GFI White Paper

Sending faxes in real time over an IP network
How to benefit from Fax over Internet Protocol (FoIP) to send faxes This technical white paper gives an introduction to Fax over Internet Protocol (FoIP) and explains the various usages and advantages of FoIP. FoIP can also be used 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).

323 session management What is H.323? What is SIP? About GFI® 3 3 3 4 4 6 7 7 8 8 8 11 Sending faxes in real time over an IP network 2 .Contents Introduction What is FoIP? The fax session Possibility of Fax over VoIP audio Calling a third party with VoIP Building your own gateway network Least cost routing Notes about this method SIP and H.

the communications medium used is analog PSTN telephone lines.117a.30.30 fax data instead of audio and T.30 fax data is encoded to audible data. and as illustrated in the diagram below.30 data is quite compressed in itself. Standardized fax data is called T. 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.38 standard and sent over to the receiving end. For example: example: G. For negotiation phase. Disconnect Disconnect Format of the data passed from sender to receiver in VoIP and FoIP The fax session A fax session is a standardized way of transferring images via a communications medium in real-time.30 data does not make use of a lossy compression scheme. What is FoIP? FoIP is a deviation from Voice over Internet Protocol (VoIP) as it makes use of a new protocol (T. this confirmation in conjunction with Error Correct Mode (ECM) makes a fax legally binding and is used for legal reasons. Sending faxes in real time over an IP network 3 . This white paper examines the different technologies and protocols available to send faxes.Introduction Faxing manually is out of date! The time when faxes used to be sent one by one over a Public Switched Telephone Network (PSTN) or Integrated Services Digital Network (ISDN) is gone. Since T.38) instead of a voice codec. it is possible to send professional faxes over the Internet at a very low cost.117a.38 to transfer fax data over IP. Using Internet technology. 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. In the illustrations below. both VoIP and FoIP have session management features in common in that both have connection. In FoIP the data is T.30 data thus resulting in an image at the recipient’s side.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. The realtime element is necessary to confirm the transfer of the images. Voice over IP (VoIP) Connect Negotiate T I M E Fax over IP (FoIP) Connect Negotiate The T. The audio data is passed as a as a audio data is passed compressed audio stream using compressed audio stream using audio codec selected audio codec selected in thein the negotiation phase. These can only send audible data in the range of just below 4 KHz which is only good for voice.30 fax data is encoded as per the T. disconnection and negotiation stages. Nowadays. decoded back to T. G. sent to the recipient. but is close enough to be useful in some way. The T. In VoIP the data is audio and is sent over an audio compression codec for example G. However. there is no need for compression but mostly data integrity. The following illustrates a typical fax session over analog Public Switched Telephone Network (PSTN) telephone lines. FoIP uses a protocol called T.

Disconnect Fax session over analog PSTN lines 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. it makes use of regular phone numbers to call a third party using VoIP. As the name ‘Direct IP’ implies. IP 10 Locally stored IP address database which is difficult to maintain. To make things worse. This method uses a Registrar (SIP) or a Gatekeeper (H. silence is also not transmitted as part of the audible data. 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. some VoIP codecs produce what is called ‘comfort noise’ so that the human caller hears silence as soothing noise. IP DB The Direct IP method Using a Registrar/Gatekeeper This method builds on the Direct IP method described above.1.1.1. since IP addresses are not very easy to remember. in general Fax over Voice over IP is not a good option and will surely fail miserably.2 Call 10.323). Direct IP This is a different methodology from normal phone numbers. The various methods are described below.1 IP phone 10.Dial Connect Negotiate T I M E T. however. where both Registrar and Gatekeeper perform the same task but called differently depending on the protocol used. VoIP uses a lossy codec that humans hear correctly. Multiple negotiations take place in between pages. Therefore. this method requires the exact IP address or URL of the third party to be known despite its impracticality. IP phone 10.1. To reduce the bandwidth used by the codec for audio data.30 fax data encoded as audio over analog lines. Sending faxes in real time over an IP network 4 .1. 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. This codec eliminates certain parts of the audio that our ear cannot hear for further compression.

