjitter=0.020917;rxcount=150;txjitter=0.000000;txcount=83;rlp=0;
rtt=14818.715000
How do I interprete the jitter value ? Is the value 0.020917 good ? Bad ? Is there a unit behind this value ?
Answer:
It's a ratio of out-of-order (jittered) to in-order packets, calculated progressively. Due to the progressive calculation, it's not exactly 3/147, in this case, but it's close enough to know that 3 packets were received out-of-order. The closer the value is to 0, the better. -- Tilghman Lesher
To get approximate value of received packets out of order do such calculations: rxcount * rxjitter (in our example 150 * 0.020917 = 3.13755 ~= 3 packets). Same for transmitting end: txcount * txjitter.
