SIP: Redefining IP Voice Communications
BY DR. HENRY SINNREICH
Global IP communications over the Internet is probably the most
significant development since the emergence of telephone networks over 120
years ago. New communication services for text, voice, and video, as well
as mobility and presence-based services, enabled by the Internet, are
leveraging a relatively new protocol known as SIP, or Session Initiation
Protocol. SIP is an IP-based communications protocol that provides a
network-based convergence solution. SIP is an open, non-proprietary
standard modeled after HTTP that seamlessly integrates with the Internet.
Because of its easy integration and ability to inter-operate with other
protocols and vendors, SIP is the foundation for flexible and scalable
VoIP solutions. This article will focus only on voice, but we suggest you
keep in mind the other IP communication services enabled by SIP as well.
Present telecommunication networks and services provide adequate and
universal telephone service at affordable rates. So, why would enterprises
want to transition from their traditional circuit-switched networks to
VoIP? Good question. An enterprise has two major incentives to complement
existing services or migrate to IP communications. The first incentive is
the potential short-term savings by moving to end-to-end VoIP and mixed
IP-PSTN and also by moving PBX voice to VoIP usage. The second incentive,
and motivation for this paper, is increasing overall productivity through
the integration of voice, data, and productivity applications. Examples of
integrated IP communications based on SIP (presence, instant messaging,
voice, video and application sharing), with applications are Microsoft Office 2003 and other
similar or related products, such as Siemens
Existing VoIP Protocols and SIP
A large number of VoIP protocols are in use at present, reflecting their
origin or the voice strategy of particular vendors. H.323 was introduced
by the ITU-T in the late 1990s, and it complements H.320, H.321, H.322,
and H.324, which are for audio/video conferencing using analog, ISDN, ATM,
and the public telephony networks, respectively. H.323 was thus mainly
modeled by ISDN and modified to work on IP LANs. It was used later by
vendors for IP LANs and WANs.
The circuit switch vendors applied the central control model as used in
PBXs for servers to control desktop phones over the IP LANs and the result
were several master-slave protocols, such as MGCP and MEGACO/H.248. The
central controllers used by such master-slave protocols are also referred
to as ï¿½softswitches.ï¿½
In contrast, SIP is a peer-to-peer protocol, while the softswitch
protocols and even H.323 presume some central control. SIP can use servers
for network-based services, but the user can choose whether to use them,
similar to pointing a browser to any Web server of choice for the home
As for the many vendor proprietary protocols, such as used by IPBX
vendors, they carry all the disadvantages of vendor lock-in of the
customers into proprietary servers and IP phones, for which in most
instances there is no benefit from competition.
Distributed Networking Applied to Telephony Services
The architectural principles of the Internet -- avoiding single points of
failure, avoiding single paths, avoiding state in the network, and
end-to-end control -- have proven to support more reliable communications
than circuit-switched networks, in spite of their ï¿½carrier strengthï¿½
network elements. Internet service has degraded somewhat with recent
widespread virus attacks, but has never failed and has been proven so far
to meet design expectations.
SIP-enabled networks are also distributed by design and have most state
and control pushed to the edge of the network, in compliance with Internet
architecture principles. Internet resilience is based on IP endpoints --
autonomous hosts that communicate over any available path and do not
depend on the network for state. This is a fundamental difference from
PSTN systems where a central authority activates and shuts down all
processes in various boxes having a stake in the communication and where
all systems in a path have to keep state for every call going through.
Higher-than-PSTN resilience for VoIP can be obtained if the underlying
customer network is multi-homed to the Internet using truly diverse
geographical paths and by deploying distributed SIP servers and
distributed functionality for all SIP components, such as application
servers, media servers, SIP-PSTN gateways, etc. This may not be the case
however in many deployments, where lowest cost is the dominant factor.
Events such as the 9/11 attacks in New York and the East Coast blackouts
have confirmed however that VoIP deployed at the edge, is more resilient
than the telecom infrastructure -- provided there is backup power on the
How SIP Improves Upon PSTN Functionality
Aside from the significant voice traffic migration to mobile telephony,
the advent of the Internet and its associated technologies are forcing
telecommunication companies and their vendors to face the facts that
telecommunication networks have a competitor in the Internet and the
deployed PSTN technology is obsolete.
In the context of VoIP, a closed system cannot be expanded with components
from competing vendors using public standards. Examples of closed systems
are softswitches and proprietary IP PBXs. In such systems, it is not
possible to choose the following from different sources: IP phones;
IP-PSTN gateways; service controllers; SIP servers; IVRs, VoiceXML and
speech recognition systems; media and announcement servers; conference
bridges; voice mail and unified messaging; and other components.
