Voice over IP

IP telephony (short for Internet Protocol telephony) and Internet telephony or Voice over IP called ( VoIP), is making phone calls over computer networks, which are constructed according to Internet standards. This typical information that is language and control information is transmitted, for example, to establish the connection via a usable also for data transmission network for telephony. The callers both computer specializing in IP telephony telephone terminals, as connected via special adapters classic telephones to establish the connection.

IP telephony is a technology that makes it possible to realize the telephone service on IP infrastructure, so that it can replace the conventional phone technology including ISDN, network and all components. Objective here is to reduce costs by a uniformly structured and to be operated network. Due to the long duration of use traditional telephony systems and the necessary new investments for IP telephony in exchange for existing providers is often realized as a long -lasting, smooth transition. Meanwhile, there are two technologies in parallel ( smooth migration ). This gives a clear need for solutions results to relate the two telephony systems (such as VoIP gateways ) and the need for specific planning of the system change taking into account the possibilities for cost and performance optimization. New providers are increasingly active exclusively with new technology (ie, IP telephony instead of conventional telephone) on the market. End of 2010, approximately 7.7 million people were using exclusively the voice-over- IP technology in Germany.

3.1 signaling protocols
3.2 Connecting with SIP
3.3 number systems 3.3.1 SIP address
3.3.2 Telephone number
3.3.3 ENUM
3.3.4 Conventional local phone numbers via a gateway
3.3.5 Special Internet numbers

3.4.1 Digitization of analog signals and digital processing
3.4.2 Transport of data
3.4.3 transmission quality

4.1 throughput
4.2 Running time (latency ) and jitter
4.3 Packet Loss
4.4 Availability

5.1 terminal 5.1.1 terminal types
5.1.2 Fax over IP Fax over IP ( FoIP )

6.1 Direct Internet telephony
6.2 Internal organization telephony
6.3 Background technique of traditional telephony

8.1 resiliency 8.1.1 Power Supply

Mediation of VoIP calls - network service

The placement of telephone calls is an essential task in computer networks. However, since many users are dynamically connected to the Internet, so that changes the IP address frequently, separates the IP address itself directly as a " phone number " for contacting the VoIP phones. A mediation service in the form of a server takes over this task and allows the telephony even with changing IP addresses of the IP phones.

VoIP phones log on to the server, so the server knows the current IP address of the phones.
Using the IP address of the phone that the server (for example, SIP server ) was made known, he can take over the mediation and the IP phone will ring as a function of this IP address (ie, at any location in the world, when the IP phone has registered from there the mediation Server via the Internet).
Communication between the IP phones can then be performed independently of the server.
There are commercial services that offer the same time with an account for the Mediation Server and a local telephone, which is also accessible via the fixed network. The IP phone calls are usually free.
If one has a fixed IP address, it is possible to set up on the machine where a mediation server (for example OpenSIPS ), then comparable to connect to the connection of several local networks in the fixed network more switching servers together. In commercial solutions often exist partner networks that establish a connection between free VoIP partner networks. The network selection, however, is generally limited, as companies have to cover with the compounds of VoIP phones in the regular landline their sales. Free, self- patch open source telephony server can technically independent of these economic limitations on the Internet, a network of exchanges form. Although SIP telephony server function technically good, currently has an institutionalized network of such SIP switch servers but not yet.

History

In addition to the telephone network gradually developed further communications infrastructure, often on the lines of telephone networks. Starting with the networking of computer systems in the 1980s, which was followed by the Internet boom of the 1990s, rose and the transmission power continuously increases sharply: Were initially with acoustic couplers 300 bits per second achieved were in January 2008, up to 100,000 000 bits per second for end users with broadband connections to the normal telephone service connections or the cable network feasible. This infrastructure provides a basis for IP-based data networks, particularly the Internet, a public network.

Development

In 1973, the first digital voice transfers were realized in the ARPANET by Network Voice Protocol between PDP-11 computers. The voice channel data transfer rate of 3490 bit / s has to be provided. Only four years later, the Network Voice Protocol described became the standard RFC 741, even before 1980, the Internet Protocol ( IP) is specified in RFC 791. Also in 1980 were (then CCITT) documented for ISDN first recommendations of the ITU -T, which was commercially introduced in 1989 and allows calls with higher voice quality and additionally integrates various services such as caller ID on a network. The default data transfer rate of ISDN grew from 3,490 bit / s with NVP - II on 64 kbit / s In the same year the development of the World Wide Web, which should later prove to be the basis for the widespread success of the Internet began.

