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With the help of authentication it is ensured that a message or a call actually originates from the person or institution posing as the sender.

The user registration is a process through which the access authorization of a user to a service is checked with his access data. Access authorizations are checked in a great number of areas of application of computer technology.

In the area of Internet telephony two VoIP devices are required that are able to identify each other before the actual communication can begin.

The mutual authentication is based on a secret known to both parties, which makes access and identity forgery by attackers extremely difficult.


As with conventional telephony, with VoIP the speech is initially captured in analog form with a microphone. This analog information is then transferred into a digital format by a converter and changed through codecs into corresponding audio-binary formats. Depending on the codec used, the data can be compressed to differing extents in this process.

Most codecs use a procedure through which - similar to MP3 files - information not important for the human ear is omitted. This reduces the amount of data and thus reduces the bandwidth required for transfer. However, if too much information is omitted, the speech quality will suffer.

The different codec procedures master the audio compression with different levels of efficiency. Some are specifically designed to achieve a low bandwidth at any cost. Depending on the codec, therefore, the necessary bandwidth and the speech quality will vary. In order for the data also to be converted correctly back into speech after the transport, the receiver must use the same codec as the sender.


The Domain Name Service ensures the determination of an IP address to a given domain name. This Domain Name Service runs on all PCs. It receives the domain name entered by a user and inquires after the corresponding IP address from a DNS Server known to it. If a DNS server cannot answer the inquiry itself, it has the possibility to inquire about the IP address from other servers.

If the Domain Name Service receives negative information from the DNS server (domain name not recognized), it can make inquiries to other DNS servers known to it or issue the user with a corresponding error message. If, on the other hand, it receives the desired IP address, the application can address the target desired by the user my means of the IP address.

The hierarchical system of DNS servers is described as the Domain Name System. The IP addresses of the DNS severs to which the Domain Name Service should make its standard inquiries are mostly delivered to the PC automatically by the Internet provider during the Internet dial-up. In local networks an assignment of addresses can also ensue via DHCP. Otherwise they have to be entered manually by the user or systems manager in the TCP/IP configuration of the PC.


The goal of ENUM is to make available different addresses, numbers and URL's under one single number.

In this way, the private home telephone, the work telephone, the fax number, cell phone numbers, business and private E-Mail addresses, video conference addresses, personal website and all other conceivable communication addresses can be summarized under one single ENUM number.

Depending on whatever application is currently being used (e.g. telephone, E-Mail program etc.) this searches under the given ENUM number for the actual target address.

Interactive Connectivity Establishment (ICE)

Since the beginning of the development of SIP-based devices for Internet telephony there have been problems with NAT traversal. Many suggestions have been made (Connection Oriented Media, STUN, TURN, SIP), but none has been universally realized.

All suggestions were gathered together in a document that is described as ICE, which stands for Interactive Connectivity Establishment. It is a methodology for NAT traversal.

It is not a new protocol but rather makes use of all noteworthy attempts at a universally functioning NAT traversal.

This methodology is hugely complex and requires a high degree of cooperation of all endpoints involved in the SIP communication. ICE always works, regardless of the type and number of NAT's.

See also STUN


The IEEE was founded in 1963 and is concerned with the standardization of local networks. It began in February 1980 with a project for the standardization of different networks. These standards were therefore named IEEE 802 norms.

IETF is the abbreviation for "Internet Engineering Task Force". It sees itself as an open community for all technicians who are devoted to the development of the Internet.

The IETF is concerned with current developments and standardizations on the Internet, which are elaborated in working groups and published in Internet drafts, for which the term RfC (Request for Comments) is also used.

Internet Service Provider (ISP)

Internet Service Providers (ISP) are service providers that give their customers access to the Internet.

For this purpose, the ISP makes an access point available to the customer, which is either linked permanently via a hard-wired connection or which can be temporarily contacted via a dial-up connection.

In addition, the ISP provides the customer with further Internet-related services, such as E-Mail accounts, storage space on FTP servers and personal homepages.


"Network Address Translation" is a method for translating the (mostly private) IP addresses of a network onto other (mostly public) IP addresses of another network.

NAT therefore enables several PCs in an LAN on the one hand to use the IP address of the Internet Access Router for Internet access and on the other it hides the LAN behind the IP address of the router registered on the Internet.

NAT therefore spares the need for each user to have a separate provider contract.

If, then, the client in the LAN sends an IP packet to the router, NAT converts the address of the sender into a valid IP address, which for instance has been assigned to it by the provider, before it is passed on onto the Internet.

If an answer to this packet comes back from the remote station, the NAT converts the receiver address back into the original IP address of the local station and delivers the packet in proper form. In theory, NAT can manage LANs with any number of clients.


Power over Ethernet is the term for the standard IEEE 802.3af, which enables end appliances that have less then 12.95 Watt power consumption to also be supplied with electricity directly through the Ethernet cable.


"Point of Presence" describes the dial-in nodes of Internet Service Providers through which the customer gains access to the Internet.

