1. White Paper:
Advantages of SIP for VoIP
2. White Paper: Advantages of SIP for VoIP October 2003
Voice over Internet Protocol (VoIP) defines a way to carry voice calls over an Internet Protocol (IP)
network including the digitisation and packetization of the voice streams. The VoIP standards
enable the creation of a telephony system where higher-level features such as advanced call
routing, voice mail, contact centres, etc., can be utilised.
How does it work?
Historically, telecommunications companies have relied on what is commonly referred to as circuit-
switched technology to transport telephone calls. This technology establishes a 'permanent'
connection between the calling and the called parties for the entire duration of the call. The
problem with circuit-switched technology is that it requires a significant amount of bandwidth
dedicated to each call, and it can only support certain types of calls (i.e. telephone to telephone).
Moreover, the hardware needed to run circuit-switched networks is very expensive, due in large
part to the fact that voice and data services must be carried on different wires and thus need
separate hardware to accommodate the two types of traffic. Naturally, the traditional telephone
companies pass along the costs of building and maintaining a circuit-switched network to the
consumer in the form of higher rates for their telephone services.
As the name implies, VoIP refers to calls that traverse networks using Internet Protocol. The voice
stream is broken down into packets, compressed, and sent toward their final destination by various
routes (as opposed to establishing a 'permanent' connection for the duration of the call),
depending on the most efficient paths given network congestion, etc. At the other end, the packets
are reassembled, decompressed, and converted back into a voice stream by various hardware and
software elements, depending on the nature of the call and its final destination.
3. White Paper: Advantages of SIP for VoIP October 2003
Advantages of VoIP
• New Integrated applications: Because VoIP is digital, it may offer features and services that
are not available with a traditional phone. (See section ‘Examples of additional functionality
made possible by using SIP for VoIP’ below)
• Cost Reduction: No call tolls as it uses your Internet connection. With VoIP you can talk for as
long as you want with anyone that has an Internet connection. You can also talk with many
people at the same time without any additional cost. – Low-cost conferencing.
• Single unified network: As voice is converted into data, it is transported on the data network
and negates the need for a voice network at all.
• Open standards: VoIP embraces an open architecture (see below) and provides the flexibility
to integrate with backend systems
• User attributes move with you: As soon as you log on to any VoIP-capable device such as PC,
Mobile Phone, any IP Phone, Satellite Office system, or Home Office adapter.
VoIP systems increasingly demonstrate greater cost-effectiveness than traditional voice networks.
As VoIP technology evolves, the cost/benefit ratio, alongside efficiency and flexibility in
implementation, will continue to increase.
The following chart compares the cost of two deployment scenarios for an enterprise of 10,000
phone users, where 50 percent of employees are divided between two large locations and the
remainder among 12 branch offices. Analysis shows that replacing the current PBX with another
PBX increases the cost by approximately 48 percent. However, replacing the existing PBX with an IP
PBX saves approximately 11 percent of the overall replacement cost.
Savings apply not only to enterprises but also to consumers, because service providers can pass the
savings resulting from lower network deployment and maintenance costs on to subscribers.
4. White Paper: Advantages of SIP for VoIP October 2003
5. White Paper: Advantages of SIP for VoIP October 2003
What Nokia mobile devices will it work on?
• Symbian 7.0 supports SIP
• Nokia phones that run Symbian 7.0
• Nokia 6600
• Nokia 6620
• Nokia 9300
• Nokia 9500
• Future devices that will support VoIP and VoWLAN
• Zeus – 6830 (Public Announcement: 2nd Nov 04)
• All IP 3G networks will also run VoIP to handset
Nokia Project: Sophia
Issues with VoIP
• Network Latencies / Quality of Service support in the Internet
• Session handovers
• Reachablility through firewalls and VPN connections
• Convergence confusion
• Call routing between IP, PSTN & GSM networks
• Regulatory issues, - currently unregulated
• IETF (Internet Engineering Task Force)
The community of engineers that
standardizes the protocols that define
how the Internet and Internet
protocols work. http://www.ietf.org/
• ITU (International Telecommunications
Union) an international organization
within the United Nations System
where governments and the private
sector coordinate global telecom
networks and services.
6. White Paper: Advantages of SIP for VoIP October 2003
• H.323 - An ITU Recommendation that defines “Packet-based multimedia communications
systems”. H.323 defines a distributed architecture for creating multimedia applications,
• SIP - Defined as IETF RFC 2543. SIP defines a distributed architecture for creating multimedia
applications, including VoIP
• MGCP - Defined as IETF RFC 2705. MGCP defines a centralized architecture for creating
multimedia applications, including VoIP
• H.248 - An ITU Recommendation that defines “Gateway Control Protocol”. H.248 is the result
of a joint-collaborate with the IETF. H.248 defines a centralized architecture, and is also
known as “Megaco”
H.323 vs. SIP
H.323 is actually a suite of protocols, and incorporates many individual protocols that have been
developed for specific applications.