With the GSM mobile, a service for mobile voice transmission at a bit rate of exactly 13 kbit / s was created (260 bit frames at a frame duration of 20 ms ) from 1992 in Germany ( D-net ). However, these 13 kbit / s refer only to the baud rate of the payload. In order to transfer the user data against transmission errors, eg by atmospheric conditions, to protect, is added to the signal by the channel coding redundancy. This leaves the data frame of 260 bit to 456 bit increase, however, during the frame duration must remain constant ( real-time requirement ). Thus, the actual bit rate of the transmission (user data redundancy for error correction ) is 22.8 kbit / s

In 1994, Michaela Merz developed by the Free Software Association of Germany mtalk, a free Voice -over - IP software for Linux. The first versions of mtalk had only rudimentary data compression. mtalk formed the basis for a whole range of VoIP software, various packages are kept for historical reasons, even from different servers for retrieval.

In the year 1995 provided a MS Windows program the Israeli company VocalTec Communications Internet telephony, but only in half duplex mode, so the caller could only speak alternately with poor voice quality. Connections to computers that are not used the same software, were not provided and therefore impossible. Just a year later allowed QuickTime Conferencing audio and video communication in full duplex mode over AppleTalk and IP networks on the one hand, on the other hand, the Real-Time Transport Protocol in RFC 1889 has been described.

Three years later, in 1998, was first adopted by a H.323 ITU-T standard frame, so that solutions from different vendors should be compatible with each other. The Session Initiation Protocol ( SIP) in RFC 2543 is specified in the following year. The following construction of the VoIP solutions brought in 2001 in Austria the first communication to the operation of a carrier voice switching network by the regulatory authority to the IPAustria forth. For the purposes of today's VoIP followed in 2002 for the improvement of the VoIP SIP extension in RFC 3261, as well as better access to other networks, the adoption of ITU Q.1912.5 interoperability between SIP and ISDN User Part

The standardization of VoIP contrary to the Skype software was released in 2004, the private, not open protocol for IP telephony, based on the peer-to -peer technology used.

Principle of operation

Making calls with IP telephony can be displayed for the participants as well as in traditional telephony. As with traditional telephony, the telephone conversation in this case signals into three basic operations, the connection, the call transfer and termination. In contrast to the classical telephony with VoIP but no dedicated " lines " turned on, but the speech is digitized and transported into small data packets over the Internet Protocol.

Signaling protocols

The assembly and disassembly of calls ( call control ) is done via a separate language from the communication protocol. The negotiation and the exchange of parameters for voice transmission happened over protocols other than that of the call control.

To establish a connection with a party in an IP -based network, the IP address of the called subscriber in the network need not necessarily be known, but on the side of the caller. Geographically fixed connections as in the fixed network (PSTN) does not exist in pure IP-based networks. The accessibility of the called party, similarly as in mobile networks, made possible by a previous authentication of the called party and a related notice of his current address. In particular, this may be a connection can be used regardless of the user's whereabouts, what is referred to as nomadic use.

Due to change of location of the subscriber, the user changes on the same PC or dynamic address allocation in building a network connection has a fixed assignment of telephone numbers to IP addresses is not possible. The solution commonly used is that users or their devices to deposit its current IP address to a service computer (server) using a user name. Connection calculator for call control, or sometimes even the terminal of the caller itself, may then at that server check the current IP address of the call partner on the selected user name, and thus establish the connection.

Common signaling protocols are:

SIP - Session Initiation Protocol, IETF RFC 3261
SIPS - Session Initiation Protocol over SSL, RFC 3261
H.323 - Packet- based multimedia communications systems, an ITU -T standard
ISDN over IP - ISDN / CAPI protocol - based
MGCP and Megaco - Media Gateway Control Protocol H.248, common specification of ITU -T and IETF
MiNET - Mitel
Skinny Client Control Protocol - Cisco (not to be confused with SCCP ( Q.71x ) of ITU -T)
Jingle - extension of the XMPP protocol, established by Google Talk

Connect to SIP

The Session Initiation Protocol (SIP ) was developed by the Internet Engineering Task Force ( IETF). As H.323 also enables vendor-independent specification of SIP using SIP -based systems in heterogeneous environments, particularly the coupling of VoIP components from different manufacturers. As with other standards also, however, the interoperability of components by adhering to the specification ( SIP compatibility ) alone is not guaranteed, but must be verified by interoperability testing in individual cases. Basically, SIP is also suitable for application scenarios via VoIP and video telephony addition.

The participants have in the SIP, a SIP address (similar to an email address) in the Uniform Resource Identifier format ( URI format ), such as "sip: [email protected] ". SIP terminals need to register once during the start phase with a SIP registrar server. To establish a connection, the terminal of the caller sends a message to that server is the DNS under the domain name "example.com" known.