However.2 registers with server having phone number 456 3. Phone 10. There are two ways of contacting third parties on a normal landline. This is the simplest and most straightforward form of dialing to a landline or internationally using VoIP.2 5. Phone 10.1. The server replies saying that phone number 456 has IP 10.1.1. The VoIP provider will then find the best route to call the third party with the least cost possible. Gateway mode So far. You are also assigned a phone number. The above steps are all done transparently from the user by the phone of VoIP application.1. Phone 10.1. The greatest drawback is that the VoIP provider must also support FoIP to be able to make fax calls using FoIP capable equipment.1.10.1. Sending faxes in real time over an IP network 5 . Phone 10.1. Register as 3. either by using a VoIP provider or by setting up your own gateway network. When a third party needs to be called.1. some third parties might only have a normal land line having PSTN or ISDN.1.1. VoIP provider This method is very common nowadays especially for low cost international calls.1 needs to call phone number 456 so it queries the server for the IP address of the phone number 456 4.1 2.1 calls 10.1. The following examples describe the process: IP phone 10.1.2 5. Register as 1.1.1. With this equipment you can call a normal landline or IP number directly by dialing only the recipient’s phone number. Normally the VoIP provider sets up an IP phone (or normal phone connected to an Internet router) in your premises.2 Registrar/Gatekeeper Stored mappings: #456 . Call 10.2 #123 .10. IP phone both Direct IP and Registrar/Gateway methods assumed that the recipient is on a VoIP network.1 registers with server having phone number 123 2. Query IP for #456 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.This method works by registering both IP address and phone number with the Registrar or Gatekeeper. 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.1.2 directly 6.1.

thus translating from PSTN to IP and vice versa. When the phone at premises C is picked up. If premises A wish to call premises C. In the meantime. The caller from premises A dials the number at premises C 2. Under the hood. Gateway (GW) – This can act as both a TE and a tunnel to transfer data where a normal TE is not capable of. To better describe this system we need to understand some technical jargon. 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. an IP call is converted and passed through a PSTN line. an IP phone or a VoIP application on your computer. Building a gateway network involves the use of VoIP gateways. 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. Other examples of gateway devices are PABXs that are IP-enabled thus having an existing ethernet connection to connect to an existing IP infrastructure. at face value. 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. especially those that have an existing IP infrastructure. A rule in the gateway is triggered to call the telephone company. »» »» Terminal endpoint (TE) – This is the starting or ending point of an IP call. The number of premises C is called and so the PSTN phone at premises C rings 4. 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. For example. Sending faxes in real time over an IP network 6 . These devices change transport medium in real-time. a caller from premises A simply has to dial the number of premises C. the following would be happening: 1. The same steps above are also used to call someone at premises B. the caller device is waiting from the gateway 3. For example. For example from IP to ISDN.Building your own gateway network This method is preferred in larger companies. As illustrated in the above diagram.

through FoIP it is possible to implement least cost routing (LCR). To solve this issue. Ethernet HUB/ IP Switch Router Sending faxes in real time over an IP network 7 . Notes about this method Some companies have a server room which is distant from the connection room for various reasons. However. 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.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. 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. 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.

