Did you know that you can use regular telephones and even FAX machines with VoIP telephone services? Even an enterprise...
VoIP system can have a mix of VoIP and analog phones. The magic is in the interfacing.
Anytime you want to connect two different networks together you need a gateway. This is also true when connecting or interfacing a digital network line carrying VoIP phone traffic with the traditional telephone network. In this case the interface device may be called a media gateway, VoIP gateway, FXS adaptor, ATA analog telephone adaptor or broadband phone adaptor.
How they work
Analog Telephone Adaptors, or ATAs, do pretty much what you would expect. They have a standard RJ-11 socket where you plug in the wire from your regular telephone. It's just like plugging that phone into a wall socket. The ATA mimics the connection to the telephone company so the phone doesn't know the difference.
The socket where you plug in the phone on the ATA is known as an FXS port. FXS stands for foreign exchange station. This is PBX system lingo. You can get FXS adaptors for routers. These are modules that plug into the back of the router and give you one, two or more FXS ports. This is how an enterprise level system can add ports to plug in regular phones. The FXS adaptors are programmed so the VoIP system sees them as phone extensions.
By the way, the opposite of an FXS port is an FXO port. FXO stands for foreign exchange office and connects to the phone lines that run to a telephone company office. Never get these port types mixed up. Sparks could fly!
An ATA is also called a broadband phone adaptor because it converts regular phone signals to and from broadband signals. On the back of your ATA, you'll find one or two of the FXS ports we talked about plus a standard RJ-45 network jack. This jack could be labeled "LAN," "WAN connection," "Broadband" or maybe "Ethernet." It's your digital network connection. A standard Ethernet cable plugs into the RJ-45 port and goes to a Router, DSL modem or Cable modem. An Ethernet switch might be built into the ATA to provide two RJ-45 jacks. One connects to the broadband line, the other goes to a computer that shares the broadband connection with the telephone. The only other connection is probably a jack for the AC power converter.
Inside the ATA
The ATA looks simple on the outside, but what is going on inside? Understand that VoIP and analog telephones are completely different species of phone service. Analog phones run on varying voltages and currents. VoIP is based on digital bits. To get from bits to volts, a special chip called a CODEC is used. CODEC stands for Coder/Decoder. Coding converts analog voice from the microphone to digital. Decoding converts digital back into analog to drive the earpiece.
The FXS port circuitry also has to mimic the high voltage AC ringing signal to the analog phone so that it will ring when called. The FXS port provides the same 45 volt DC power to the phone that it normally gets from the telephone company. Note that most of the ringing circuits have enough power to drive 3 to 5 phones. Yes, you can connect multiple analog phones to the same FSX port just like you would hook up multiple extensions to a single phone line. The REN or ringer equivalence number for each FXS port will tell you how many.
Dialing analog phones is done through a keypad that generates somewhat musical tones sometimes called "touchtones". The technical name for this is DTMF or dual tone multi-frequency. Each of the 10 standard keys 0 through 9, plus the # and * keys, generate a unique tone made up of two frequencies each. A DTMF detector senses these and converts them to digital bits so the VoIP system knows what phone number you are dialing. Most VoIP systems use H.323 or SIP standards for control and signaling, although in enterprise wide systems you may find other standards. Make sure that the adaptor you select will work with the VoIP protocols you have implemented.
Analog telephone adaptors have some other functions unique to VoIP telephony. VAD or voice activity detection senses when someone is talking or when the line is quiet. Standard analog telephony always has a complete circuit tied up for each call. VAD lets a VoIP system only send packets when someone is talking. That frees up bandwidth for other uses when both parties are on the line but not speaking at the moment.
Comfort noise and other features
Comfort noise generation or CNG goes along with VAD. If nobody is talking, the VAD circuit will stop sending packets through the broadband connection. Your phone will go dead silent for an instant and you might think it was disconnected. The comfort noise generator creates a low hissing sound, like phone line noise, in the background so you have the "comfort" of knowing that your phone call is still connected.
Because VoIP calls are transmitted over networks that can get congested, a dynamic jitter buffer is often included to smooth out the voice signal. Excessive jitter is caused by packets being sent by alternate routes or getting delayed by other traffic. The jitter buffer helps to feed voice packets to the CODEC at a more constant rate. That makes for a clearer sounding voice signal.
Another source of voice distortion is echo. The regular PSTN or public switched telephone network (the phone company) provides circuits to cancel echoes on long distance phone lines. VoIP has to provide its own mechanism to deal with annoying background echoes, so echo canceling circuitry is included in the analog to VoIP phone adaptors.
FAX tones present a dilemma to many VoIP systems. Some CODECs save bandwidth by compressing the voice signals so that many more phones or computers can share the same network line. This compression can completely distort FAX machine tones. Some ATAs accommodate this by detecting FAX tones and then switching to the most compatible CODEC, usually the G.711 standard, and shutting off VAD and CNG for the duration of the FAX call. FAX can work over VoIP, but only if the ATA and service provider support this.
Caller ID is another system that needs special accommodation. The caller ID is transmitted over regular phone lines using a system called FSK or frequency shift keying. The FSK tones with the caller ID info get sent during the ringing cycle before you pick up the phone. An ATA needs to generate these same FSK tones based on the caller ID data supplied by the VoIP system if you want to have this feature available.
Other functions of an Analog Telephone Adaptor relate to the network side of the connection. The ATA is a network peripheral like a PC or printer. Most are set up to get their network address assigned automatically by the router they are plugged into. This is called DHCP for dynamic host configuration protocol. If no router is being used or if the network uses all static IP addresses, then the ATA needs to provide a way to reprogram its address manually. Some ATAs do this by entering data on the telephone keyboard. More sophisticated devices include a web based interface for programming or downloading new software via a PC.
Nothing simple about it, is there? Fortunately, the functions needed for analog telephones and VoIP are pretty well standardized and designed into specialized circuits and chips. A VoIP phone, also called a broadband phone, or SIP phone when it uses the Session Initiation Protocol standard for VoIP, contains many of the same functions, including CODECs, jitter buffers, voice activity detection, and comfort noise generation.
If you'd like to learn more about analog telephone adaptors, the Cisco ATA 186 is a good example. Broadvox and Packet8 are VoIP services that include an analog telephone adaptor as part of their residential and small business VoIP service plans.