H.323 Protocol Suite
Video Audio Data Transport
H.261 G.711 T.122 H.225
H.263 G.722 T.124 H.235
G.723.1 T.125 H.245
G.728 T.126 H.450.1
G.729 T.127 H.450.2
Full implementation of H.323 requires a lot of overhead. SIP is a much more streamlined protocol,
developed specifically for IP telephony. Smaller and more efficient than H.323, SIP takes advantage
of existing protocols to handle certain parts of the process. For example, Media Gateway Control
Protocol (MGCP) is used by SIP to establish a gateway connecting to the PSTN system.
SIP allows two or more participants to establish a session consisting of multiple media streams
using text-based request and response messages. A user, termed a SIP endpoint, is addressed by a
SIP URL in the form of an e-mail address, such as sip:firstname.lastname@example.org or sip:email@example.com. The
application used for communication is called the user agent (UA). Call initiation and modification is
done through INVITE messages of SIP. Two endpoints can communicate with each other directly, or
they can make use of a SIP entity called the redirect server. The user first sends the request for call
initiation to this server, which queries a location service to retrieve the IP address and port of the
other user. The location service keeps track of the current location of the users.
Since the underlying elements of SIP are so much like HTTP, creating network-based services such
as time-dependent call forwarding is quick and straightforward. Developers can design and
7. White Paper: Advantages of SIP for VoIP October 2003
implement new SIP-based voice services just as quickly and easily as they develop web pages; and
by not requiring major hardware upgrades to application servers, but rather enabling new
software-based services using SIP, service providers can reduce the time associated with deploying
new features from months to days. For subscribers, this means ever-improving communications
service, plus lower initial and recurring telephone service costs.
Examples of additional functionality made possible by using SIP for VoIP
• Ringing tone and caller image is delivered within signaling (SIP transports MIME payload xxx)
• URLs can be passed within signaling, seamless email/media-on-demand integration (e.g. call
may be forwarded into rtsp URL: video mail answering service)
• Receive voicemail messages via email
• Possibility to create richer profiles "If caller is Bob, send soccer_results.html file to him (in SIP
payload or by mail)"
• Simple scripts: "If the time is past 4 p.m. and caller is boss, forward to voice mail."
• Another form of SIP extension work is to define the usage of SIP in new context. For example,
there is on-going work in IETF to use SIP to control networked appliances (e.g. “turn the lamp
on”). New method (DO) has been proposed for this idea. Similar ideas (using SIP for new kind of
services) will probably follow.
SIP is generic protocol for every IP capable access networks. There lies the opportunity for Nokia to
win more business in fast growing industry.
SIP and HTTP form a powerful architecture for application development community. The snowball
effect grows the market as already existing developer community finds it easy to implement
compelling services for terminal users.
Cisco has broad support for SIP across its entire product line:
• IOS Gateways: 1751, 7200, 2600, 36x0,
AS5300, AS5350, AS5400 Series,
• Call Agents/Soft switches: BTS 10200,
• Endpoints: ATA 186, 7940, 7960
• Infrastructure: Cisco SIP Proxy Server,
PIX Firewall, IOS NAT
Cisco-driven solutions are being deployed
today with Cisco SIP Global Long Distance
having been deployed by several carriers
including some with integrated H.323
8. White Paper: Advantages of SIP for VoIP October 2003
Cisco-backed SIP IP Business Solutions are starting to emerge. SIP voice application solutions have
been deployed by 5+ carriers (including Windows Messenger PC-to-Phone support)
Microsoft Live Communication Server 2003
For years Microsoft has had computer telephony features embedded in its operating systems in
products such as NetMeeting, an H.323-videoconferencing application, and Exchange Conferencing
Server, for managing data, voice and videoconferencing.
Windows XP brought along Microsoft Windows Messenger, which turned some heads in the
telecom community for its use of SIP.
Speculation about what Microsoft will do in telephony grew when company representatives began
showing up at industry events such as Voice on the Net (VON), and later when it announced
development of its Real Time Communication (RTC) Server, code-named Greenwich. RTC is renamed
Live Communication Server 2003. The server acts as a control node for managing conferences
among SIP clients.
Windows CE 5.0 includes a greater VoIP focus so that IP phone
and other device makers, as well as service providers, can
expand their VoIP applications.
The Windows CE 5.0 SIP stack is interoperable with Microsoft
Windows XP and Microsoft Live Communications Server,
enabling collaboration and communication through instant
messaging and providing status information (presence)
between desktop computers and Windows CE–based IP phones.
In addition, Microsoft Visual Studio .NET for native Microsoft
Win32–based applications, or for managed applications built
on the Microsoft .NET Compact Framework, provides an easy-to-
use tool to efficiently develop feature-rich IP phones and VoIP
Beyond the new feature, over the past 12 months Microsoft has
focused on expanding its VoIP partners for Windows CE. At VON,
it announced 13 new VoIP manufacturer partners focused on embedding Windows CE into IP
phones and other devices as well as a host of new system integrators.
See the following resources for more information:
• RFC 3261: SIP: Session Initiation Protocol:
• SIP Extensions for Presence:
9. White Paper: Advantages of SIP for VoIP October 2003
• Nokia Sofia IP Telephony Software suite:
• Microsoft Real-Time Communications: Protocols and Technologies:
• Cisco SIP: