|| August 29th, 2010

060300273 Hidalgo Arellano Salvador Indra

IT3437 Network Selected Topics Homework:2 Team: 2

Asterisk and VoIP

Asterisk is a free, open source software implementation of a telephone private branch

exchange (PBX) originally created in 1999 by Mark Spencer of Digium.

Like any PBX, it allows attached telephones to make calls to one another, and to connect to
other telephone services including the public switched telephone network (PSTN)
and Voice over Internet Protocol (VoIP) services. Its name comes from the asterisk symbol,
“*”.
Asterisk is released under a dual license model, using the GNU General Public
License (GPL) as a free software license and a proprietary software license to permit
licensees to distribute proprietary, unpublished system components.
Originally designed for Linux, Asterisk now also runs on a variety of different operating
systems including NetBSD, OpenBSD, FreeBSD,Mac OS X, and Solaris. A port
to Microsoft Windows is known as AsteriskWin32. 1

Asterisk protocols
Natively supported VOIP protocols: 

 SIP, Session initiation protocol

 IAX, IAX2 - Inter Asterisk Exchange protocol (open-source trunking protocol
primarily used by Asterisk)
 MGCP: Media gateway control protocol, IETF 
 H.323 (With the addition of OpenH323 )

There are third-party addons that greatly enhance this list, for example alternative H.323
channels, SCCP/Skinny but also for example OSP. 

SIP
SIP, the session initiation protocol, is the IETF protocol for VOIP and other text and
multimedia sessions, like instant messaging, video, online games and other services. 

SIP can be regarded as the enabler protocol for telephony and voice over IP (VoIP)
services. The following features of SIP play a major role in the enablement of IP telephony
and VoIP: 
 Name Translation and User Location: Ensuring that the call reaches the called party
wherever they are located. Carrying out any mapping of descriptive information to
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location information. Ensuring that details of the nature of the call (Session) are
supported.
 Feature Negotiation: This allows the group involved in a call (this may be a multi-
party call) to agree on the features supported � recognizing that not all the
parties can support the same level of features. For example video may or may not be
supported; as any form of MIME type is supported by SIP, there is plenty of scope
for negotiation.
 Call Participant Management: During a call a participant can bring other users onto
the call or cancel connections to other users. In addition, users could be transferred
or placed on hold.
 Call feature changes: A user should be able to change the call characteristics during
the course of the call. For example, a call may have been set up as 'voice-only', but
in the course of the call, the users may need to enable a video function. A third party
joining a call may require different features to be enabled in order to participate in
the call
 Media negotiation: The inherent SIP mechanisms that enable negotiation of the
media used in a call, enable selection of the appropriate codec for establishing a call
between the various devices. This way, less advanced devices can participate in the
call, provided the appropriate codec is selected.

Inter-Asterisk eXchange

IAX is the Inter-Asterisk eXchange protocol native to Asterisk PBX and supported by a

number of other softswitches and PBXs. It is used for enabling VoIP connections between
servers beside client–server communication.

IAX now most commonly refers to IAX2, the second version of the IAX protocol. The
original IAX protocol has been deprecated in favor of IAX2.
The IAX2 protocol was published as an informational (non-standards-track) RFC 5456 by
discretion of the RFC Editor in February 2010.

IAX2 supports trunking, multiplexing channels over a single link. When trunking, data

from multiple calls are merged into a single stream of packets between two endpoints,
reducing the IP overhead without creating additional latency. This is advantageous
in VoIP transmissions, in which IP headers use a large percentage of bandwidth.

MGCP

MGCP is an implementation of the Media Gateway Control Protocol architecture[1] for

controlling media gateways on Internet Protocol (IP) networks and the public switched
telephone network (PSTN). The general base architecture and programming interface is
described in RFC 2805 and the current specific MGCP definition is RFC
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3435 (obsoleted RFC 2705). It is a successor to the Simple Gateway Control
Protocol (SGCP).

MGCP is a signalling and call control protocol used within Voice over IP (VoIP) systems

that typically interoperate with the public switched telephone network (PSTN). As such it
implements a PSTN-over-IP model with the power of the network residing in a call control
center (softswitch, similar to the central office of the PSTN) and the endpoints being "low-
intelligence" devices, mostly simply executing control commands. The protocol represents
a decomposition of other VoIP models, such as H.323, in which the media gateways (e.g.,
H.323's gatekeeper) have higher levels of signalling intelligence.

H.323
H.323 is an ITU VOIP protocol. It was created at about the same time as SIP, but was more
widely adopted and deployed earlier. Today, most of the world's VoIP traffic is carried over
H.323 networks, with billions of minutes of traffic being carried every month. 

H.323's strengths lie in its ability to serve in a variey of roles, including multimedia
communication (voice, video, and data conferencing), as well as applications where
interworking with the PSTN is vital. H.323 was designed from the outset with multimedia
communications over IP networks in mind, making it the perfect solution for real-time
multimedia communication over packet-based networks. 

http://en.wikipedia.org/wiki/Asterisk_(PBX)
http://www.voip-info.org/wiki/view/SIP
http://en.wikipedia.org/wiki/Inter-Asterisk_eXchange
http://en.wikipedia.org/wiki/Media_Gateway_Control_Protocol_(MGCP)
http://www.voip-info.org/wiki/view/H.323