245/H.323 H. an IP based PBX is a gatekeeper plus supplementary services. In simple terms. instant messaging. respectively between IP and QSIG.323.239 describes dual stream use in videoconferencing. similar to the ISDN call model.323 was the first VoIP standard to adopt the IETF standard RTP to transport audio and video over IP networks.323 are session control protocols for VoIP and only setup the session. not only defining the basic call model.323 (www.323 Connect (H. for example.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.225) Data Disconnect (H.323 is an umbrella recommendation from the ITU-T. H. H. H.323 references many other ITU-T protocols like: »» »» »» »» H. It is currently implemented by various Internet real-time applications such as NetMeeting and GnomeMeeting (the latter using the OpenH323 implementation). Internet telephony. online games. A call model. within the context of H. or IP telephony) and IP-based videoconferencing (www. The strength of H. voice. and terminating an interactive user session that involves multimedia elements such as needed to address business communication expectations. video and data. the other for presentation.SIP and H. but in addition the supplementary services.245 control protocol for multimedia communication. SIP was accepted as a 3GPP signaling protocol and permanent element of the IMS architecture.323 was the relatively early availability of a set of standards.450 describes the Supplementary Services H. eases the introduction of IP Telephony into existing networks of ISDN based PBX systems. along with H. if SIP is used to connect.245) What is H. PSTN or SS7. For example: SIP T I M E Connect (SIP) Data (RTP) Disconnect (SIP) H. which defines the protocols to provide audio-visual communication sessions on any packet network.323 can be used. Either SIP or H. and virtual reality. describes the messages and procedures used for opening and closing logical channels for audio.235 describes security in H.wikipedia. In November 2000. capability exchange. but only one at a time.wikipedia. or rather a collection of protocols used throughout the IP session. It is a part of the series of protocols which also address communications over ISDN. Both SIP and H. modifying. control and indications H.323 is commonly used in Voice over IP (VoIP. H.931 protocol and is suited for interworking scenarios between IP and ISDN.323 session management SIP and H. SIP has to be used to disconnect. H. usually one for live video. It is one of the leading signaling protocols for Voice over IP.323? H. Sending faxes in real time over an IP network 8 . What is SIP? Session Initiation Protocol (SIP) is a protocol developed by the IETF MMUSIC Working Group and proposed standard for initiating.323 are session management protocols.323 is based on the ISDN Q.

SIP and H. which describes the media content of the session. However. but that use SIP and RTP for communication. characterized by highly complex central network architecture and dumb endpoints (traditional telephone handsets). Microsoft Windows Messenger uses SIP and in June.323. SIP is characterized by its proponents as having roots in the IP community rather than the telecom industry. are commercially available from several vendors. SIP also requires proxy and registrar network elements to work as a practical service. The promoters of SIP have said that the rapid innovation and application development that has characterized the web will now mark the telephony industry. software SIP endpoints are common. led by Ubiquity Software and Dynamicsoft have implemented products based on the proposed standards. Implementation and terminology are different. In typical use. unrealized applications. Note that the version number remains 2. the two organizations have endorsed both protocols in some fashion. this approach is impractical for a public service. Although many other VoIP signaling protocols exist. However. devices with the look. Some of these can use Electronic Numbering (ENUM) to translate existing phone numbers to SIP addresses using DNS. though the two protocols themselves could hardly be more different. Although two SIP endpoints can communicate without any intervening SIP infrastructure (which is why the protocol is described as peer-to-peer). Sonus and many more) which can act as proxy and registrar. SIP also implements many of the more advanced call processing features present in Signaling System 7 (SS7). There are various softswitch implementations (by Nortel. a new version of their AOL Instant Messenger compatible client that supports audio and video chat through SIP.323 VoIP protocol has been traditionally more associated with the ITU. features that permit familiar telephone-like operations are present: dialing a number. Apple Computer announced. 2003. SIP is similar to HTTP and shares some of its design principles: It is human readable and request-response structured. too. RTP is the carrier for the actual voice or video content itself.323. Other companies. SIP has been standardized and governed primarily by the IETF while the H. hearing ring-back tones or a busy signal. what IP ports to use. e.0. SIP shares many HTTP status codes. the codec being used etc. SIP “sessions” are simply packet streams of the Real-time Transport Protocol (RTP). SIP is a peer-to-peer protocol. Many SIP features are implemented in the communicating endpoints as opposed to traditional SS7 features. It requires only a very simple (and thus highly scalable) core network with intelligence distributed to the network edge.0) was defined in RFC 2543. SIP proponents also claim it to be simpler than H. even though your service provider might normally act as a gateway to the PSTN for traditional phone numbers (and charge you for it). some would counter that while SIP originally had a goal of simplicity. SS7 is a highly centralized protocol. and shape of a traditional telephone. SIP works in concert with several other protocols and is only involved in the signaling portion of a communication session. such as the familiar ‘404 not found’. causing a phone to ring. SIP acts as a carrier for the Session Description Protocol (SDP).g. As such. The first proposed standard version (SIP 2. The protocol was further clarified in RFC 3261. iChat AV.A goal for SIP was to provide a superset of the call processing functions and features present in the Public Switched Telephone Network (PSTN). building on the Java JAIN specification. although many implementations are still using interim draft versions. Hardware endpoints. embedded in endpoints (terminating devices built in either hardware or software). so calls to other SIP users can bypass the telephone network.323 are not limited to voice communication but can mediate any kind of communication session from voice to video or future. Sending faxes in real time over an IP network 9 . feel. in its current state it has become as complex as H. and released in public beta. which are implemented in the network. Today.

