Sending Faxes in Real-Time Over an IP Network

Faxing manually is out of date!

Using Internet technology it is possible to send professional faxes over the Internet at a very low cost. This can be achieved through fax over Internet Protocol (FoIP) which is a technology that allows faxes to be sent in real-time over an IP network.

What is FoIP?

FoIP is a deviation from voice over Internet Protocol (VoIP) as it makes use of a new protocol (T.38) instead of a voice codec. However, both VoIP and FoIP have session management features in common in that both have connection, disconnection and negotiation stages.

In VoIP the data is audio and is sent over an audio compression codec for example G.117a which is a lossy compression scheme: a compression method whereby data that is compressed and then decompressed may well be different from the original, but is "close enough" to be useful in some way. In FoIP the data is T.30 fax data instead of audio and T.30 data does not make use of a lossy compression scheme. Since T.30 data is quite compressed in itself, there is no need for compression but mostly data integrity, and as illustrated in the diagram below, FoIP uses a protocol called T.38 to transfer fax data over IP.

The Fax Session

A fax session is a standardized way of transferring images via a communications medium in real-time. The real- time element is necessary to confirm the transfer of the images. Nowadays, this confirmation in conjunction with Error Correct Mode (ECM) makes a fax legally binding and is used for legal reasons. 

Standardized fax data is called T.30. This data is the same for all types of fax sessions with the difference being that it is encoded in various ways depending on the communications medium used.

Possibility of Fax Over VoIP Audio

It is possible to send a fax of normal audio VoIP but the success rates are very low. This method is referred to as Fax over Voice over IP. VoIP uses a lossy codec that humans hear correctly. This codec eliminates certain parts of the audio that our ear cannot hear for further compression.

To reduce the bandwidth used by the codec for audio data, silence is also not transmitted as part of the audible data. To make things worse, some VoIP codecs produce what is called ‘comfort noise’ so that the human caller hears silence as soothing noise. Fax machines do not like comfort noise since they can ‘hear’ better than humans do and need the audible data intact including the silent parts and also the non-human-audible parts. Therefore, in general Fax over Voice over IP is not a good option and will surely fail miserably.

Calling a Third Party with VoIP

There are many ways to call a third party using VoIP and all these methods are also relevant for FoIP. The various methods are described below.

Direct IP

This is a different methodology from normal phone numbers. As the name ‘Direct IP’ implies, this method requires the exact IP address or URL of the third party to be known despite its impracticality. 

Using a Registrar/Gatekeeper

This method builds on the Direct IP method described above, however, since IP addresses are not very easy to remember, it makes use of regular phone numbers to call a third party using VoIP. This method uses a Registrar (SIP) or a Gatekeeper (H.323), where both Registrar and Gatekeeper perform the same task but called differently depending on the protocol used.

This method works by registering both IP address and phone number with the Registrar or Gatekeeper. When a third party needs to be called, it is then sufficient to dial the phone number of the third party only and the Registrar or Gatekeeper are then queried for the IP address of the specific phone number to call.

The following examples describe the process:

• Phone 10.1.1.1 registers with server having phone number 123
• Phone 10.1.1.2 registers with server having phone number 456
• Phone 10.1.1.1 needs to call phone number 456 so it queries the server for the IP address of the phone number 456.
• The server replies saying that phone number 456 has IP 10.1.1.2
• Phone 10.1.1.1 calls 10.1.1.2 directly.
• The above steps are all done transparently from the user by the phone of VoIP application.

The drawback of this method is that all IP phones must be registered with the same network of Registrars or Gatekeepers to be able to resolve to the phone number.

Gateway mode

So far, both Direct IP and Registrar/Gateway methods assumed that the recipient is on a VoIP network. However, some third parties might only have a normal landline having PSTN or ISDN. There are two ways of contacting third parties on a normal landline, either by using a VoIP provider or by setting up your own gateway network.

VoIP Provider

This method is very common nowadays especially for low cost international calls. Normally the VoIP provider sets up an IP phone (or normal phone connected to an Internet router) in your premises. You are also assigned a phone number. With this equipment you can call a normal landline or IP number directly by dialing only the recipient’s phone number. The VoIP provider will then find the best route to call thethird party with the least cost possible.

This is the simplest and most straightforward form of dialing to a landline or internationally using VoIP. The greatest drawback is that the VoIP provider must also support FoIP to be able to make fax calls using FoIP capable equipment.

