VoIP FAQ 
VoIP FAQ | |
There are 24 entries in the FAQ.
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Questions:
What is DID Number? VoIP H.323 SIP IAX MGCP E.164 FXO FXS IVR DID/DDI - Direct Inward Dialing T.38 FOIP Voice Activation Detection (VAD)? CODECS RTP Silence Suppression ? Echo Cancellation RTCP RTP Softphone IP Phone Analog Telephone Adaptor (ATA) VoIP Gateway IP Protocol| Questions and Answers | |
| What is DID Number? | |
 | DID is a short for Direct Inward Dial Number ,it is Regular telephone number rings through to your VoIP or a normal phone. The caller will dial your local DID (example Canada number) and the call will reach you while you are in another country (like in China).It is also called virtual phone numbers. |
| VoIP | |
| VoIP (voice over IP) is an IP telephony term for a set of facilities used to manage the delivery of voice information over the Internet. VoIP involves sending voice information in digital form in discrete packets rather than by using the traditional circuit-committed protocols of the public switched telephone network (PSTN=Public Switch Telephone Network). A major advantage of VoIP and Internet telephony is that it avoids the tolls charged by ordinary telephone service. |
| H.323 | |
| H.323 is an umbrella recommendation from the ITU-T, that defines the protocols to provide audio-visual communication sessions on any packet network. It is currently implemented by various Internet real-time applications such as NetMeeting and Ekiga (the latter using the OpenH323 implementation). It is a part of the H.32x series of protocols which also address communications over ISDN, PSTN or SS7. H.323 is commonly used in Voice over IP (VoIP, Internet Telephony, or IP Telephony) and IP-based videoconferencing. Its purpose is thus similar to that of the Session Initiation Protocol. |
| SIP | |
| "The Session Initiation Protocol (SIP) is an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants. These sessions include Internet telephone calls, multimedia distribution, and multimedia conferences." (cit. RFC 3261). It was originally designed by Henning Schulzrinne (Columbia University) and Mark Handley (UCL) starting in 1996. The latest version of the specification is RFC 3261 from the IETF SIP Working Group. In November 2000, SIP was accepted as a 3GPP signaling protocol and permanent element of the IMS architecture. It is widely used as signaling protocol for Voice over IP, along with H.323 and others. SIP is addressing neutral, with addresses expressed as URL/URIs of various types, such as H.323 address, E.164 telephone numbers or email like addresses. SIP is a lightweight, transport-independent, text-based protocol. SIP has the following features: * Lightweight, in that SIP has only six methods, reducing complexity * Transport-independent, because SIP can be used with UDP, TCP, ATM & so on. * Text-based, allowing for low overhead |
| IAX | |
| IAX (Inter-Asterisk eXchange protocol) is an alternative to the SIP and H323 protocols for initiating and transmitting voice (and other) data over IP. It's a protocol which has been produced by the group developing the Linux-based PBX software Asterisk rather than being a standards-based system. It was developed because it was felt that SIP insufficiently supported: Interoperability with firewalls, a high performance, low overhead protocol, internationalization support, remote dialplan polling, flexible authentication, multimedia support, call statistic gathering, and call parameter communication. |
| MGCP | |
| Short for Media Gateway Control Protocol, developed by Telcordia and Level 3 Communications, a control and signal standards to compete with the older H.323 standard for the conversion of audio signals carried on telephone circuits (PSTN ) to data packets carried over the Internet or other packet networks. The reason new standards are being developed is because of the growing popularity of Voice over IP (VoIP ). Regular phones are relatively inexpensive because they don't need to be complex; they are fixed to a specific switch at a central switching location. IP phones and devices, on the other hand, are not fixed to a specific switch, so they must contain processors that enable them to function and be intelligent on their own, independent from a central switching location. This makes the terminal (phone or device) more complex, and therefore, more expensive. The MGCP is meant to simplify standards for this new technology by eliminating the need for complex, processor-intense IP telephony devices, thus simplifying and lowering the cost of these terminals. |
| E.164 | |
