As IP packets traverse networks and the biggest network of them all the internet on a hop-by-hop basis not all packets follow the same route. This occurs because the internet routers’ links are not stable and so they are constantly reviewing and updating their routing information. This can result in packets being dropped or lost when a link flaps or changes state. In order to manage this packet loss TCP/IP – the transport protocol for IP – has to have a mechanism for the recipient to not only re-order packets that arrived out of sequence but to detect packets that never arrive at all. To achieve this the recipient acknowledges packets that arrive safely to the sender. Consequently, if no acknowledgement arrives the sender assumes the packet lost and resends it.
Packet loss management is a great feature on data networks. However, it is of little use on real-time streaming voice or a video application where packet resending and packet reordering is pointless.
Packet loss though is still very important, as it is one of the key performance indicators for judging the quality of a network. Packet loss on a LAN data network will generally be very low unless the network is experiencing traffic congestion. On the internet however, it will be much higher and will determine the call quality. VoIP in the early days suffered from unpredictable and unstable call quality due predominantly to packet loss, high delays and unpredictable difference between packet delays leading to packets being declared lost.
Packet loss in VoIP over the internet is unavoidable; however, steps can be taken to mitigate quality issues. These include applying QoS to routers under your control and provisioning sufficient dedicated bandwidth to VoIP internet links. Providing VoIP priority and sufficient bandwidth will ensure at least on your own network that packet loss is minimized.