This connection request is forwarded by the server to the terminal of the called party. If the message can be processed there, the device sends an appropriate message back to the server, which forwards it to the caller. At this time, ringing the terminal of the called party, the caller hears a ringback tone.

In the course of setting up a session with all relevant information about the properties and capabilities are exchanged between the terminals. A direct communication between the two terminals has not yet taken place to date. For the actual phone call, the server is no longer necessary, the devices send their data directly to, that is, the data exchange in the context of the conversation runs past the server. For the transmission of this data in real time is usually the Real-Time Transport Protocol (RTP) is used.

To end the call, one of the terminals sends a SIP message to the server, which sends it to the other party. Both devices then close the connection.

SIP provides, however, as well as H.323, also the possibility of a direct connection setup between two terminals without using a SIP registrar server, only the IP address before. These, however, must put all existing entries are deleted for SIP registrar server with many devices.

Number systems

Although the IP addresses of the nodes can be used for establishing the connection, these are the users but not always known and may also change. There is therefore currently a number of approaches, to give the participants an individual and convenient mnemonic, independent of the IP addresses of Internet phone number. Ranging from pure SIP numbers, there are approaches to the integration of Internet telephony in the existing numbering plan of the traditional telephone networks to a whole new system. Important aspects of the European Union and the German Federal Network Agency ( FNA, formerly Regulatory Authority ), the compliance with the rules and medium-term integration of emergency response systems.

SIP address

For users who want to make calls over the Internet with other Internet users, many service providers offer SIP addresses. SIP addresses are different than phone numbers or MSNs, not tied to a connection, but as email accounts from any Internet connection in the world available. While this is true also for phone numbers that are assigned to incoming connections a SIP address, yet provides the SIP address va the caller benefits. So phone connections for example, are using the SIP address between two devices is possible, instead, as dialing a telephone number to be routed must necessarily always be via the telephone network.

To get your own SIP address in URI format, you can register at one of the many free or paid providers. Since many vendors either SIP addresses assigned with pure number sequences ( for example [email protected] ) or assign a numeric alias for non-numeric address, and IP phones can be used with a normal keypad for dialing to select interlocutors who when to the same SIP servers have registered. Customers of a SIP service provider can be selected via their SIP address and other call, unless the provider of the called party allows the external SIP request. Most providers of SIP addresses allow access from the conventional telephone network, as it ( the passage from the telephone network to connect the called party ) can generate additional revenue through the termination charges. About this fee detour you can also call the subscribers of other SIP service provider if your provider or the interlocutor locks accordingly. There are agreements that allow customers to communicate directly with each other through the telephone number from some providers. In this case, an Internet connection between the participants is reached, however, the participation of both SIP provider. In general, it is possible within the same provider network, the " internal phone number" ( this is the part of the SIP address before the @ sign) using a commercially available phone to select number field. For this reason, most SIP addresses in this part contain only digits.

Many SIP adapter, which are designed for a conventional telephone connection with number pad, provide the ability to store in the internal phone book SIP address instead of a phone number and activate this SIP address using an assigned speed dial on the phone. In these cases, SIP addresses can be selected at least indirectly by means of a conventional telephone.

Telephone number

A phone number is principally in IP telephony is not absolutely necessary. However, since most compounds using the conventional telephone network reached, the assignment of a SIP address to a traditional telephone number is usually at least for incoming connections still required. For outgoing calls a phone number, however, is unnecessary. To transfer a valid phone number as the sender ID can next to the " internal phone number " (see SIP address ) from many providers, the CLIP function ( no screening ) can be used in a user-defined phone number is transmitted, through which the user is thoroughly accessible. In some countries (eg Germany ) it is prescribed that the provider verifies the specified number as belonging to the customer via a callback ( eg a telephone dialogue system with PIN transmission ).

The separation between providers for inbound and outbound connections for example, is useful if there is anyway via the Internet service provider a phone number for incoming connections and only for outgoing connections, an alternative (often cheaper ) provider is required. For this reason, most providers offer a free phone number only optional at extra cost on, especially if a free telephone connection is offered without PC.

