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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? 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 .323 session management What is H.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.

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fax session over analog Public Switched Telephone Network (PSTN) telephone lines.Introduction Faxing manually is out of date!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

however. IP DB The Direct IP method Using a Registrar/Gatekeeper This method builds on the Direct IP method described above.323). The various methods are described below. this method requires the exact IP address or URL of the third party to be known despite its impracticality.1.30 fax data encoded as audio over analog lines. 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. To reduce the bandwidth used by the codec for audio data. Therefore. To make things worse. IP 10 Locally stored IP address database which is difficult to maintain. This method is referred to as Fax over Voice over IP. where both Registrar and Gatekeeper perform the same task but called differently depending on the protocol used. in general Fax over Voice over IP is not a good option and will surely fail miserably.1 IP phone 10. As the name ‘Direct IP’ implies. 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. silence is also not transmitted as part of the audible data. Sending faxes in real time over an IP network 4 . since IP addresses are not very easy to remember. Multiple negotiations take place in between pages. some VoIP codecs produce what is called ‘comfort noise’ so that the human caller hears silence as soothing noise.1. VoIP uses a lossy codec that humans hear correctly.1. it makes use of regular phone numbers to call a third party using VoIP. 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.2 Call 10. Direct IP This is a different methodology from normal phone numbers. IP phone 10.1. This codec eliminates certain parts of the audio that our ear cannot hear for further compression. This method uses a Registrar (SIP) or a Gatekeeper (H.1.Dial Connect Negotiate T I M E T.

1.1. Normally the VoIP provider sets up an IP phone (or normal phone connected to an Internet router) in your premises. Phone 10.1.2 registers with server having phone number 456 3. You are also assigned a phone number.2 Registrar/Gatekeeper Stored mappings: #456 . The greatest drawback is that the VoIP provider must also support FoIP to be able to make fax calls using FoIP capable equipment.1. Register as 3. The VoIP provider will then find the best route to call the third party with the least cost possible.1.1 2.1. There are two ways of contacting third parties on a normal landline.1. Phone 10. both Direct IP and Registrar/Gateway methods assumed that the recipient is on a VoIP network.1.10.1.1 registers with server having phone number 123 2.1.1.1 needs to call phone number 456 so it queries the server for the IP address of the phone number 456 4. Sending faxes in real time over an IP network 5 . When a third party needs to be called.2 directly 6. either by using a VoIP provider or by setting up your own gateway network.1. With this equipment you can call a normal landline or IP number directly by dialing only the recipient’s phone number. Register as 1.1 calls 10. Gateway mode So far. Call 10.1.This method works by registering both IP address and phone number with the Registrar or Gatekeeper. 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. Phone 10. Phone 10. The server replies saying that phone number 456 has IP 10. However.1. VoIP provider This method is very common nowadays especially for low cost international calls.2 5.1. The following examples describe the process: 1. This is the simplest and most straightforward form of dialing to a landline or internationally using VoIP.1.10. some third parties might only have a normal land line having PSTN or ISDN.1.1. IP phone 10.1. The above steps are all done transparently from the user by the phone of VoIP application.1.2 5. 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.2 #123 .1 IP phone 10.1.1.

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

However. Ethernet HUB/ IP Switch Router Sending faxes in real time over an IP network 7 . 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. To solve this issue.Least cost routing International companies can benefit when they build their own gateway by considering least cost routing. 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. 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. through FoIP it is possible to implement least cost routing (LCR).

