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Analog Telephone Adapter (ATA)


An analog telephone adapter (ATA) is a device that connects an analog telephone, fax machine, or other similar equipment to a computer or network, allowing for internet communication. ATAs allow users to talk on their phones as if they were connected to a traditional telephone network. In this way, they gain the flexibility of the internet while continuing to use their existing analog equipment, and they frequently reduce their overall phone costs.

Analog Telephone Adapter (ATA)

These days, the majority of ATAs can connect directly, without the use of a computer, to an IP network. The ATA is physically attached to a network device, like a modem, router, hub, or switch. It also has direct communication with a voice over IP (VoIP) server, which can be accessed via the internet or the local network. When a phone transmits a signal, the ATA converts it from analog to digital format; conversely, when a signal is received, the ATA converts from digital to analog format. An ATA that connects telephones to an IP network is also known as a VoIP adapter or a VoIP gateway.

In some cases, an ATA connects directly to a computer, typically via a USB port, rather than to the network. While this configuration is not as popular as it once was, when it is, the computer usually runs a softphone program that serves as a bridge between the VoIP server and the ATA. The software performs all other tasks required to enable internet communications in addition to converting data between digital and analog signals.

Analog Telephone Adapter (ATA)

Modern analog telephone adapters

The majority of today's ATAs connect directly to network devices, but they differ significantly. One of the differences is the number of analog devices that can be connected to the ATA. A variety of port types are used to establish connectivity, including the following:

  • Foreign exchange station (FXS). This port allows VoIP communication by physically connecting the ATA to an analog device, like a phone or fax, using an RJ-11 connector. Many FXS ports are frequently present in an ATA.
  • Foreign exchange office (FXO). An RJ-11 connector is used by an FXO port to physically link the ATA to a standard phone line. An FXO port is sometimes included in an ATA, providing alternative access to a public switched telephone network.
  • Ethernet. An RJ-45 connector is used by an Ethernet port to physically link the ATA to an IP network device. There are cases where the ATA has two Ethernet ports: one for connecting to the internet and another for connecting to a computer or router, or other downstream device.

The most basic ATA available today has one Ethernet port, one FXS port, and one jack for a power adapter. However, in a business setting, an ATA frequently includes multiple FXS ports as well as two Ethernet 10/100 megabits per second ports. An ATA is a part of the service package offered by some internet phone service providers, like Vonage.

Analog Telephone Adapter (ATA)

VoIP is the protocol that an ATA uses for communication. Session Initiation Protocol (SIP), a signaling protocol that can start, maintain, alter, and terminate real-time communications, is one of the most widely used protocol. H.323, Inter-Asterisk eXchange, and Media Gateway Control Protocol are other widely used protocols. Codecs are also used by an ATA to convert analog to digital phone signals and vice versa. G.711, G.729, and the Internet Low Bitrate Codec are examples of common codecs.

How Does ATA Work?

Typically, an ATA has two sets of outlets: one for your LAN or VoIP service, and another for your regular phone. Naturally, you can attach an RJ-11 (phone line cable) jack on one side and an RJ-45 (VoIP or Ethernet cable) jack on the other.

An ATA connects via a VoIP Protocol, like SIP or H.323, to the service of the remote VoIP service provider. A voice codec is used for both the encoding and decoding of voice signals. Although you can connect an ATA to a computer or softphone, they don't require software or a computer because they communicate directly with the VoIP service.

Features of an ATA

The most typical characteristics of an ATA are:

  • The capacity to handle VoIP protocols: The better one is, the more protocols one can support. Today, all new ATAs support H.323 and SIP.
  • Ports: In order to create an interface between the VoIP service and the phone network, an ATA needs to have at least one LAN (RJ-45) port and one RJ-11 port. Some ATAs even include additional ports, such as an RJ-45 port for connecting to a computer. This can be used to make phone calls to a PC. A few ATAs have USB ports, making it easier for users to connect them to computers and other devices.
  • Call Switching: VoIP and PSTN are frequently used interchangeably. You can quickly switch between these two thanks to the ATA's call switching features.
  • Standard Service Features. These days, having a number of service features, such as caller ID, call waiting, call transfer, call forwarding, etc., is normal and useful. All these should be supported by a good ATA.
  • Three-way conferencing. You can talk to multiple people at once with many ATAs because they support three-way conferencing. This proves to be extremely helpful, particularly in a business context.
  • Power failure tolerance. The ATA is powered by electricity. Normally, when there is a power outage, it stops functioning. This shouldn't imply that you should be totally unable to communicate. A good ATA should automatically switch to the PSTN line if there is a power failure.
  • Voice quality. Every day, ATA manufacturers sharpen their saws. Some ATAs with improved technologies like Digital Signal Processing (DSP) offer exceptional high-fidelity voice quality.
  • Interoperability. An ATA may be a component of an already intricate hardware setup in a business context. For this reason, a good ATA should be fully compliant and interoperable with other hardware devices. These are only the most common characteristics of a good ATA. Modern ATAs include numerous additional features.

