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Supports a variety of hardware interfaces for connecting telephony channels.

Traditional TDM hardware resources including echo cancelling, HDLC controllers, conferencing DSP's and DAX's are replaced with software equivalents. With software TDM, switching is still done in near-real-time, and call qualities are excellent. The pseudo-TDM architecture extends the TDM bus across Ethernet networks. Zaptel devices support data modes on clear channel interfaces, including Cisco HDLC, PPP, and Frame Rela

Interfaces for connectivity to traditional legacy telephone services that do support TDM switching

Packet Voice Protocols

These are standard protocols for communications over packet networks like IP or Frame Relay. These interfaces do not rely on specialized hardware. These interfaces will work without specialized hardware.

Session Initiation Protocol (SIP)
Inter- Exchange (IAX) versions 1 and
Media Gateway Control Protocol (MGCP)
ITU H.32
Voice over Frame Relay (VoFR)

Telephony Interface

The LinuxTelephony Interface was developed primarily by Quicknet, Inc. with help from Alan Cox. This interface is geared toward single analog interfaces and provides support for low bit-rate codecs.

The following products are known to work with although they may not work as well as devices.

Quicknet Internet Phonejack (ISA, FXS)
Quicknet Internet Phonejack PCI (PCI, FXS)
Quicknet Internet Linejack (ISA, XO or FXS)
Quicknet Internet Phonecard (PCMCIA, FXS)
Creative Labs VoIP Blaster (limited support)

The OSS and ALSA console drivers allow a single sound card to function as a "console phone" for placing and receiving test calls. Using auto answer/auto hang up, the console can create an intercom

supports Adtran's proprietary Voice over Frame Relay protocol. The following products are known to talk to using VoFR. You will need a Sangoma Wanpipe or other frame relay interface to talk to them

Adtran Atlas 800
Adtran Atlas 800+
Adtran Atlas 550

Supported VoIP Protocols

supports two industry standard and one specific VoIP protocols.

IAX is the specific VoIP protocol. It is the standard VoIP protocol for networking. It provides transparent interoperation with NAT and PAT (IP masquerade) firewalls. It supports placing, receiving, and transferring calls and call registration. With IAX, phones are totally portable. Just connect a phone or server anywhere on the Internet. They will register with their home PBX and instantly route calls appropriately.

IAX is extremely low-overhead. IAX has four bytes of header, as compared to at least 12 bytes of header for RTP based protocols like SIP and H.323. IAX control messages are substantially smaller.

IAX supports internationalization. A requesting PBX or phone can receive content from the providing PBX in its native language.

IAX supports authentication on incoming and outgoing calls. provides fine-grained control over access. Limits can be placed on access to only specific portions of the dial plan.

With IAX dial plan polling, the dial plan for a collection or cluster of PBX's can be centralized. Each PBX only needs to know its local extensions, and can query the central PBX for further information as required

SIP is the IETF standard for VoIP. SIP is described at greater length in a following . SIP control syntax resembles SMTP, HTTP, FTP and other IETF protocols. SIP runs over TCP/IP and manages Real Time Protocol RTP) sessions. RTP transfers the data for a VoIP session. SIP is the emerging standard in VoIP because it is simple compared to other protocols like H.323 and human-readable. The SIP interoperates successfully with multiple vendors including SNOM and Cisco

H.323 is the ITU standard for VoIP. Support for H.323 was contributed by Michael Mansous of InAccess Networks and is based on the OpenH.323 project.

While H.323 support is present in , H.323 is a dying standard. Whenever possible you should use a more modern interface likeSIP or IAX.

A codec (compressor/decompressor) is used to compress analog voice into a digital data stream or to decompress the data back into an analog signal. can operate with a wide variety of codecs an file formats. Because of its open architecture, it is easy to incorporate additional codecs or file formats.

There are two common 64 kbps PCM compression standards, micro-law and a-law. Both use logarithmic compression to effectively achieve 12 to 13 bits of linear compression in 8 bits. Logarithmic compression reduces higher volumes or frequencies exponentially. Micro-law is slightly better in compressing low level signals and has a slightly better signal-to-noise ratio. Micro-law is commonl used in North America, a-law is commonly used in Europe

Atek VoIP Solution provides seamless, transparent translation between any of the following codecs.

In addition, other codecs, such as G.723.1 and G.729 can be passed through transparently.

Note that you should use the alaw, ulaw, or linear codecs to use in-band DTMF. Note that most codecs are too lossy to support fax transmissions.

System uses files to store audio data including voicemail and music on hold. supports a wide variety of file formats for audio files. Supported formats includ

Quality of Service

Quality of Service (QoS) is the ability of a network to provide improved service to selected network traffic. QoS support is available in a variety of networking equipment. QoS tools can let you manage the end-to-end efficiency of your voice traffic. A detailed discussion of QoS is beyond the scope of this book. You can pursue this topic elswhere, including RFC3290.