For connection of a telephone number, there are basically two options:

Most internet telephone providers offer telephone numbers for incoming calls to, as this additional revenue can be generated
Other providers, such as the services of the Dellmont group ( Voipbuster, Megavoip etc. ) offer the opportunity to the registered with a third party DDI number ( Direct Dialing In ) to be mapped to its own SIP port ( assign ). In this case, the porting of the number when changing the SIP provider is not required. Care to let this possibility, telephone number and SIP Acount by separate providers, has so far not widely enforced in Germany, but is quite common in other countries
Some providers refrain entirely from mediation incoming calls and offer this possibility is not optional to

ENUM

Phone numbers can be looked up by Telephone Number Mapping (ENUM ) on the Internet. This process is driven by some network operators and both the German (DENIC ) and the Austrian ( Nic.at ) domain registry.

For ENUM, the number is reversed and provided with dots between each digit, preceded as a subdomain of the top-level domain " arpa " with the second-level domain " e164 ". From 49 12345 6789 is so 9.8.7.6.5.4.3.2.1.9.4.e164.arpa for example. However, this solution requires that the customer already has a phone number.

Due to the EU directives on number portability when changing the telephone provider ENUM is currently experiencing (at least in Austria ) the expected upturn. Before phone providers give a call based on our own databases, it is checked whether there is called to the number and the service used in ENUM DNS entry. If so, the call will be to the address specified in the DNS gives ( PSTN or SIP subscribers ).

Unpopular with major commercial vendors is the public recognition of ENUM. This makes it possible on the one hand attackers, automated telemarketing calls free, so-called SPIT ( Spam over IP Telephony) to use. On the other hand, customer data could be consulted. However, ENUM directory operators to prevent automated bulk queries, so that both can narrow hazards through appropriate measures. Another, perhaps major reason for the reticence of many providers towards ENUM is that through free calls account for revenue sources.

Conventional local phone numbers via a gateway

VoIP providers may have their own gateways get free phone numbers from the stock number of the German local networks and loans to their customers. In addition, these customers are then also be reached from the conventional telephone network. However, the Federal Network Agency limited such offers to subscribers who are resident in these local networks. The sometimes difficult to grasp for a location and connection-independent service reasoning is that otherwise, the reference provided, the code for place of residence, should be dissolved. The providers are therefore obliged to check whether the customer in the desired local network actually lives and to create numbers from all local networks where they have customers (want to). For cost reasons, most of the smaller numbers provide VoIP providers only in the larger local networks. If the customer lives outside of an available area codes, make many providers 0180x numbers available. However, this method is only transitional allowed.

When establishing a connection if the VoIP provider using the SIP protocol, the customer has in addition to the local call number at the same time a SIP number. However, many providers shall notify their subscribers with only the assigned landline number. In addition, many of these providers Internet calls from callers who have not registered with them or one of their partner block. This is a free Internet phone call can only be made when both parties from the same supplier (or a partner providers) have registered.

For most businesses and government agencies take over the entire previous numbering plan of the existing conventional connection is ( area code, main phone number and all extension numbers ) requirement for a change to an IP telephony service provider. For SIP, the offer so far only to a few providers.

Special Internet numbers

In Austria, especially for convergent services - which also falls under the Internet Telephony - the preselection 43,780, and the location-independent code 43,720 created. A similar approach was also recommended by the German regulatory authority. After a 032 area code can - similar to mobile phone with a " Block ID " - a VoIP operators are selected then then select the actual final number of the subscriber. The 032 subscriber number is assigned independently of the local network boundaries of geographic numbers and can thus be maintained when moving to other local networks. Since no explicit geographical location is connected with the area code 032, the 032 numbers are generally ideal for nomadic use at different locations.

The 032 numbers were in the past with most VoIP providers do not enforce, but, for example, of the two largest phone companies ( German Telekom and Vodafone ( formerly Arcor ) ) used for their VoIP offerings and increasingly for other value-added services. A lack of accessibility of the 032 number range is now represented on only at a few call-by- call providers; from mobile networks are the numbers since the last release by the lack of major mobile operators, Vodafone, in October 2007, now accessible.

Often the cost of calls to 032 numbers from mobile phone networks for customers, however, are still significantly higher than for calls to landlines. Calls from a landline to a 032- number are, however, often normal telephone conversations equated toll technically ( so even with the current (2009) rates of Deutsche Telekom landlines ).

Call transfer

Just as with traditional telephony, the acoustic signals of the language are first converted analog with a microphone ( on the phone ) into electrical signals. These analog electrical signals are then digitized ( coded). Optionally, it can also be compressed ( spreads are for ITU-T G.723.1 or G.729 Annex A) to reduce the amount of data to be transmitted. The transport of the data thus converted is then performed via a public or private telecommunications network. Due to the method used for the transport of packet-switched data to be divided into many small packets.