Applications of Analog Telephone Adapter

1. Fax Analyzer

A media gateway appliance called a Fax Analyzer is used to analyze and troubleshoot fax connections and transmissions over IP and PSTN networks. For accessing IP networks and the PSTN, the analyzer offers a minimum of one IP port and one Foreign Exchange Port (FXS) port. The IP port allows access to remote control and monitoring of the appliance from another site, as well as the capture of sessions and Fax over IP data packets. It is possible for the FXS port to record audio during a PSTN fax transmission. T.38 real-time fax functions can be monitored and debugged offline using previously recorded files, or via IP transfers.

A T.38 gateway gathers data sent from an analog fax machine and sends it across the IP network in T.38 packets, which can be transferred via RTP over UDP, UDPTL over UDP, or TPKT over TCP. This process is known as a T.38 fax transfer. For processing, the T.38 packet stream and session control are recorded in a file. The entire T.38 stream can be recreated by utilizing the sequence numbers contained in each packet as well as any redundant or forward error correction (FEC) packets sent. To replicate the fax image, the T.30 dialogue that is part of the T.38 stream is processed.

Images are extracted from a recorded file containing the intercepted PSTN audio signals for fax over PSTN using fax demodulation software. A fax decoder for both conventional fax machines and those that utilize manufacturer-specific T.30 'Non-Standard Facilities' (NSF) features is included in the facsimile monitor. Extensive software libraries are offered by VOCAL to facilitate image extraction and capture for debugging and analysis of various fax standards and transmission protocols.


  • Capture audio signals V.17, V.34, and T.30 (PCM)
  • capture T.38 packets, fax over IP, and session control
  • Analyze T.30 fax messages utilizing the G3 and Super G3 protocols.
  • Analyze the content, timing, and control sequences of fax messages.
  • Compare the timing and content of messages for various Fax over PSTN or Fax over IP connections.
  • Partial fax image recovery and fax image decoding
  • Fax replay via PSTN
  • Playback of G.711 Fax Pass-Through and T.38 protocols via IP messaging
  • Interactive remote debugging and analysis


  • Easy installation with local support technicians.
  • Identify and resolve fax transmission issues quickly.
  • Efficient use of remote engineering expertise.

2. Simultaneous Calling using Single ATA

Voice over IP (VoIP) technologies can be used with multiple analog telephone devices (such as two phones or a phone and fax machine) by using an Analog Telephone Adapter (ATA) with two FXS (RJ-11) ports. Separate profiles are maintained to describe the type and features of the VoIP session that can be supported by each device. Universal Resource Identifiers (URIs) are typically assigned phone numbers to ATAs, but each line can have a different URI (FXS).

The Session Initiation Protocol (SIP) and the Session Description Protocol (SDP) are used to connect the two end points when a call is made or received. Real-Time Transport Protocol (RTP) is used to transfer audio signals between the end points when they are connected.

The same phone number can be used for simultaneous phone calls when an ATA is set up with a single URI. The corresponding ATA-connected device receives the data streams from the various calling sessions. When a call comes in and the first phone is being used, the ATA can ring the second phone if a phone is connected to each FXS port. The ATA can identify the kind of session to create for a received call and ring the phone or fax machine, if necessary, if a phone and fax are connected to the FXS ports. Alternatively, a user can send a fax while talking on the phone.

An ATA with a URI for each line allows for multiple phone calls using different phone numbers. For example, a user may have separate local and long-distance phone numbers, as well as different service providers. In this case, the ATA routes the various data streams to the corresponding ATA-connected device's phone number.


  • Use a single ATA to link two analog phones to VoIP services.
  • Assign a single phone number for all analog devices.
  • Assign unique phone numbers for each analog device.