| E.164 offers a method of using your current telephone number as a means of being contacted in the IP world. Your telephone number gets mapped into a DNS zone, and the zone can contain your contact information (VoIP, instant messenger, email, anything). See e164.orge164.org for further details. |
| FXO | |
| FXO stands for Foreign eXchange Office: simply put, this interface connects to the analog PSTN line coming from the central office. FXS stands for Foreign eXchange Station: simply put, this interface connects to devices such as analog phones and fax machines. |
| FXS | |
| FXS stands for Foreign Exchange Station, and it's the New Communication Age version of a telephone switchboard. It's not so obvious now that most calls are direct dial; in the old days, you dialed in to an operator and told him or her what number you wanted to be connected to; the operator would then dial the number and connect you to your intended chat partner. You may not realize it in the modern age, but phone calls still take place that way; today, however, the switchboard is an automated one. |
| IVR | |
| In telephony, Interactive Voice Response – is a computerised system that allows a person, typically a telephone caller, to select an option from a voice menu and otherwise interface with a computer system. |
| DID/DDI - Direct Inward Dialing | |
| DID - Direct Inward Dialing (also called DDI in Europe) is a feature offered by telephone companies for use with their customers' PBX system, whereby the telephone company (telco) allocates a range of numbers associated with one or more phone lines. Its purpose is to allow a company to assign a personal number to each employee, without requiring a separate phone line for each. That way, telephony traffic can be split up and managed more easily. DID requires that you purchase an ISDN or Digital line and ask the telephone company to assign a range of numbers. You then need DID capable equipment at your premises which consists of BRI, E1 or T1 cards or gateways. |
| T.38 | |
| T.38 is an ITU standard which deals with sending fax messages over IP networks. It is used together with Session Initiation Protocol and Session Description Protocol. The need for reliable faxing over IP networks (the Internet) has increased as the popularity of Voice over IP has increased. Faxes operate by scanning a document, converting the document into data, and sending that data as sounds over a telephone line to a receiving fax which decodes it. This does not operate correctly on many VoIP systems, designed to convert voice sounds to data over the internet, which lose tones required for faxing. The T.38 standard, when implemented on a VoIP ATA and on the VoIP gateway it connects to, is designed to convert fax sounds to data and enable reliable faxing. |
| FOIP | |
| FoIP, short for fax over Internet Protocol, the technology that enables the internetworking of fax machines with a packet-based network. Using FoIP, a fax is transmitted via the Internet rather than the traditional method of sending faxes via the telephone line. Using FoIP, the digital data from the fax machine is separated into packets for transmission (as opposed to the traditional method of converting the fax data into analog to be sent over the PSTN). The digital data requires less bandwidth than the analog data, so FoIP is more efficient than analog faxing. |
| Voice Activation Detection (VAD)? | |
| In Voice over IP (VoIP), voice activation detection (VAD) is a software application that allows a data network carrying voice traffic over the Internet to detect the absence of audio and conserve bandwidth by preventing the transmission of "silent packets" over the network. Most conversations include about 50% silence; VAD (also called "silence suppression") can be enabled to monitor signals for voice activity so that when silence is detected for a specified amount of time, the application informs the Packet Voice Protocol and prevents the encoder output from being transported across the network. Voice activation detection can also be used to forward idle noise characteristics (sometimes called ambient or comfort noise) to a remote IP telephone or gateway. The universal standard for digitized voice, 64 Kbps, is a constant bit rate whether the speaker is actively speaking, is pausing between thoughts, or is totally silent. Without idle noise giving the illusion of a constant transmission stream during silence suppression, the listener would be likely to think the line had gone dead. |
| CODECS | |
| A codec (Coder/Decoder) converts analog signals to a digital bitstream, and another identical codec at the far end of the communication converts the digital bitstream back into an analog signal.