Digitization of analog signals and digital processing

The analog speech signal is sampled with a suitable sampling rate for digitizing and the results (samples ) into a periodic sequence of digital signals by an analog -to-digital converter (ADC).

The data rate of that digital data stream is the product of the sampling rate and the resolution of the ADC in bits. It can be reduced if required, prior to transmission by means of coding. Depending on the codec used ( coder-decoder ) different compression factors are possible. Many codecs use this lossy process in which are unimportant to the human auditory information omitted. Which reduces the amount of data, thus reducing the bandwidth needed for transmission substantially without degrading the aural impression appreciably. However, if too much information is omitted, there is a noticeable degradation of the voice quality.

Different codecs use different coding methods are used. Some are specifically designed starting from the standard telephone quality ( sampling rate 8 kHz, 8 bit ADC resolution ) to achieve a significantly lower data rate than the 64 kbit / s ITU standard G.711. Other codecs such as G.722 (see also HD voice ), however, encode from upsampled and resolution digital language with radio or even CD quality ( 7 kHz and bandwidth of the transmitted speech ) with nevertheless a moderate demand for transmission bit rate.

Depending on the digitizing and encoding thus varies the frequency portion of the encoded speech, the bandwidth required for transmission, and the resulting speech quality (source coding). In addition, the coding method may still be adapted to certain typical disturbances to be compensated on the transport path ( channel coding ). For the data after transport can be converted into understandable language for the human ear again, the recipient must use a coder to corresponding decoder, which leads to many devices to ensure interoperability contain multiple codecs.

Transport of data

Normally, each terminal sends the encoded voice data regardless of the signaling " directly " over the network to the IP address of the remote site. So the call data does not flow through servers of a VoIP provider.

The actual transport of the data takes place via the Real-Time Transport Protocol (RTP) or SRTP controlled by the Real Time Control Protocol ( RTCP ). RTP is used for transmission in general, the User Datagram Protocol ( UDP). UDP is used, since it is a minimal connectionless network protocol, which was in contrast not designed for Transmission Control Protocol ( TCP) for reliability. This means that reception of the language is not confirmed, so no transmission under warranty. The advantage of UDP is its lower latency compared to that of TCP, as not to wait for a confirmation and erroneous packets are not retransmitted and thus the overall data flow is not further delayed. A completely error-free transmission is due to the redundancy of spoken language (and the ability of the codecs used to correct errors ) is not necessary. For a liquid interview a small run-time is much more important.

Transmission quality

The demands on the network for data transmission and IP telephony differ significantly. In addition to the required transmission capacity (approximately 100-120 kbit / s for a conversation encoded with G.711 ) have particular quality characteristics such as average delay, variability of delay ( jitter) and packet loss rate significant influence on the resulting speech quality. By prioritizing and appropriate network planning, it is possible to achieve a similar level with the traditional telephony voice quality and reliability of telephone service over IP networks regardless of the traffic load.

As the Internet in its present form (as of 2008 ) does not guarantee secure transmission quality between participants, it may well be transmission interference, echoes, drop outs or disconnections, so that the voice quality is not quite equivalent to that of conventional telephone networks, but usually better than in mobile networks is. With a good DSL connection ( bottleneck is the bit rate towards the net [ upstream ], they should be permanently between 120 and 200 kbit / s per telephone connection are ) you can already get quite an approximately equal and affordable alternative to traditional phone line. For international calls to the USA and Japan, the voice quality is better than with call-by -call primaries using a robust speech codecs such as the iLBC currently (2007 /2008).

An identification and preference ( prioritization) of the " voice " over other data packets on the Internet is useful. Although the IPv4 protocol used on the Internet today offers such possibilities (DiffServ ), but they are not, or not consistently observed by the routers on the Internet generally. However, carefully planned and configured private IP networks can be an excellent "Quality of Service (QoS ) " guarantee (even with Ethernet as physical layer ) and thereby enable the telephony event of an overload in the data area with the usual quality. Status quo on the Internet, however, is so far the best-effort transport, ie the equal treatment of all packets. The still mostly usable telephony quality is due to the over-capacity of networks. To further QoS standards for the future, multimedia -heavy Internet is worked in a number of committees and research projects ( MUSE, DSL Forum, ITU -T).

Also from the follow-up protocol IPv6 QoS with respect to not expect miracles. IPv6 brings a new element flows. So far, however, there is probably still no clarity on how this is to be used. Whether the infrastructure these markers (priority, DSCP code) taken into account or not, is ultimately a financial issue. The future will show if the Internet service providers will also provide higher-quality IP flows for more money.