QoS provides priority service to selected traffic to optimize the use of available bandwidth, control jitter and latency and improve loss characteristics. QoS tools provide control over congestion management, queue management, traffic shaping and policing, and link efficiency. This makes it easier for mission-critical applications to co-exist on a network. Optimizing QoS for one data flow should not make other data flows fail. Many routers and switches provide facilities for managing QosS

For example, you may have a small office with a DSL line. The DSL line might have 384 kbps of bandwidth bi-directionally. QoS tools would allow you to dedicate 128 kbps of the bandwidth of the DSL line specifically to telephony. This would mean there would always be bandwidth for telephone calls no matter how busy the Internet connection gets carrying other traffic.

includes many applications. These applications perform useful functions like dialing a telephone number or saving a voicemail message. These applications are described at length in the on configuration.

Two separate networks are available, the PSTN and the Internet. They each provide different services. Telephone numbers are used to address a specific device on the PSTN. IP addresses are used to address a specific device on the Internet.

Because the public telephone network is optimized for voice, it is not well suited for data transmission. Since voice can easily be digitized, the Internet is well suited to transmitting digitized voice. Because of this, the current PSTN with all its channels is growing obsolete. Over the coming years the PSTN is moving to a new IP Internet Protocol) architecture. Many telephone carriers already have a serious financial commitment to this change

Telephone calls can be routed over an IP network including the Internet. If two users are connected to , they can communicate over a data network, no telephone company i needed.

Accepting calls from users on the PSTN requires a telephone number. Telephone numbers are only hosted on the PSTN. Telephone numbers are rented from a supplier, a telephone company.

Making or receiving telephone calls from the PSTN requires a connection to the PSTN. Direct connections to the PSTN can be rented from a telephone company.

The PSTN is built with channels, for example the pair of wires that run from your phone to a phone company switch, or the channels that make up a T1 circuit. A channel provides a dedicated connectio between one telephone and another telephone for the duration of the call. Consult the title T-Carrier for an in-depth description of T1 lines and an extremely brief introduction to SONET.

When you make a telephone call over the PSTN, you consume a channel for the entire call. Only your telephone call goes over the channel. You and the called party have exclusive use of the channel for a long as the call lasts

A POTS (Plain Old Telephone Service) line has a single telephone number associated with it. Calls to that telephone number are routed over a dedicated circuit. An server connected to a POTS line can send and receive calls over that circuit.

You can rent POTS lines from a telephone company, if they are not out on strike. You can connect these POTS lines to your system. cards allow you to connect a POTS line to your server.

There may be different companies (alternate carriers) in your area that provide telephone numbers and connections. Alternate carriers often rent at least part of their network, for example the wires to you premises, from your local telephone company.

A direct connection to the PSTN can be a larger connection, for example a T-Carrier connection or some other even larger connection. cards interface with T-Carrier lines. Your telephone numbers are associated with this connection. Calls to your telephone numbers are routed to you server over the T-Carrier connection.

A T-Carrier connection provides multiple channels. A T1 line provides 24 voice channels. If you have twenty-four users in your office, and twenty-four telephone numbers, and a T1 line, every user has a available line. This means twenty-four incoming or outgoing calls can be placed concurrently.

There can be more telephone numbers, or users, than circuits. You can have more telephone numbers than T-Carrier channels. If you have fifty telephone numbers and a T1 circuit, calls to any of the fifty numbers can be sent over any of the twenty-three T1 channels to your server. The world wid telephone system has many more users than channels. That's why you get a busy signal after an emergency when everyone is trying to get a channel

The service provided with a T-Carrier line signals what number is ringing. This allows to appropriately route the incoming call.

In addition to a telephone number and connections, telephone companies provide additional services like local or long distance calling. You can usually get long distance or international calling from a variety or providers.

A new generation of telephone companies provides the best of both worlds. These companies will provide telephone numbers, and route calls over the Internet or PSTN.

You can connect to an Internet telephone company that provides a bridge to the PSTN. Instead of a connection to the PSTN, you use a connection to the Internet. A call placed to your telephone number is sent from that provider to your server over the Internet.

A T-Carrier circuit can connect to a telephone company, or to an Internet provider. T-Carrier lines connected to a telephone company use the individual channels for individual telephone calls.

Sending voice over the PSTN is expensive compared to sending voice as data over the Internet. Unlike an Internet connection, PSTN channels aren't shared.

There are a variety of ways to connect to the Internet. The following table compares some of them. Some connections are symmetrical, that is they are just as fast in both directions. Some connections like a satellite connection, are much faster in one direction, for example down from the satellite to you.