The RTCP extended report VoIP metrics block specified by RFC 3611 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. 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 signaling protocol extensions. 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.
A telephone connected to a land line has a direct relationship between a telephone number and a physical location, which is maintained by the telephone company and available to emergency responders via the national emergency response service centers in form of emergency subscriber lists. When an emergency call is received by a center the location is automatically determined from its databases and displayed on the operator console.
Using a separate virtual circuit identifier (VCI) for audio over IP has the potential to reduce latency on shared connections. ATM's potential for latency reduction is greatest on slow links, because worst-case latency decreases with increasing link speed. A full-size (1500 byte) Ethernet frame takes 94 ms to transmit at 128 kbit/s but only 8 ms at 1.5 Mbit/s. If this is the bottleneck link, this latency is probably small enough to ensure good VoIP performance without MTU reductions or multiple ATM VCs. The latest generations of DSL, VDSL and VDSL2, carry Ethernet without intermediate ATM/AAL5 layers, and they generally support IEEE 802.1p priority tagging so that VoIP can be queued ahead of less time-critical traffic.[16]
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.
Because they're working across such a multitude of channels, many of today's phone systems are adopting the moniker of Unified Communications-as-a-Service (UCaaS). These are generally cloud-based, virtual PBXes (private branch exchanges) that include at least one, usually multiple, software clients to enhance their functionality on the web, desktop, and a variety of mobile devices. UCaaS systems have a wide variety of feature sets based on the tried and true VoIP. Even residential VoIP systems come with features that are simply impossible using a conventional telephone system.

SIP networks usually have a proxy server and a SIP gateway. The proxy sever helps lighten the functional requirements of SIP endpoints. It also acts as both client and server, but it adds functionality around call routing and policy-based management. SIP gateways handle the routing and connectivity requirements for connecting SIP calls to other networks. Typically, the advanced features of the VoIP vendors we review here are largely based on the proprietary management technology they build into their proxy servers and gateways. By offering VoIP solutions where these elements of a SIP solution are hosted in the cloud, the providers reviewed here have more flexibility in building advanced features since they have more control over deployment and reliability.
The technical details of many VoIP protocols create challenges in routing VoIP traffic through firewalls and network address translators, used to interconnect to transit networks or the Internet. Private session border controllers are often employed to enable VoIP calls to and from protected networks. Other methods to traverse NAT devices involve assistive protocols such as STUN and Interactive Connectivity Establishment (ICE).
The T.38 protocol is designed to compensate for the differences between traditional packet-less communications over analog lines and packet-based transmissions which are the basis for IP communications. The fax machine may be a standard device connected to an analog telephone adapter (ATA), or it may be a software application or dedicated network device operating via an Ethernet interface.[35] Originally, T.38 was designed to use UDP or TCP transmission methods across an IP network. UDP provides near real-time characteristics due to the "no recovery rule" when a UDP packet is lost or an error occurs during transmission.[36]

A key attraction of VoIP is that it gives these systems the flexibility to work in a wide variety of environments ranging from analog desk phones to softphones piggy-backing on a cell phone. These systems can often also integrate all or part of their softphone clients into other back-office applications, like your customer relationship management (CRM) or help desk platforms. Simply picture the standard interface of such an app that suddenly sports a dial pad and some function buttons as a pop-up screen and you'll have a very basic idea of how some of this works. In addition, these cloud based systems can have a variety of phone numbers in global locations, so that your customers can have free access to your phone at little or no charge.  
The security concerns of VoIP telephone systems are similar to those of other Internet-connected devices. This means that hackers with knowledge of VoIP vulnerabilities can perform denial-of-service attacks, harvest customer data, record conversations, and compromise voicemail messages. Compromised VoIP user account or session credentials may enable an attacker to incur substantial charges from third-party services, such as long-distance or international calling.
In addition to making sure your internet service can handle your VoIP traffic, you also need to make sure your local area network (LAN) can handle it. What makes network management tricky with VoIP is that if you simply drop it onto your network, that traffic will get processed the same as any other traffic, meaning your shared accounting application or those 20 gigabytes worth of files your assistant just stored in the cloud.
Since Verizon is a massive company, customer service ratings are in line with what you’d usually read in the comments section—meaning, people are far more motivated to complain than praise. Navigating Verizon’s bundling plans for businesses may be like a choose-your-own-adventure odyssey, but its support lines are segregated well, with separate contacts for small, medium, and larger businesses. Verizon also offers competitive SLAs (service level agreements) for quality and service and 24/7 support via phone, email, and tickets.
Mass-market VoIP services use existing broadband Internet access, by which subscribers place and receive telephone calls in much the same manner as they would via the public switched telephone network (PSTN). Full-service VoIP phone companies provide inbound and outbound service with direct inbound dialing. Many offer unlimited domestic calling and sometimes international calls for a flat monthly subscription fee. Phone calls between subscribers of the same provider are usually free when flat-fee service is not available.[citation needed]

SIP networks usually have a proxy server and a SIP gateway. The proxy sever helps lighten the functional requirements of SIP endpoints. It also acts as both client and server, but it adds functionality around call routing and policy-based management. SIP gateways handle the routing and connectivity requirements for connecting SIP calls to other networks. Typically, the advanced features of the VoIP vendors we review here are largely based on the proprietary management technology they build into their proxy servers and gateways. By offering VoIP solutions where these elements of a SIP solution are hosted in the cloud, the providers reviewed here have more flexibility in building advanced features since they have more control over deployment and reliability.
In the United Arab Emirates (UAE), it is illegal to provide or use unauthorized VoIP services, to the extent that web sites of unlicensed VoIP providers have been blocked. However, some VoIPs such as Skype were allowed.[56] In January 2018, internet service providers in UAE blocked all VoIP apps, including Skype, but permitting only 2 "government-approved" VoIP apps (C’ME and BOTIM) for a fixed rate of Dh52.50 a month for use on mobile devices, and Dh105 a month to use over a computer connected."[57][58] In opposition, a petition on Change.org garnered over 5000 signatures, in response to which the website was blocked in UAE.[59]

Operators of "Interconnected" VoIP (fully connected to the PSTN) are mandated to provide Enhanced 911 service without special request, provide for customer location updates, clearly disclose any limitations on their E-911 functionality to their consumers, obtain affirmative acknowledgements of these disclosures from all consumers,[65] and 'may not allow their customers to “opt-out” of 911 service.'[66] VoIP operators also receive the benefit of certain US telecommunications regulations, including an entitlement to interconnection and exchange of traffic with incumbent local exchange carriers via wholesale carriers. Providers of "nomadic" VoIP service—those who are unable to determine the location of their users—are exempt from state telecommunications regulation.[67]

VoIP solutions aimed at businesses have evolved into unified communications services that treat all communications—phone calls, faxes, voice mail, e-mail, web conferences, and more—as discrete units that can all be delivered via any means and to any handset, including cellphones. Two kinds of service providers are operating in this space: one set is focused on VoIP for medium to large enterprises, while another is targeting the small-to-medium business (SMB) market.[13]


The only additional piece of equipment that you need is an Analog Telephone Adapter (also referred to as an ATA) that allows you to connect your existing telephone to your home Internet. This equipment is typically provided on a free lease basis from the home VoIP provider that you choose, or you can use you own device if you prefer. You can also use IP phone(s) instead of using the ATA with your existing analog phones. The sound quality is better but there is more up front cost as IP phones are more expensive than the ATA devices.
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