Quality characteristics

In order to perform a high quality communication over voice-over- IP, the data packets used for voice transport must be arrive at the opposite, that they can be assembled to form a true picture of the original, time- continuous data stream. The factors listed below determine the quality of the system.

In the intranet of the operator of the network can autonomously determine the quality of voice transmission through the server configuration and router equipment and the distribution of access points. On the Internet provider temporarily be involved in the entire chain will determine the transmission quality.

Throughput

The required throughput (amount of data that can be processed by a system or subsystem per unit of time) depends primarily upon the coding used. An uncompressed conversation typically has a data rate of 64 kbit / s (payload ). Depending on the compression method used, the bandwidth required for the pure IP telephony is just a maximum of 100 kbit / s ( 64 kbit / s net plus the overheads of various communication protocols).

As the network is shared with other data services, in particular in the home, a data connection ( such as a DSL connection) with a bandwidth of at least 100 kbit / s recommended in both directions. Here it should be noted that, in the ADSL method often used upstream bit-rate is significantly lower than the downstream bit rate.

Running time ( latency) and jitter

The transport of data takes time. It is used as transit time or latency ( engl. delay or latency ) and is in traditional telephony is essentially the sum of the signal propagation delays on the transmission channels. In telephony over IP networks further delays by the packaging and interim storage, and where appropriate, data reduction, compression and decompression of data are added. In telephony ( is it what technology they implemented independently ) submitted in accordance with ITU -T Recommendation G.114 to 400 milliseconds one-way running time ( mouth to ear ) the limit is up to the quality of real-time communication is still considered acceptable. From about 125 milliseconds, however, the term man can already be perceived as disturbing. Therefore, in general, the ITU-T recommended not to exceed for highly interactive communication form a single run-time of 150 milliseconds.

Jitter refers to the variation over time between the reception of two data packets. To compensate for this so -called " buffer memory " ( Jitter ) is used, which effect an additional deliberate delay of the received data in order to subsequently output the data isochronously. Packets that arrive later, can not be incorporated into the output data stream. The size of the buffer (in milliseconds) is added to the runtime. So it allows the choice between more delay or higher packet loss rate.

Packet loss

From packet loss is when data packets sent does not reach the receiver and are therefore discarded. For real- time applications, we also speak of packet loss when the packet reaches the receiver though, but arrives too late to be inserted into the output current. For telephony according to ITU- T G.114, a packet loss rate ( packet loss rate) up to 5% still classified as acceptable.

Availability

The availability of the overall system is obtained from the individual availability of the participating components and their interconnection (cascaded - in series or redundant - in parallel). Thus, the availability of an IP telephony system depends primarily on the network design. A U.S. study in June 2005 examined the availability of IP telephony in the United States. On average, almost 97 % have been achieved. This corresponds to a failure to complete a total of 11 days a year. In addition there are many German DSL providers a so-called 24 -hour forced separation, which means that at constant unused cable there is separation.

Architecture

There are different architectures for VoIP. Widely used are: the architecture according to the H.323 standard framework of the ITU- T that provides the elements terminal, gateway, gatekeeper and MCU and the architecture according to the de facto standard of the IETF SIP. There are also a number of non-standard solutions for VoIP.

Terminal

A terminal is in the ITU terminology of " multimedia endpoint " of communication, in the strict sense, therefore, the terminal for input and output of the speech information. His ( approximate ) equivalent in the SIP terminology of IETF is the user agent.

Terminal types

There are three basic types of terminals with which you can use IP telephony.

With a running on the PC software, known as a softphone.

With a directly be connected to the LAN ( S) IP phone or a wireless telephone for wireless networks. In this case, no PC is needed to make phone calls (except possibly for configuration tasks or to facilitate certain transactions such as the acquisition of short codes, the entry of alphanumeric data, or the like. ).
With a conventional telephone that is connected via an analog or ISDN phone adapter for VoIP ( ATA and ITA) to the LAN. ATA and ITA are available now integrated directly as a connection for phones in DSL routers. Also in this case, no PC is needed for one-time set up the user data on the other hand already for telephony operation.

But terminals for GSM mobile telephony have the opportunity to lead IP calls on available wireless network (see open source operating system Openmoko ). These types of terminals connect for reasons of cost GSM mobile and IP telephony, to use with Wi-Fi throughout the lower-cost IP telephony with the mobile phone can.

However, problems with the use of Voice over WLAN are yet the lack of standards for bandwidth management on the air track (too much user activity on the same access point causes critical packet loss rate of the VoIP connection ) and handover ( termination of the connection with movement of the terminal to another Access Point ) as well as battery-operated terminals, the high power consumption.