Other points of interconnection are also described as PoP - for example the dial-in node in a local network (Intranet) or the points of interconnection between subnetworks of the Wide Area Network.

The access points for customers who use broadband Internet access technologies such as ADSL are also described as broadband PoP.


"Public Switched Telephone Network" describes public switched telephone networks that are based on analog technology. Decisive for this classification of a network is the technology used in the switching centers.

The main purpose of a PSTN is the conveying of connections of the analog telephone service (POTS). In addition, data connections over analog modems as well as analog fax connections can also be switched.


"Plain Old Telephone Service" describes the analog telephony with which a frequency range of 300 Hz to 3.4 kHz is used. This results in a bandwidth of 3.1 kHz.

Quality of service (QoS)

Quality of Service (QoS) is a term that is frequently used in relation to VoIP.

If the DSL line is working to capacity, speech packets of a telephone call only arrive at the telephone with delay or even not at all. Packaged in a router it guarantees bandwidths for certain services.

If, therefore, the line is working to capacity, QoS slows down the download speed as soon as a call arrives. In this way, the necessary bandwidth for the call is guaranteed.


In contrast to H.323, SIP was developed by the IETF (Internet Engineering Taskforce) with the Internet in mind and is therefore oriented towards the architecture of common Internet applications.

From the beginning, attention was paid to easy implementability, scalability, expandability and flexibility. SIP can be used to manage any number of sessions with one or several participants. However, it is not limited to Voice over IP as sessions can be any number of multimedia streams or conferences.

The security standard SIPS not only prevents eavesdropping and message manipulation, but also ensures the proxy server about the identity of the snom client phone and protects against identity spoofing.

Through the use of AES (Advanced Encryption Standard) in the counter mode for secure RTP one single key stream emerges for each RTP packet, which makes it practically impossible to retrieve an original RTP stream and abuse it.


Realtime Transport Protocol / Secure Real Time Transport Protocol

The task of RTP is to transport the actual data stream in a connection, for example in terms of Internet telephony the audio data. Transporting means coding, packing and sending of data.

SRTP is ideal for the protection of VoIP traffic as it can be used in conjunction with header compression and does not have any effect on the quality of the IP service. This brings with it decisive advantages above all for the data traffic, which uses speech codecs with low transfer rates.

See also TLS/SSL

Simple traversal of UDP over NAT (STUN)

The "Simple traversal of UDP over NAT" is a data protocol that enables an IP telephone to recognize the existence and the type of an NAT or firewall and to bypass it.

A telephone that supports STUN can independently replace its "private" IP and ports in its data stream with the "public" IP and ports. For this it requires a STUN server in the "public" Internet.

In this way, after correct configuration, the SIP signalling and the speech transfer can ensue through the NAT/firewall without necessitating changes to the NAT.

See also ICE


Only the description "Transport Layer Security" indicates that the concern in terms of this technology is with a protocol of the transport layer. This layer guarantees a reliable and transparent data transfer between two systems.

It also acts as an interface between the data layers above it and the network-oriented layers below. The central task is the connection setup and control between processes.

In the transport layer the Secure Socket Layer (SSL) introduced by Netscape has played a central role so far for the exchange of relevant information. With its Private-Communication-Technology-Protocol (PCT), Microsoft has attempted to push through a very similar security technology. Today it seems certain that both will be replaced in the future by TLS. The TLS specification is largely based on Secure Socket Layer (SSL).

The goal of the Transport Layer Security (TLS) is to provide a mechanism that allows data protection and data integrity between two applications. A VoIP telephone linked with TLS can be configured in such a way that only secure SIP signalling with other devices takes place.

It therefore belongs to the protocols with which the communication betweens server and client can be cryptographically protected. For the authentication of the communication partners certificates can be used.

See also FAQ: How can I setup snom3xx phones for TLS support?


Universal Plug and Play is an architecture for pervasive peer-to-peer network connection with PCs and intelligent devices or applications, above all for SOHOs (small offices/home offices).

UpnP is based on Internet standards and technologies such as TCP/IP, HTTP, and XML, in order to enable a connection to be made automatically between the end devices.

User Agent

With VoIP the end points of a connection over Internet Protocol are generally described as user agents. The somewhat more precise description SIP user agent implies that these user agents are deployed together with the signalling protocol SIP.

A user agent is either the actual VoIP end device, such as an IP telephone or VoIP software on a PC, or a network node that converts the VoIP connection to another medium. Network nodes acting as SIP user agents can, for example, be a SIP gateway or a VoIP telecommunications system operating as a bridge.

Voice over IP (VoIP)

IP telephony, also known as Voice over IP (abbrev. VoIP) is the telephoning over a computer network on the basis of the Internet Protocol.

If IP telephony is used to carry out conversations over the Internet one speaks of Internet telephony.

The fundamental difference to conventional telephony consists in the fact that the speech information is not transferred over a switched connection in a telephone network, but rather divided into IP packets, which reach their goal on non-designated paths in a network.

IP telephony can share its infrastructure, i.e. the network, with other communication services.

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