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Comparison of H323 Vs SIP Protocols Used in VoIP and IP Telephony

Comparison H323 and sip

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Nagendra Chaudhary
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15 views

Comparison of H323 Vs SIP Protocols Used in VoIP and IP Telephony

Comparison H323 and sip

Uploaded by

Nagendra Chaudhary
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as DOCX, PDF, TXT or read online on Scribd
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Comparison of H323 vs SIP Protocols

Used in VoIP and IP Telephony


Written By Lazaros Agapidis

Voice over IP (VoIP) is a technology that allows you to transmit voice over
an IP network. This enables both telephony and data to be transmitted
over the same network packet-based infrastructure.

There are two fundamental protocols used to transmit the signaling that is
necessary to make IP telephony operate correctly. These protocols are
Session Initiation Protocol (SIP) and H.323.

These are two very different protocols that emerged from very different
beginnings. In this article, we’ll examine them more closely to understand
how they operate and where they are typically employed.

So, let’s discuss and compare SIP vs H323 starting with a high-level
comparison table:

H323 vs SIP – Comparison Table


The following table compares the H.323 and SIP protocols:

Description SIP H.323

Origins IETF Internet based ITU-T based on ISDN


Routing architecture Flat Hierarchical

Gatekeeper, optional, but nece


Control Systems SIP server/IP PBX, mandatory
larger deployments

Endpoint addressing Uses SIP URLs Uses aliases mapped by Gate

Natively compatible with IP and the


Yes No
Internet

Design Modular flexible Monolithic, inflexible

None provided beyond voi


Additional features Instant messaging, presence
video

Not readily interoperable but can be


Interoperability with traditional
implemented with appropriate Backward compatible
telephony
adaptors/voice gateways

VoIP, IP Telephony, and Signaling


In order to understand SIP and H.323, as well as their differences and
operation, let’s take a brief look at what VoIP is and what role it plays in IP
telephony.

VoIP

VoIP is a set of technologies and methodologies that digitizes and


packetizes voice at the source device and prepares it to be sent over the
network in the form of IP packets.

These packets are received and reassembled at the destination device and
the original voice is reproduced and heard by the receiver.

IP telephony leverages VoIP to enable users to call each other using the
well-known process of picking up the handset of a phone and dialing a
number.
Signaling

Like traditional telephony, IP telephony requires signaling mechanisms.


Signaling is involved in initiating, maintaining, modifying, and terminating a
call.

When you pick up the handset of a phone, the phone goes off-hook which
sends a signal. Dialing numbers send signaling to the server responsible
for routing telephone calls.

Functions such as making a phone ring, hearing ringtone, dialing a number,


implementing call display, call waiting, call hold, and various other
advanced telephony features, all use signaling to successfully operate.

MORE READING: Cisco UC560 Dial Plan for Voice Mail Configuration Example

Signaling is achieved in conventional telephony using a physically separate


data channel. For VoIP and IP telephony, signaling is achieved using
protocols such as SIP and H.323 that create communication sessions
between devices that are separate and distinct from the actual exchange of
voice packets.

Session Initiation Protocol (SIP)


History

SIP was conceived quite early on in 1996 and by 1999 it was published as
a standard by the Internet Engineering Task Force (IETF) in RFC 2543.
Image Source

The goal of its founding developers and of the standardization organization


that has adopted it since has been to provide a signaling and call setup
protocol for IP-based communications that can mimic and reproduce the
call processing functions and features of the Public Switched Telephone
Network (PSTN).

At the same time, SIP was designed to be extendable to support additional


multimedia services such as video conferencing, and media streaming as
well as specialized functionalities that include presence, instant messaging,
file transfer, fax over IP, and even online gaming.

Born out of the Internet

Unlike other telephony protocols, SIP is lauded by its proponents for having
roots in the Internet community rather than the telephony industry.

This is demonstrated by the fact that SIP has been standardized by the
IETF whereas other voice protocols such as H.323 and ISDN have been
traditionally associated with the International Telecommunications Union
(ITU).
SIP, as its name suggests, is involved in the control mechanisms related to
the initiation and termination of sessions needed to allow voice and video
applications to function.

It defines the format of the control messages (and once again, not the voice
packets) transmitted between participants in a media exchange.

Call setup, call teardown, and Dual Tone Multi-Frequency (DTMF) signals
are just some of the call control messages that SIP transmits.

These are among the features that have been employed in traditional
telephony for decades that SIP essentially duplicates within the VoIP
domain. SIP was designed to mimic the functionality of the PSTN and
conventional PBXs to avoid the need of retraining users when moving from
conventional to IP telephony.

The goal was to allow a user to use a SIP-enabled telephone without any
change in the tones, functionality, and general feel of the calling experience
that users have become so familiar with over the years.

An advanced protocol

Even so, SIP was designed to be modular and flexible, to be able to deliver
much more than what the traditional PSTN offered.

MORE READING: SIP Trunking With Call Manager Express

This modularity allows sip to continually be developed to incorporate


advanced features and functionalities that take advantage of the IP
infrastructure upon which SIP is based.

VoIP systems based on SIP can easily expand VoIP network services by
adding video and mobile users to their existing infrastructure with very little
intervention into the existing system.

This increases the options an organization is provided with, as the addition


of features is often implemented by simply obtaining a license, a software
package, or a system server, depending on the type of feature in question.

H.323 VoIP Protocol Suite


History
The most popular alternative signaling protocol to SIP is H.323 which was
developed by the ITU-Telecom Standardization Sector (ITU-T).

Image Source

Although it can be used for strictly voice conversations, it is most often


applied today in video conferencing equipment and leverages the Q.931
standard, which defines legacy ISDN circuits.

The original intention was to allow video conferencing systems to use ISDN
infrastructure which seemed to be a very promising technology at the time,
although it has since been adapted to run over IP networks as well.

Protocol Suite

H.323 is more properly referred to as a system specification or standard


that includes various protocols providing multiple services. These protocols
are further described below:

H.225.0 Call Signaling – This is essentially SIP’s counterpart as it is the


fundamental signaling protocol in the H.323 suite.

H.245 Control protocol – This protocol standardizes the methodology of


the exchange of capability information between endpoints and opens and
closes logical channels for voice and video.

H.225.0 Registration, Admission and Status (RAS) – This feature is


unique to H.323 in that it doesn’t have a counterpart in a SIP environment.
Where SIP is highly flat in its architecture, the H.225.0 RAS protocol
provides a hierarchical structure to call signaling, with a device called a
Gatekeeper at its center. This is especially useful for large enterprises with
multiple campuses, locations, and branch offices requiring a centralized
and interconnected telephony network.

Conclusion
As all communications systems are slowly converging towards leveraging a
single IP-based communication infrastructure, SIP is quickly becoming the
protocol of choice.

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