Inquire About Our Services or
CALL (410) 999-9350
For More Information!

What is IAX Termination?

IAX is the Inter-Asterisk™ eXchange protocol used by Asterisk™ , an open source PBX server from Digium. It is used to enable VoIP connections between Asterisk™ servers, and between servers and clients that also use the IAX protocol. IAX now most commonly refers to IAX2, the second version of the IAX protocol. The original IAX protocol has been deprecated almost universally in favor of IAX2. IAX2 is very robust and full-featured yet simple as far as protocols go. It is agnostic to codecs and number of streams, meaning that it can be used as a transport for virtually any type of data. IAX2 uses a single UDP data stream (usually on port 4569) to communicate between endpoints, both for signaling and data. The voice traffic is transmitted in-band, making IAX2 easier to firewall and more likely to work behind NAT. (This is in contrast to SIP, which uses an out-of-band RTP stream to deliver information.)

IAX2 supports trunking, wherein a single link carries data and signaling for multiple channels. When trunking, data from multiple calls are merged into a single set of packets, meaning that one IP datagram can deliver information for more than one call, reducing the effective IP overhead without creating additional latency. This is a big advantage for VoIP users, where IP headers are large percentage of the bandwidth usage.

The IAX2 Protocol or Inter-Asterisk™ Exchange Protocol was created by Mark Spencer for Asterisk™ for VoIP signaling. The protocol sets up internal sessions and these sessions can use whichever codec they want for voice transmission. The Inter-Asterisk™ Exchange protocol essentially provides control and transmission of streaming media over IP (Internet Protocol) networks. IAX is extremely flexible and can be used with any type of streaming media including video however it is mainly designed for control of IP voice calls. IAX’s design was based on many common control and transmission standards today including Session Initiation Protocol (SIP, which is the most common), Media Gateway Control Protocol (MGCP) and Real-time Transfer Protocol (RTP).

The Primary goals for IAX was to minimize bandwidth used in media transmissions with particular attention drawn to control and individual voice calls and to provide native support for NAT (Network Address Translation) transparency. The basic structure of IAX is that it multiplexes signaling and multiple media streams over a single UDP (user datagram protocol) stream between two computers. IAX is a binary protocol and is designed and organized in a manner to reduce overhead especially in regards to voice streams.

Asterisk™

Asterisk™ is a complete PBX in software. It runs on Linux, BSD and MacOSX and provides all of the features you would expect from a PBX and more. Asterisk™ does voice over IP in many protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware.

Asterisk™ provides Voicemail services with Directory, Call Conferencing, IVR (Interactive Voice Response) and Call Queuing. It has support for three-way calling, Caller ID services, ADSI, SIP and H.323 (as both client and gateway). Check the Features section for a more complete list.

Asterisk™ needs no additional hardware for Voice over IP. For interconnection with digital and analog telephony equipment, Asterisk™ supports a number of hardware devices, most notably all of the hardware manufactured by Asterisk™ 's sponsors, Digium™. Digium has single and quad span T1 and E1 interfaces for interconnection to PRI (Primary Rate Interface) lines and channel banks as well as a single port FXO (Foreign Exchange Office) card and a one to four-port modular FXS (Foreign Exchange Station) and FXO card.

Asterisk™ supports a wide range of TDM protocols for the handling and transmission of voice over traditional telephony interfaces. Asterisk™ supports US and European standard signalling types used in standard business phone systems, allowing it to bridge between next generation voice-data integrated networks and existing infrastructure. Asterisk™ not only supports traditional phone equipment, it enhances them with additional capabilities.

Using the Inter-Asterisk™ eXchange (IAX™) Voice over IP protocol, Asterisk™ merges voice and data traffic seamlessly across disparate networks. While using Packet Voice, it is possible to send data such as URL information and images in-line with voice traffic, allowing advanced integration of information.

Asterisk™ provides a central switching core, with four APIs (Application Programming Interfaces) for modular loading of telephony applications, hardware interfaces, file format handling, and codecs. It allows for transparent switching between all supported interfaces, allowing it to tie together a diverse mixture of telephony systems into a single switching network.

Asterisk™ is a Trademarked Property of Digium™.

  Home  |  VoIP for Call Centers  |  BinFone Services  |  VoIP Termination  |  Local and Toll Free Numbers
Hosting and Colocation  |  Long Distance  |  Check Out Our Rates  |  International A to Z Rates  |  Sign Up Now
IAX Termination  |  SIP Termination  |  FAQ  |  Contact Us for a Quotation  |  Usage Policy
All Content ©2005-2006 by BinFone Telecom.