Either SIP or H. describes the messages and procedures used for opening and closing logical channels for audio.com). the other for presentation. not only defining the basic call model. instant messaging.225) Data Disconnect (H.wikipedia. along with H. capability exchange.32x series of protocols which also address communications over ISDN. within the context of H. for example. Internet telephony.323 is based on the ISDN Q.245 control protocol for multimedia communication.323. and terminating an interactive user session that involves multimedia elements such as video. In simple terms. SIP has to be used to disconnect.323 Connect (H.235 describes security in H. In November 2000.323 H. which defines the protocols to provide audio-visual communication sessions on any packet network. respectively between IP and QSIG. similar to the ISDN call model. an IP based PBX is a gatekeeper plus supplementary services. What is SIP? Session Initiation Protocol (SIP) is a protocol developed by the IETF MMUSIC Working Group and proposed standard for initiating. PSTN or SS7.323 is an umbrella recommendation from the ITU-T. needed to address business communication expectations.323 was the first VoIP standard to adopt the IETF standard RTP to transport audio and video over IP networks.323 are session management protocols. online games.931 protocol and is suited for interworking scenarios between IP and ISDN. Sending faxes in real time over an IP network 8 .323 is commonly used in Voice over IP (VoIP. SIP was accepted as a 3GPP signaling protocol and permanent element of the IMS architecture. For example: SIP T I M E Connect (SIP) Data (RTP) Disconnect (SIP) H.com).323 was the relatively early availability of a set of standards. voice.245/H.239 describes dual stream use in videoconferencing. It is currently implemented by various Internet real-time applications such as NetMeeting and GnomeMeeting (the latter using the OpenH323 implementation). control and indications H.450 describes the Supplementary Services 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. usually one for live video. The strength of H. eases the introduction of IP Telephony into existing networks of ISDN based PBX systems. and virtual reality. modifying. It is one of the leading signaling protocols for Voice over IP.245) What is H. video and data. or IP telephony) and IP-based videoconferencing (www. H. H. A call model. but only one at a time. H. but in addition the supplementary services. if SIP is used to connect.323 (www. It is a part of the H.323 references many other ITU-T protocols like: »» »» »» »» H.323 are session control protocols for VoIP and only setup the session. Both SIP and H.SIP and H.323 session management SIP and H.323? H. or rather a collection of protocols used throughout the IP session. H.323 can be used. H.wikipedia.

feel. iChat AV. Today. and shape of a traditional telephone. Note that the version number remains 2. are commercially available from several vendors. Some of these can use Electronic Numbering (ENUM) to translate existing phone numbers to SIP addresses using DNS. However. As such. although many implementations are still using interim draft versions. However. Microsoft Windows Messenger uses SIP and in June. led by Ubiquity Software and Dynamicsoft have implemented products based on the proposed standards. SIP is a peer-to-peer protocol. Sonus and many more) which can act as proxy and registrar.g. The promoters of SIP have said that the rapid innovation and application development that has characterized the web will now mark the telephony industry. which describes the media content of the session. Sending faxes in real time over an IP network 9 . SIP is characterized by its proponents as having roots in the IP community rather than the telecom industry. Hardware endpoints. SIP proponents also claim it to be simpler than H. The protocol was further clarified in RFC 3261.323 are not limited to voice communication but can mediate any kind of communication session from voice to video or future. causing a phone to ring.323. It requires only a very simple (and thus highly scalable) core network with intelligence distributed to the network edge. hearing ring-back tones or a busy signal. in its current state it has become as complex as H. such as the familiar ‘404 not found’. SIP acts as a carrier for the Session Description Protocol (SDP). this approach is impractical for a public service. features that permit familiar telephone-like operations are present: dialing a number. SIP works in concert with several other protocols and is only involved in the signaling portion of a communication session. building on the Java JAIN specification.323.0.0) was defined in RFC 2543. Implementation and terminology are different. the codec being used etc. which are implemented in the network. SIP is similar to HTTP and shares some of its design principles: It is human readable and request-response structured. though the two protocols themselves could hardly be more different. too. what IP ports to use. Apple Computer announced. embedded in endpoints (terminating devices built in either hardware or software). The first proposed standard version (SIP 2. even though your service provider might normally act as a gateway to the PSTN for traditional phone numbers (and charge you for it). unrealized applications. Many SIP features are implemented in the communicating endpoints as opposed to traditional SS7 features. SIP shares many HTTP status codes. SIP also requires proxy and registrar network elements to work as a practical service. so calls to other SIP users can bypass the telephone network. SIP has been standardized and governed primarily by the IETF while the H.A goal for SIP was to provide a superset of the call processing functions and features present in the Public Switched Telephone Network (PSTN).323 VoIP protocol has been traditionally more associated with the ITU. but that use SIP and RTP for communication. characterized by highly complex central network architecture and dumb endpoints (traditional telephone handsets). a new version of their AOL Instant Messenger compatible client that supports audio and video chat through SIP. software SIP endpoints are common. devices with the look. Although two SIP endpoints can communicate without any intervening SIP infrastructure (which is why the protocol is described as peer-to-peer). There are various softswitch implementations (by Nortel. some would counter that while SIP originally had a goal of simplicity. Other companies. SIP also implements many of the more advanced call processing features present in Signaling System 7 (SS7). In typical use. the two organizations have endorsed both protocols in some fashion. RTP is the carrier for the actual voice or video content itself. e. SIP “sessions” are simply packet streams of the Real-time Transport Protocol (RTP). Although many other VoIP signaling protocols exist. SIP and H. SS7 is a highly centralized protocol. 2003. and released in public beta.

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

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

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