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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 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.

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 In the illustrations below.30.Introduction Faxing manually is out of date!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

1 IP phone 10. 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. To make things worse. Therefore. IP phone 10. some VoIP codecs produce what is called ‘comfort noise’ so that the human caller hears silence as soothing noise. VoIP uses a lossy codec that humans hear correctly. it makes use of regular phone numbers to call a third party using VoIP. 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. silence is also not transmitted as part of the audible data. however. This method uses a Registrar (SIP) or a Gatekeeper (H.30 fax data encoded as audio over analog lines. As the name ‘Direct IP’ implies.1.1. Direct IP This is a different methodology from normal phone numbers. Sending faxes in real time over an IP network 4 .323). since IP addresses are not very easy to remember. 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.Dial Connect Negotiate T I M E T.1.1. The various methods are described below. 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 10.1. IP 10 Locally stored IP address database which is difficult to maintain. This codec eliminates certain parts of the audio that our ear cannot hear for further compression. IP DB The Direct IP method Using a Registrar/Gatekeeper This method builds on the Direct IP method described above. Multiple negotiations take place in between pages. This method is referred to as Fax over Voice over IP. this method requires the exact IP address or URL of the third party to be known despite its impracticality. To reduce the bandwidth used by the codec for audio data.

1 2.1.1. The greatest drawback is that the VoIP provider must also support FoIP to be able to make fax calls using FoIP capable equipment. either by using a VoIP provider or by setting up your own gateway network. When a third party needs to be called. Normally the VoIP provider sets up an IP phone (or normal phone connected to an Internet router) in your premises. Register as 3.1.2 #123 . The following examples describe the process: 1.1.2 directly 6.1.2 5.1. The above steps are all done transparently from the user by the phone of VoIP application.1. some third parties might only have a normal land line having PSTN or ISDN. VoIP provider This method is very common nowadays especially for low cost international calls.1. both Direct IP and Registrar/Gateway methods assumed that the recipient is on a VoIP 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. However.1.1. 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 is the simplest and most straightforward form of dialing to a landline or internationally using VoIP. The server replies saying that phone number 456 has IP 10. IP phone 10.1 needs to call phone number 456 so it queries the server for the IP address of the phone number 456 4.This method works by registering both IP address and phone number with the Registrar or Gatekeeper.2 5. The VoIP provider will then find the best route to call the third party with the least cost possible. There are two ways of contacting third parties on a normal landline. You are also assigned a phone number. Phone 10.10. Phone 10.1.2 Registrar/Gatekeeper Stored mappings: #456 .1. Gateway mode So far.1 IP phone 10.1 calls 10.1. Call 10.2 registers with server having phone number 456 3. Phone 10.1 registers with server having phone number 123 2. Phone 10.1.1. 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.1.1. Sending faxes in real time over an IP network 5 .

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

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

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

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

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

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

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