Voice over Internet Protocol (VoIP) is a general term for a family of transmission technologies for delivery of voice communications over IP networks such as the Internet or other packet-switched networks. Other terms frequently encountered and synonymous with VoIP are IP telephony, Internet telephony, voice over broadband (VoBB), broadband telephony, and broadband phone.
Internet telephony refers to communications services — voice, facsimile, and/or voice-messaging applications — that are transported via the Internet, rather than the public switched telephone network (PSTN). The basic steps involved in originating an Internet telephone call are conversion of the analogue voice signal to digital format and compression/translation of the signal into Internet protocol (IP) packets for transmission over the Internet; the process is reversed at the receiving end.
VoIP systems employ session control protocols to control the set-up and tear-down of calls as well as audio codec’s which encode speech allowing transmission over an IP network as digital audio via an audio stream. Codec use is varied between different implementations of VoIP (and often a range of codec’s are used); some implementations rely on narrowband and compressed speech, while others support high fidelity stereo codec’s.
Challenges
Quality of service (QoS)
Because the underlying IP network is inherently less reliable, in contrast to the circuit-switched public telephone network, and does not provide a mechanism to ensure that data packets are delivered in sequential order, or provide quality-of-service (QoS) guarantees, VoIP implementations may face problems mitigating latency and jitter.
Voice, and all other data, travel in packets over IP networks with fixed maximum capacity. This system is more prone to congestion and DoS attacks than traditional 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.
Fixed delays cannot be controlled (as they are caused by the physical distance the packets travel), however some delays can be minimized by marking voice packets as being delay-sensitive (see, for example, DiffServ). Fixed delays are especially problematic when satellite circuits are involved, because of long round-trip propagation delay (400–600 milliseconds for links through geostationary satellites).
A cause of packet loss and delay is congestion, which can be avoided by means of teletraffic engineering.
The receiving node must restructure IP packets that may be out of order, delayed or missing, while ensuring that the audio stream maintains a proper time consistency. Variation in delay is called jitter. The effects of jitter can be mitigated by storing voice packets in a jitter buffer upon arrival and before producing analogue audio, although this further increases delay. This avoids a condition known as buffer under-run, in which the voice engine is missing audio since the next voice packet has not yet arrived. When IP packets are lost or delayed at any point in the network between VoIP users there will be a momentary dropout of voice if all packet delay and loss mechanisms cannot compensate.
It has been suggested to rely on the packetized nature of media in VoIP communications and transmit the stream of packets from the source phone to the destination phone simultaneously across different routes (multi-path routing). In such a way, temporary failures have less impact on the communication quality.
In capillary routing it has been suggested to use at the packet level Fountain codes or particularly raptor codes for transmitting extra redundant packets making the communication more reliable.
A number of protocols have been defined to support the reporting of QoS/QoE for VoIP calls. These include RTCP Extended Report(RFC 3611), SIP RTCP Summary Reports, H.460.9 Annex B (forH.323), H.248.30 and MGCP extensions. The RFC 3611 VoIP Metrics block is generated by an IP phone or gateway during a live call and contains information on packet loss rate, packet discard rate (because of jitter), packet loss/discard burst metrics (burst length/density, gap length/density), network delay, end system delay, signal / noise / echo level, Mean Opinion Scores (MOS) and R factors and configuration information related to the jitter buffer.
RFC 3611 VoIP metrics reports are exchanged between IP endpoints on an occasional basis during a call, and an end of call message sent via SIP RTCP Summary Report or one of the other signalling protocol extensions. RFC 3611 VoIP metrics reports are intended to support real time feedback related to QoS problems, the exchange of information between the endpoints for improved call quality calculation and a variety of other applications.
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