The receiver must resequence IP packets that arrive out of order and recover gracefully when packets arrive too late or not at all. Packet delay variation results from changes in queuing delay along a given network path due to competition from other users for the same transmission links. VoIP receivers accommodate this variation by storing incoming packets briefly in a playout buffer, deliberately increasing latency to improve the chance that each packet will be on hand when it is time for the voice engine to play it. The added delay is thus a compromise between excessive latency and excessive dropout, i.e. momentary audio interruptions.
Overall, VoIP is simply the better option for the vast majority of customers. Dropping your landline means no more hidden fees or metered long distance calling charges. Everything is charged at one low rate by most providers and your ability to customize your phone service to exactly what you need is far greater. Unless you've got some highly unique circumstances that somehow mandate a landline, VoIP is simply the better choice.
The following table provides a high level summary of how residential VoIP service compares to other alternative solutions for home phone service. The table compares this service to a regular landline, a bundled phone service from a cable company such as double or triple play, and a cell phone service. The cell phone is included as some people decide to just get rid of their wired phone and use their cell phone for all calls. Free services such as Skype are not included as they are not effective, like for like, landline replacements in our opinion.
Cable companies and Internet providers will also provide a bridge device where your phones stay the same and the VoIPing simply happens on the back-end. Just remember that these devices dictate what kinds of features the provider can offer you, so be sure you know what these devices are capable of since there'll likely be more than one model to choose from.

In 1999, a discrete cosine transform (DCT) audio data compression algorithm called the modified discrete cosine transform (MDCT) was adopted for the Siren codec, used in the G.722.1 wideband audio coding standard.[74][75] The same year, the MDCT was adapted into the LD-MDCT speech coding algorithm, used for the AAC-LD format and intended for significantly improved audio quality in VoIP applications.[76] MDCT has since been widely used in VoIP applications, such as the G.729.1 wideband codec introduced in 2006,[77] Apple's Facetime (using AAC-LD) introduced in 2010,[78] the CELT codec introduced in 2011,[79] the Opus codec introduced in 2012,[80] and WhatsApp's voice calling feature introduced in 2015.[81]


Though many consumer VoIP solutions do not support encryption of the signaling path or the media, securing a VoIP phone is conceptually easier to implement than on traditional telephone circuits. A result of the lack of encryption is that it is relatively easy to eavesdrop on VoIP calls when access to the data network is possible.[38] Free open-source solutions, such as Wireshark, facilitate capturing VoIP conversations.
From an end user point of view, the actual phone service works in the same way, you pick up the phone to answer a call or to dial a number just like with a landline service. Number porting means you can keep your existing phone number and simply switch it over to your new service provider. The residential VoIP providers take care of the call routing and you can call any destination and receive calls from anyone, just like with regular home phone service.
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