Standards being developed by such organizations as 3GPP and 3GPP2 define applications of the basic SIP model which facilitate commercialization and enable support for network-centric capabilities such as CALEA. called SIMPLE. Most SIP clients can use STUN to traverse full cone. who has implemented SIP. Some efforts have been made to integrate SIP-based VoIP with the XMPP presence specification used by Jabber. Most notably Google Talk. For example. the network can not easily support legal interception of calls (referred to in the United States by the law governing wiretaps. Emergency calls (calls to E911 in the USA) are difficult to route. Commercial application The Real-time Transport Protocol (RTP) used to carry the media stream does not traverse NAT routers. enable CALEA and traversal of older. Gizmo Project. has been proposed and is under development. SIP-unaware NAT devices. restricted cone. SIP’s peer-to-peer nature does not enable network-provided services. and plans to integrate SIP. SIMPLE can also carry presence information. Presence information is most recognizable today as buddy status in IM clients such as MSN Messenger and AIM. CALEA).Instant messaging (IM) and presence A standard instant messaging protocol based on SIP. conveying a person’s willingness and ability to engage in communications. because of the inherent mobility of IP end points and the lack of any network location capability. has integrated XMPP in their client and service. and port restricted cone NAT but not symmetrical NAT. Sending faxes in real time over an IP network 10 . practical solutions to these problems are being proven. RTP Proxies. as commercial SIP services begin to take off. Also some newer routers now recognize and pass SIP traffic. special purpose SIP line speed processors analogous to HTTP proxies commonly used in the early 1990s. As envisioned by its originators. It is difficult to identify the proper Public Service Answering Point. who has extended XMPP to integrated voice. (PSAP). However.

which together support hundreds of thousands of installations worldwide. About GFI GFI Software provides web and mail security. More information about GFI can be found at http://www. SIPfoundry has made available and actively develops a variety of SIP stacks. client applications and SDKs. With award-winning technology. or The new market for consumer SIP devices continues to expand. allow customers to bring their own SIP devices. The company has offices in the United States (North Carolina. Sending faxes in real time over an IP network 11 . and a strong focus on the unique requirements of SMEs. Hong Kong. Philippines and Romania. in the cloud or as a hybrid of both delivery models. GFI products are available either as on-premise solutions.Companies such as Vonage and SIP phone were consumer SIP pioneers and have a fast growing subscriber base. such as BroadVoice.gfi. Some VoIP phone companies. The open source community started to provide more and more of the SIP technology required to build both end points as well as proxy and registrar servers leading to a commoditization of the technology. the Asterisk PBX. Major carriers like AT&T and Level (3) are now following suit. California and Florida). including SIP-capable telephone sets. backup and fax. The traditional telecommunications industry (including companies such as Lucent Technologies and Nortel) is now focused on developing systems based on the architecture model and SIP extensions as defined by 3GPP in their IP Multimedia Subsystem (IMS). a competitive pricing strategy. Australia. Malta. archiving. networking and security software and hosted IT solutions for small to medium-sized enterprises (SMEs) via an extensive global partner community. GFI is a channel-focused company with thousands of partners throughout the world and is also a Microsoft Gold Certified Partner. GFI satisfies the IT needs of organizations on a global scale. Austria. UK (London and Dundee). which accelerates global adoption. in addition to entire IP PBX solutions that compete in the market against mostly proprietary IP PBX implementations from established vendors.

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