Since the birth of man, he has been moving from one age to the other, climbing up the ladder of success rung by rung. Man has started his life by hunting for everything. In his pursuits, the flame of knowledge began to quiver and thus burst into the most violent volcano. From the Stone Age till the modern era, man has practically swirling the entire world around him, (not moving himself). Since the break of dawn on the twenty-first century, technology has been man's new mantra. He thinks it and breathes it. Every set of goals is judged by technology.
Technology is one of man's greatest weapons. Many fear it, whereas many caress it. Some use it abominably whereas some require it as a necessity. In this day, technology zooms and buzzes around our heads, making us "King of the mountain". Many of our lives are in luxury because technology blesses us with its presence in our homes and offices and other business fields. It all started with a pair of wool or skins of the cattle. From that it went into an inferno of light and electricity. Then the telephone was the next. And from this point our topic starts.
Telephone and its uses are common to all. This device was used for the purpose of talking to family, friends and other loved ones. This was further worked on until it was transformed into a teeny tiny device, known as "Cellular phone". The discovery of "Internet"- A blessing in disguise or another hidden horror- caused a whirlpool to sweep the public of its feet. Internet and phone combined to give off, God knows, millions of new deices like IP-PBX, (Internet Protocol -Private Branch Exchange).
However, it is in man's nature not to be content with what he has. So he has taken up to keep on modifying his ultimatums. In this case he stripped of IP-PBX's joy and created Session Initiation Protocol (SIP). The purpose of this essay is to make sure to differentiate between both of them and to give a view on which is better.
With the progressive growth of technology the worn out PBX systems have been moved over by the new and more work efficient private branch exchanges known as IP PBX. As the name suggests the IP PBX works on the method of both VoIP and PBX . This system shifts calls over from the internet to the local lines. The joining of external phone lines helps in accommodating more users in a quicker and effective way. The IP PBX serves somewhere between the VoIP user and the more traditional user or sometimes between 2 local users similar to how a normal PBX works. The IP PBX works together with the voice signals through the IP, adding considerable advantages of the IP telephony to this particular system.
The IP PBX can work both as a physical system as well as a virtual system. The physical system joins cards with FXO/FXS ports to work, while the virtual system is known as more of a software solution and it also works with soft phones and VoIP gateways, thus lowering the overall cost without compromising over the call quality. Complete functions of a traditional PBX are carried out by the IP PBX with additional special and advanced functions that are only possible via the IP system.
The most notable advantage of the IP PBX is its flexibility and easy compatibility. The system is "redundant"; this basically means in a way it has a backward compatibility. The latest systems use the old TDM technology which has the advanced messaging platform. This allows the system to have more advanced functions like voicemail, conferencing, call forwarding etc. The system proves to be very flexible as moving of the system or addition of new users can be done very easily.
There are two major types of the IP PBX; the one which is hosted and the non hosted IP PBX. The hosted IP PBX implies that only basic VoIP accessories are required by the actual company and the main equipment is hosted by a separate company. The hosted system means we just have to install the software which is being provided by the company hosting our PBX. This reduces the overall cost of equipment and maintenance. The non-hosted PBX system includes buying of all the required equipments e.g. the PBX device used for signal conversion as well as the gateways and software required to handle calls effectively.
An IP (Internet Protocol) PBX (Private branch exchange) is simply a business telephone system which is designed to deliver voice or video using a data network and interpret with the normal Public Switched Telephone Network (PSTN).
VoIP (Voice over Internet Protocol) gateways can be used together with traditional PBX functionality which gives businesses the ability to use their managed intranet to help lessen long distance expenses, enjoy the advantages of a single network for voice and data as well as advanced CTI features or use a pure IP system which mostly gives higher cost savings, increased redundancy and greater mobility.
An IP-PBX can exist as hardware, or virtually, in the form of a software system. Because a large part of IP PBX functional ability is present as software, it is relatively cheap to add additional functionality, such as conferencing, XML-RPC control of live calls, Interactive voice response (IVR), Public switched telephone network (PSTN) interconnection ability with both analog and digital circuits, Voice over IP protocols including SIP, Inter-Asterisk exchange, Jingle (extension of XMPP protocol introduced by Google Talk) and others.
The IP PBX is a computer, simple and easy. For most basic PBX's this does not even have to be a powerful computer, as the computing requirements are very simple. The PBX runs software that carries out all of the telephone "magic." This involves direct inward dialing, call forwarding, call parking, voice mail, etc. Some of the software available is expensive. Other softwares are what are known as "open-source," in which it is not developed by a single individual or a single company, but by volunteers throughout the world making different additions and changes. These usually run on Linux which is a well-known and well-thought of open-source operating system. The PBX is connected by network cabling to the POE Switch. A switch is basically a device that transfers the network packets from one computer to the other computer. Almost all networks have a switch at their core. A POE (Power Over Ethernet) Switch is used when using IP PBX's because they can provide power for peripherals through the network cabling. A POE switch is not always necessary, but without one, you have to have external power supplies. The POE switch does away with the need for these "power bricks," automatically reducing the office clutter, and increasing usage.
IP telephones are special telephones that make use of network cabling. They are actually small computers themselves. Standard analog phones can not be used as IP phones, though they can be linked to the IP PBX with the help of an ATA. Proprietary digital phones coming from proprietary PBX's cannot be used. Many IP telephones have built-in two-port switches, so that the desktop computer can share one network connection with the phone, often getting rid of the need for additional cabling. IP phones are available from all of the good telephone manufacturing companies: Aastra, Polycom, Cisco, Avaya, 3Com, etc.
"Soft-phones" are also available. A soft-phone is actually a software that runs on a computer, and carries out all of the functions of an IP phone. They require headsets, or computer speakers and microphones joined to the computer.