Building Your Own Gateway Network

This method is preferred in larger companies, especially those that have an existing IP infrastructure. Building a gateway network involves the use of VoIP Gateways. These devices change transport medium in real-time. For example an IP call is converted and passed through a PSTN line. To better describe this system we need to understand some technical jargon.

Terminal Endpoint (TE) – This is the starting or ending point of an IP call. For example, an IP phone or a VoIP application on your computer. Gateway (GW) – this can act as both a TE and a tunnel to transfer data where a normal TE is not capable of. For example from IP to ISDN.

As illustrated in the above diagram, premises A and B have an IP gateway that translates the PSTN/ISDN calls to IP and vice versa while premises C only have normal PSTN lines. If premises A wish to call premises C, at face value, a caller from premises A simply has to dial the number of premises C. Under the hood, the following would be happening:

• The caller from premises A dials the number at premises C.
• The Gateway at premises A gets the number and depending on its internal dialing rules can either transfer to another internal IP phone connected to it or dial the number out to the telephone company. In the meantime, the caller device is waiting from the gateway.
• A rule in the gateway is triggered to call the telephone company. The number of premises C is called and so the PSTN phone at premises C rings.
• When the phone at premises C is picked up, the gateway at premises A knows this and so establishes an IP call between itself and the caller and between itself and the telephone company, thus translating from PSTN to IP and vice versa.

The same steps above are also used to call someone at premises B. The reverse of what the gateway does at premises A is done at premises B to establish a call between a caller from premises A and a receiver at premises B.

Other examples of gateway devices are PABXs that are IP-enabled thus having an existing Ethernet connection to connect to an existing IP infrastructure.

Least Cost Routing

International companies can benefit when they build their own gateway by considering least cost routing. It is well known that calls from one country to another (example from a US office to the UK office) are more costly than calls within the same country. However, through FOIP it is possible to implement Least Cost Routing (LCR). This results in cost-effectiveness that is achieved through a reduction in international calls dialed since calls are translated into a local call at the recipient’s country.

Notes about this method: some companies have a server room which is distant from the connection room for various reasons. These companies cannot have a server connected directly to the ISDN or PSTN connections but they only have an IP network in the server room. To solve this issue, this company has to install a VoIP gateway in the connection room to translate from ISDN/PSTN to IP and vice versa so that the server room can have access to the ISDN/PSTN connections via VoIP.

SIP and H.323 Session Management

SIP and H.323 are session management protocols, or rather a collection of protocols used throughout the IP session. Both SIP and H.323 are session control protocols for VoIP and only setup the session. Either SIP or H.323 can be used, but only one at a time for example if SIP is used to connect, SIP has to be used to disconnect.

What is H.323?

H.323 is an umbrella recommendation from the ITU-T, which defines the protocols to provide audio-visual communication sessions on any packet network. It is currently implemented by various internet real-time applications such as NetMeeting and GnomeMeeting (the latter using the OpenH323 implementation). It is a part of the H.32x series of protocols which also address communications over ISDN, PSTN or SS7. H.323 is commonly used in Voice over IP (VoIP, Internet Telephony, or IP Telephony) and IP-based videoconferencing.

H.323 was originally created to provide a mechanism for transporting multimedia applications over LANs but it has rapidly evolved to address the growing needs of VoIP networks. The strength of H.323 was the relatively early availability of a set of standards, not only defining the basic call model, but in addition the supplementary services, needed to address business communication expectations. H.323 was the first VoIP standard to adopt the IETF standard RTP to transport audio and video over IP networks.

H.323 is based on the ISDN Q.931 protocol and is suited for interworking scenarios between IP and ISDN, respectively between IP and QSIG. A call model, similar to the ISDN call model, eases the introduction of IP Telephony into existing networks of ISDN based PBX systems. In simple terms, within the context of H.323, an IP based PBX is a Gatekeeper plus supplementary services.

• H.323 references many other ITU-T protocols like:
• H.225.0 protocol is used to describe call signaling, the media (audio and video), the stream, media stream synchronization and control message formats. 
• H.245 control protocol for multimedia communication, describes the messages and procedures used for opening and closing logical channels for audio, video and data, capability exchange, control and indications. 
• H.450 describes the Supplementary Services. 
• H.235 describes security in H.323.
• H.239 describes dual stream use in videoconferencing, usually one for live video, the other for presentation.