SIP, however, brings choice and several enhancements to traditional
telephony; including call transfer in voice mode, automatic callback
feature, multiparty conference calls, Web-sharing and data collaboration,
and mobility. Visual voice mail alerts also falls into this category. The
technology transforms e-mail messages into an MP3 or .Wav file, which can
then be ï¿½viewedï¿½ (listened to) as an e-mail attachment on a secure Web
Innovative IP Communications Based on SIP
Using the development of the Internet communication protocols and SIP, IP
communications enable radically new capabilities and applications.
Probably the most far-reaching disruptive engineering decision of IP
communications is to integrate voice and all other media (text, voice,
video games, etc) with the Web with regards to addressing, protocols, and
SIP-based communications functionality will drive improved productivity by
making it easier and more efficient for end-users to communicate. At the
same time, SIP-based communications enhance flexibility by providing
enterprises a variety of platforms, endpoints, and servers to utilize.
The Internet development community, for example, prefers text-based
messages for easy code development and debugging. SIP uses text-based
messages for simplicity and easy troubleshooting. In addition to working
with standard telephone numbers and extensions, SIP can also use
e-mail-like addresses. These addresses can be imbedded in documents and
Web pages for ï¿½click-to-dialï¿½ applications. For example, SIP addresses
can look like sip:[email protected] or sip:[email protected].
Presence-Based Communications and Instant Messaging
Presence will be a major new communications capability that is not
available in circuit-switched telecom networks. A simple example of
ï¿½presenceï¿½ is an instant message (IM) clientï¿½s ï¿½buddy list,ï¿½
which lists the userï¿½s ï¿½buddiesï¿½ and their current state -- online
or offline. Additional state information, such as whether they are
currently active or idle and whether they are currently typing a message
response or not, is also provided using presence.
Presence can be used for such services as:
ï¿½ Make ï¿½politeï¿½ calls, only when you see an encouraging or
ï¿½ Avoid phone tag during busy hours;
ï¿½ Automatic call-back on presence;
ï¿½ Ad-hoc conference calls based on presence;
ï¿½ Avoid waiting for call center agents -- replace ACD (Automatic Call
Distributor) with agent presence showing also the length of the queue;
ï¿½ On-the-air presence for mobile phones;
ï¿½ Presence coupled with location.
The following scenario would only occur in a SIP-enabled environment: A
mobile user or a user on the PSTN is trying to call a user that has an IP
communication service and can be reached at a SIP phone and/or IP devices.
The call is first routed by the PSTN gateway to the SIP server for the
called party and from there to the SIP phone. If the called party does not
answer, the SIP servers will proxy the call to the unified messaging (UM)
server. The caller can now leave a voice message. The notification for
ï¿½Message Waitingï¿½ will appear as a flashing light on the SIP phone (as
is usual on PBX and mobile phones), as an e-mail notification and also on
a UM Web page.
Making a VoIP Investment
One U.S.-based company that benefited from SIP-based communications,
specifically VoIP, is Storage Area Networks, Inc. (SANZ).
SANZ was finding that operating separate data and voice platforms was
costing a great amount of time and money as newly acquired companies had
to be added to their enterprise network. The company also spent a
considerable amount on long-distance charges between satellite offices.
Given that, SANZ searched for an alternative technology solution that
would reduce costs, simplify voice and network services deployment and
support the companyï¿½s technology needs.
To face its challenges, SANZ selected a network-based IP communications
service to streamline their needs over a single data network. In doing so,
the company consolidated local, long-distance, data, and other network
services through just one connection. They quickly found that a
significant advantage of VoIP is the ability to avoid the toll charges of
ordinary telephone service; In addition, the technology prioritizes voice
packets over data, ensuring optimal quality of service.
With that challenge surmounted, SANZ is now focusing on what it really
wants to do, that is cultivating its business through strategic
As voice is now viable over IP, the advantages of using the global IP
network for converged voice and data services are just beginning to reach
the marketplace. Furthermore, responsible IT managers will invest in new
services that reduce high operational costs associated with existing
services, such as PSTN/PBX telephony. By deploying IP voice communications
and rich end-to-end controlled IP, IT managers will quickly realize
implementation cost is negligible compared to the huge investments and
operational costs required by traditional PBXs, by non-standards-based
(not SIP compliant) IPBXs and by PSTNs. Last but not least, there is a
wide selection of standards compliant vendor equipments to choose from.
Dr. Henry Sinnreich is a Distinguished Member of Engineering at MCI where his responsibilities include the
architecture of IP communication services, SIP telephony devices,
wireless, mobility for IP communications and quality of service as well as
network support for communications. MCI owns, operates, monitors, and
maintains one of the largest communications networks in the world. The
companyï¿½s network facilities span the globe in more than 125 countries
and over 2,800 cities.
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