Fax over IP Fax over IP ( FoIP )

To send fax via ISDN or analog connections the T.30 protocol is used in the voice channel. Due to the high reliability of a voice channel connection in traditional TDM - based networks, secure transmission is usually guaranteed. This is the case in IP networks but not because language is here unsecured usually transmitted ( RTP over UDP), despite the same encoding of the language, such as the G.711 codec, which is used in TDM -based networks and IP networks will. IP packets can be lost and are usually in the amount of up to 5% of losses for the human ear imperceptible. The fax transport over an IP network by means of such a voice codecs, optimized for human speech coding used therein, but leads to loss of information or disconnects the fax.

To send faxes over IP networks, the following encodings or protocols in the voice channel are used:

About a speech codec: Fax over VoIP, reliable transmission is not always possible
E-mail
T.37 (email based)
Real- time: T.38

This results in different approaches to use Fax over IP ( FoIP ):

One uses a conventional analog fax machine and want to use this in an IP network as well as in a TDM -based telephone network with analog or ISDN connection. (This is the most demanded solution. )
One uses a fax machine with direct T.38 or e -mail support and network connection and simultaneously ensures that a gateway with T.38 or e- mail support are available with access to the PSTN telephone network and a gatekeeper available.
There are also fax machines, which are designed for direct sending and receiving faxes via T.38.

Gateway

This will allow connection to traditional telephone networks can be prepared are switching computer, the so-called gateways, are required. These are connected to both the communication network of the IP phone, as well as with the conventional telephone network (PSTN). Receive a request from an IP phone, they pass them on to the telephone network by calling the number. Receive a call from the telephone network, they conduct an inquiry to the appropriate IP phone on.

Gatekeeper

A gatekeeper is an optional component in the H.323 environment and fulfills central functions such as terminal registration or establishment and release of connections between registered terminals.

Multipoint Control Unit ( MCU)

The optional Multipoint Control Unit (MCU ) is used for H.323 used in applications where connections between more than two terminals are required ( telephone or video conference). Here, the negotiation of terminal attributes, and control of the conference. If necessary. A reaction of different codecs and bit rates and the distribution of information -mixed multicast.

Applications

Direct Internet telephony

The IP telephony is used to lead the world calls directly via the Internet, called Internet telephony. The app uses the phone network is no longer used.

For end users ( home and home office ) reasons for using particular:

Save fees by IP telephony. As terminals can be connected analogue or ISDN terminals, sound -capable computer (preferably with handset or headset ) is used as well as special IP phones both Specific Adapter (ATA, ITA ). For calls between two IP telephony subscribers usually fall no call charges.
Also the connection to and from participants in the conventional telephone network is possible. She will be produced by a vendor-supplied transition, the Gateway service. But normally fall to special charges for outgoing calls through gateways.
Regardless of location, accessibility is always given at the same address or telephone number.

Organization internet calling

Within organizations, such as business IP telephony is increasingly used to bring together the telephone network and the computer network. The data transfer of the telephone calls, both for signaling and the transmission of digitized speech, is carried out via the computer network (LAN). Thus, the infrastructure costs through uniformity of cabling and active system components can be reduced. The IP phones are usually connected as a desktop PC to the network connector. Conventional terminals have to be replaced or to be adapted.

The telephony services, in particular the subscriber management and call handling are provided via IP -enabled telephone systems, which are also connected to the network. PBXs at different locations can be coupled via the extranet (WAN) and existing data lines with spare capacity. Not all of these different locations have to be equipped with their own telephone system. Sites where no local telephone system is installed are referred to as remote units. For connections to the ordinary telephone network, such as the public switched telephone network (PSTN), are transitions, called gateways, inserted between the IP network and the conventional network.

The structure of the overall system is described in so-called scenarios, which can contain multiple transitions between conventional and VoIP telephony. Known as the migration transition from traditional telephony to VoIP is usually gradual. Successive be part of a company, preferably new departments, equipped with the new technology.

By combining PBXs that provide both IP and legacy ports, a creeping migration ( Smooth migration ) is possible using conventional connections can continue to operate and are gradually replaced by IP connections. These PBXs are also referred to as hybrid systems.

Voice quality and reliability of telephone technology hang about switching to VoIP completely from the network technology from what is given special consideration in the planning and administration of networks and requires much higher hardware requirements.

Background - technology conventional telephony

Conventional telephone networks in Europe are based on the circuit-switched PCM30 method. On the part of the operators of telephone networks can also be used IP telephony without that would bring a change for the conversation participants for the transmission of conversations. The use of IP telephony can take place for parts of the network or the entire network.