  • Use the same phone number to place multiple simultaneous calls.
  • Maximize calls from a single phone number to save costs.
  • Use various calling plans and phone numbers to make simultaneous calls.
  • Reduce expenses by utilizing separate internet-based calling plans.

3. Access Multiple Networks

An Analog Telephone Adapter (ATA) media gateway allows analog phone equipment to connect to networks with varying capacities. The media gateway has one FXS (RJ-11) port for connecting analog phone equipment and two Ethernet (RJ-45) ports for connecting to various networks. In order to connect to a destination on either network, convert data packets received from a network, and send audio signals to an analog device, the media gateway software must support the various protocols and/or codecs.

For instance, the RJ-45 ports are linked to a LAN and WAN, but the analog fax is connected to the FXS. For improved compression, dependability, and less bandwidth consumption, the WAN employs G.729A and T.38 Fax over IP with UDPTL, while the LAN uses the G.711 codec and Fax pass-through with RTP. When a call is dialed, the Session Initiation Protocol (SIP) and Session Description Protocol (SDP) create a network connection with the appropriate type and characteristics for the two endpoints. The fax transmission starts using the right protocol and codec for that network once the connection has been made. A wide range of network protocols and codecs are available for sending and receiving data and fax over IP networks thanks to VOCAL's vast software libraries.


  • Connect analog telephones to multiple networks.
  • Support various protocols and codecs.


  • Utilize low-cost internet services for multiple Voice and Fax over IP connections.
  • Choose an alternate path to get the most affordable voice connections.

4. SIP enabled ATA

Voice over IP (VoIP) technologies enable analog phones to place and receive phone calls over the internet with the use of an analog telephone adapter (ATA). When an analog phone is connected to an adapter that supports Session Initiation Protocol (SIP), it appears and functions like a SIP phone to VoIP service providers. The device profile for the unit connected to the adapter identifies the VoIP features it supports and is used in conjunction with the Session Description Protocol (SDP) to configure the type and characteristics of the connection to be established.

SIP is used for session setup, initiation, and termination. A SIP INVITE is sent by the sender to the invitee, who then receives it via a proxy in order to establish a connection between user agents. It specifies the type and characteristics of the proposed connection. The invitee responds with 200 OK through the proxy, indicating which compatible options to use for this session. Voice connections are established through the NAT/firewall using the Simple Traversal of UDP through NAT (STUN) method if the SIP ATA is protected by a router. Real-Time-Protocol (RTP), which easily gets through the NAT/firewall, is used to transmit voice data once the connection has been made.

The call initiator issues a BYE, the invitee responds with a 200 OK, and the session ends to end the connection.


  • Link VoIP services offered over the internet to analog phone equipment.
  • Access VoIP services via NAT/firewall.
  • Use SIP to initiate, set up, and terminate calls.


  • Make use of cheaper VoIP internet services.
  • Make use of VoIP services' enhanced features when using analog phones.

5. Integrated Voice Data Router

With the use of an integrated voice data router, analog telephone equipment and personal computers can connect to IP networks (ATA). Normally, a router has five Ethernet ports: one for connecting via a cable or DSL modem to an external WAN, and four more for attaching IP phones, PCs, and other network-capable devices to a private LAN. Analog phones, fax machines, and other devices can access Voice over IP (VoIP) and Fax over IP (FoIP) services by having a Foreign Exchange Service (FXS) port (RJ-11), which combines voice and data services into a single device.

Media Access Control (MAC) addressing, security services, and basic routing are all offered by the router. It uses a firewall and Network Address Translation (NAT) to defend the private LAN against outside threats. VoIP services use Real Time Protocol (RTP) to transmit voice data and Session Initiation Protocol (SIP) and Simple Traversal of UDP through NAT (STUN) to establish voice connections through the firewall and NAT. VoIP applications rely on real-time packet delivery, which means that both delayed and lost packets can cause distortion in audio output. Data traffic is therefore subordinate to voice or RTP traffic. To achieve voice QoS levels, the router prioritizes RTP packets over other data packets.


  • Connect VoIP services on the internet to analog phone equipment
  • Connect network devices and PCs to a private LAN.
  • support NAT and firewall protection.
  • Facilitate data and voice access to an external WAN.


  • Easy access to internet data and voice services on a single device
  • Lower cost alternative to using separate voice and data connections.

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