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| RTP Silence Suppression ? | |
| Endpoints sending audio as an RTP stream are not required to send packets during silent periods. The capability to stop sending RTP packets during silent periods is known as "Silence Suppression" or VAD (Voice Activity Detection). Whether to use Silence Suppression is usually a configuration option on endpoints. When processing a stream of RTP packets, here is what RFC 3389 has to say about detecting Silence Suppression: |
| Echo Cancellation | |
| Echo cancellation is the process of removing echo from a voice communication in order to improve the voice call quality. Echo cancellation is often needed because speech compression techniques and packet processing delays generate echo. There are 2 types of echo: acoustic echo and hybrid echo. Echo cancellation not only improves quality but it also reduces bandwidth consumption because of its silence suppression technique. |
| RTCP | |
| RTP Control Protocol (RTCP) is a sister protocol of the Real-time Transport Protocol (RTP). It is defined in RFC 3550 (which obsoletes RFC 1889). RTCP stands for Real-time Transport Control Protocol, provides out-of-band control information for an RTP flow. It partners RTP in the delivery and packaging of multimedia data, but does not transport any data itself. It is used periodically to transmit control packets to participants in a streaming multimedia session. The primary function of RTCP is to provide feedback on the quality of service being provided by RTP. It gathers statistics on a media connection and information such as bytes sent, packets sent, lost packets, jitter, feedback and round trip delay. An application may use this information to increase the quality of service perhaps by limiting flow, or maybe using a low compression codec instead of a high compression codec. RTCP is used for QoS reporting. There are several type of RTCP packets: Sender report packet, Receiver report packet, Source Description RTCP Packet, Goodbye RTCP Packet and Application Specific RTCP packets. RTCP itself does not provide any flow encryption or authentication means. SRTCP protocol can be used for that purpose. |
| RTP | |
| The Real-time Transport Protocol (or RTP) defines a standardized packet format for delivering audio and video over the Internet. It was developed by the Audio-Video Transport Working Group of the IETF and first published in 1996 as RFC 1889 which was obsoleted in 2003 by RFC 3550. RTP does not have a standard TCP or UDP port on which it communicates. The only standard that it obeys is that UDP communications are done via an even port and the next higher odd port is used for RTP Control Protocol (RTCP) communications. Although there are no standards assigned, RTP is generally configured to use ports 16384-32767. RTP can carry any data with real-time characteristics, such as interactive audio and video. Call setup and tear-down is usually performed by the SIP protocol. The fact that RTP uses a dynamic port range makes it difficult for it to traverse firewalls. In order to get around this problem, it is often necessary to set up a STUN server. |
| Softphone | |
| In computing, a softphone is a piece of software for making telephone calls over the Internet using a general purpose computer, rather than using dedicated hardware. Often a softphone is designed to behave like a traditional telephone, sometimes appearing as an image of a phone, with a display panel and buttons with which the user can interact. A softphone is usually used with a headset connected to the sound card of the PC, with a USB phone or with a "Plain Old Telephone" connected to the PC using an adapter (some types of sound cards or chatcord) |
| IP Phone | |
| A telephone that converts voice into IP packets and vice versa for voice over IP (VoIP) telephone service. The term usually refers to a telephone with built-in IP signaling protocols such as H.323 or SIP that is used in conjunction with an IP PBX in an enterprise. However, it may also refer to a software-based phone (softphone) that is installed in the user's PC and requires that calls be made from the PC |
| Analog Telephone Adaptor (ATA) | |
| An analog telephone adaptor (ATA) is a device used to connect a standard telephone to a computer or network so that the user can make calls over the Internet. Internet-based long distance calls can be substantially cheaper than calls transmitted over the traditional telephone system, and ATAs are typically cheaper than specialized VoIP phones that connect directly to a computer's Universal Serial Bus (USB) port. |
| VoIP Gateway | |
| A VoIP gateway is a network device that converts voice and fax calls, in real time, between the public switched telephone network (PSTN) and an IP network. The primary functions of a VoIP gateway include voice and fax compression/decompression, packetization, call routing, and control signaling. |
| IP Protocol | |
| The Internet Protocol (IP) is the method or protocol by which data is sent from one computer to another on the Internet. Each computer (known as a host) on the Internet has at least one IP address that uniquely identifies it from all other computers on the Internet. When you send or receive data (for example, an e-mail note or a Web page), the message gets divided into little chunks called packets. Each of these packets contains both the sender's Internet address and the receiver's address. Any packet is sent first to a gateway computer that understands a small part of the Internet. The gateway computer reads the destination address and forwards the packet to an adjacent gateway that in turn reads the destination address and so forth across the Internet until one gateway recognizes the packet as belonging to a computer within its immediate neighborhood or domain. That gateway then forwards the packet directly to the computer whose address is specified. |