Even more IP telephony, for example, used by call-by- call providers for international calls. Calls are routed between the local telephone network and the telephone network of the destination country via the Internet, resulting in cost advantages.

Next Generation Networks ( NGN) only use packet switching networks for telecommunications. The aim is to make efficient use of network resources and to create a common platform for all services. In this case, a separation between the transport and service level. According to press reports from the year 2004, the German Telekom plans to have completed the full conversion of its network by 2015.

Connection charges

If both parties are connected to the Internet, falling in Internet telephony normally, apart from the cost of internet use, no additional costs. So calls are using an open SIP server world free for participants with an internet flat rate. However, some VoIP providers limit the scope of free telephony to users who have registered with them or one of their partner. Also, in case the user to talk toll-free telephony remains the possibility of direct addressing his interlocutor about the IP address without using a VoIP service provider.

For calls from the Internet to a subscriber in the classic telephone network, a gateway is needed that accomplishes the connection. For its use to pay costs that consist of the provision of infrastructure and the cost of calls to the telephone network.

For international calls to a subscriber in the classic telephone network, the location of gateways is critical: Up to the gateway of cheap internet access is used, then the phone prices are valid gateway provider.

If an existing enterprise network used for IP telephony, there are no call time -dependent link costs. In addition to the cost of VoIP-enabled network components (routers and LAN switch ) the pro rata cost of the network bandwidth must be included in an economic analysis. The necessary bandwidth is derived from the dependent on the codec used bandwidth per call and the expected number of simultaneous calls.

Safety aspects

The integration of voice transmission into the IP network to new challenges to IT security.

The VoIP packets are transmitted over a so-called " shared medium ", ie over a network, which several participants and different services share. Under certain circumstances it may be possible for an attacker to tap the data on the transmission and record the conversation. Example, there are programs that help the data stream also switched environments tapped by means of " ARP spoofing " and it again an audio file can be generated.

Although there is the possibility of the transmission with Secure Real - Time Transport Protocol ( SRTP) to encrypt, but which is rarely used by the users, since most VoIP providers do not support it. Another reason is certainly the ignorance of this possibility, in addition, a encryption also affect the voice quality, which is why users often decide in favor of voice quality against the higher security.

The Session Initiation Protocol (SIP ) is often used may also not be considered in all encountered in practice forms as probable. Although it has security mechanisms ( for example, Call- IDs based on hash functions ), but provides opportunities for attack denial of service attacks.

Another safety-relevant area is not indeed exclusively limited to this technique, however, favored by the low costs that are incurred for the talks. So there is the possibility of a kind of " VoIP spam ", also SPIT ( "Spam over Internet Telephony " ) called.

When vishing, the counterpart to phishing, criminals lead unsuspecting bank customers to fake hotlines, designed to obtain their passwords.

In addition, the phreaking with VoIP could speak experiencing a revival. The scenario is based on that in the VoIP communication signaling (e.g., SIP) is decoupled from the speech data (payload, such as RTP). Two specially prepared clients build through the SIP proxy on a conversation and behave completely standards compliant. After the call setup the SIP proxy a conversation breakdown is signaled. This sees the session as terminated and recorded the conversation. The RTP data stream, however, is maintained by the clients. The caller then calls away for free.

Reliability

By eliminating the traditional telephone lines, the local data network companies a single point of failure for communication among employees dar. Was this without VoIP in case of failure of a network component such as a switch or router still be reached by phone, the VoIP is no longer the case or only to a limited extent via mobile phones. An investment in a redundant network can reduce this risk.

Power supply

In classical ( circuit-switched ) telephone networks, connections have been operated with an official remote power supply that powers the terminal independent of the local power supply with energy. While this remote power supply for devices to analog subscriber lines still for a full operation, with ISDN is sufficient for a single device in emergency mode, it is insufficient for a power supply of equipment for the operation of VoIP ( for example, routers, terminals ).

Should still be possible VoIP functionality even when a local power failure in these ports, so have all the components, DSL modems, routers, VoIP terminals, are protected by an uninterruptible power supply.

A similar situation exists, however, in many modern analog telephones. Particularly most cordless telephones also work without local power supply of the base station.

Location and emergency calls

Since the phone number is not strictly localized, localization of the calling party is restricted. The problem is mainly for emergency calls to be answered as possible from the nearest monitoring station. It also relates to deals that have geographical access numbers to provide region-specific information ( information services, service, or call center, special numbers ).

Localization generally

Since the phone numbers are not bound, the country code depends solely on the SIP provider. Therefore, leaves out the country code (eg 49 for Germany ) does not refer to where the call actually comes.

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