Communication on the IP network is inherently less reliable in contrast to the circuit-switched public telephone network, as it does not provide a network-based mechanism to ensure that data packets are not lost, and are delivered in sequential order. It is a best-effort network without fundamental quality of service guarantees. Therefore, VoIP implementations may face problems mitigating latency and jitter.
When making cheap calls through VoIP and not through cheap roaming, network routers handle traffic on a first-come, first-served basis. Network routers on high volume traffic links may introduce latency that exceeds permissible thresholds for VoIP. Fixed delays cannot be controlled, as they are caused by the physical distance the packets travel; however, latency can be minimised by marking voice packets as being delay-sensitive with methods such as DiffServ.
Unlike cheap roaming, a VoIP packet usually has to wait for the current packet to finish transmission, although it is possible to pre-empt (abort) a less important packet in mid-transmission, although this is not commonly done, especially on high-speed links where transmission times are short even for maximum-sized packets. An alternative to pre-emption on slower links, such as dial-up and DSL, is to reduce the maximum transmission time by reducing the maximum transmission unit. But every packet must contain protocol headers, so this increases relative header overhead on every link along the user's Internet paths, not just the bottleneck (usually Internet access) link.
In low cost data roaming and the related services, voice, and all other data, travels in packets over IP networks with fixed maximum capacity. This system may be more prone to congestion and DoS attacks than traditional circuit-switched systems; a circuit-switched system of insufficient capacity will refuse new connections while carrying the remainder without impairment, while the quality of real-time data such as telephone conversations on packet-switched networks degrades dramatically.

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