Following are the principal characteristics of IP PBX system:
Switching
System
Class 5 features
Billing
Management Interface
Web Interface
IP IVR
Soft-Phone
Resellers
Callback
Call Shop
E-shop
SMS Module
PBX hosted Platform Module
Unified Messaging
There are three different types of IP PBX structures. They are listed below along with each of their limitations;
IP
An IP architecture does all of its switching in the IP world which literally means no transcoding between the TDM bus and IP bus. Customers are benefited from this type of architecture because it is scalable and additional hardware is not necessary to do the digital signal processing (DSP) to convert the media. Examples of this architecture are 3Com NBX, 3Com VCX, Asterisk, and Cisco Call Manager.
Hybrid IP
Hybrid IP architecture has to convert its media between the IP and TDM world. This means increased resources are required to transcode. Usually these types of systems have limitations of the number of IP phones that can be used.Examples are Avaya Communications Manager, Avaya IP Office, Nortel BCM, and Nortel Succession.
Legacy TDM switch
There is no hassle for converting media between IP and the TDM bus because this is a TDM solution. These are older type systems. Examples are Avaya Definity, Nortel Norstar Avaya Partner, Nortel Meridian.
However, sometimes IP-PBX can depend on the Operating System. These are of three types;
Linux
Linux provides a solid space for IP PBX platforms. While these systems are usually quite reliable, the installation and training costs are often considerably higher than other systems. Examples which use Linux include 3Com VCX, Alcatel Omni and Asterisk.
Windows
Windows allows products to easily interact with other applications. These types of systems are usually less reliable and security maintenance is often a big issue. Examples which use Windows include Cisco Call Manager, Nortel BCM, Siemens HiCom, Siemens HiPath, Toshiba Strata CS, Vertical Communications Instant Office, and Vertical Communications TeleVantage.
Real-time
Real-time operating systems are very reliable for call completions. Often separate application servers are necessary for other solutions. Since the applications are not present on the PBX, there is not much opportunity for an errant program to disturb the call processing. Examples which use this type of OS include VxWorks - 3Com NBX, Nortel Succession, Nortel Norstar, proprietary OS - Avaya Definity, Nortel Meridian.
Session Initiation Protocol (SIP)
The Session Initiation Protocol (SIP) is basically an Internet Engineering Task Force (IETF) standard call controlling protocol, it is based on research done at Columbia University by Henning Schulzrinne and his team. The very first SIP RFC, number 2543, got published in 1999. Since then, a lot of work has been done, and various RFCs have been published to increase and extend SIP capabilities.
SIP is designed to give access to signaling and session management for voice and multimedia interlinks over packet-based networks. It is a person-to-person protocol with brilliant end points and spread out call control, like H.323. Gateways that utilize SIP are not dependent on a call agent, although the protocol does define many functional entities that allow SIP endpoints localize each other and arrange a session.
SIP was structured as one module in an IP communications solution. This modular architecture allows it to integrate with and make use of the services of other already existing protocols, like Session Description Protocol (SDP), Real-Time Transport Protocol (RTP), Resource Reservation Protocol (RSVP), RADIUS, and Lightweight Directory Access Protocol (LDAP). SIP usually uses User Datagram Protocol (UDP) as its primary transport protocol, but it can also use TCP. The default SIP port for both TCP or UDP is 5060. To provide more security, Transport Layer Security (TLS) support has been included beginning with Cisco IOS Software Release 12.3(14)T. SIP specifications do not include all the aspects of a call, as done by H.323. Instead, its job is to make, modify, and end sessions between applications, no matter what the media type or application function may be. The session can vary from just a two-party phone call to a multiuser, multimedia conference or an interactive gaming session. SIP does not lay out the type of session, just its management.
One of the unique parts of SIP is the concept of presence. The public switched telephone network (PSTN) can provide basic presence information-whether a phone is on- or off- hook-when a call is initiated. However, SIP takes that beyond. It can give access to information on the agreement of the other party to get calls, not just the ability, before the call is even attempted. This is quite similar in concept to instant messaging applications-you can choose which users appear on your list, and they can choose to display many different status types, like offline, busy, and so on. Users who subscribe to that instant messaging service are aware of the availability of the people on their list before they try to contact them. With SIP, you can collect presence information from many devices, such as cell phones, SIP phones, personal digital assistants (PDA),as well as applications.
SIP is already greatly influencing the marketplace. A growing number of IP Telephony Service Providers (ITSP), such as Vonage, have already started using it. Typical telephony providers, such as AT&T, have made SIP-aware networks for both internal and customer use. Cellular phone providers are using SIP to offer additional services in their 3G networks. The Microsoft real-time communications platform- includes instant messaging, voice, video, and application-sharing-is based on SIP. Cisco applications such as MeetingPlace, CallManager, and CallManager Express (CME) also support SIP. Some hospitals are implementing SIP to make heart monitors and other devices to be able send instant messages to nurses. More applications and extensions are created for SIP and their use is increasing.
SIP invitations are used to carry out session descriptions that allow participants to agree on a set of media types which are compatible to each other. SIP uses elements called proxy servers to help direct requests to the user's current location at that time, authenticate it and authorize users for using services,and implement provider call-routing policies. SIP also provides a registration function that allows users to upload their current locations for use through proxy servers. SIP runs on various different transport protocols.
The SIP protocol is actually an Application Layer protocol constructed to be independent of the transport layer underlying it; it can run on Transmission Control Protocol (TCP), User Datagram Protocol (UDP), or Stream Control Transmission Protocol (SCTP). It is text-based, incorporating the elements of Hypertext Transfer Protocol (HTTP) and the Simple Mail Transfer Protocol (SMTP).
A proxy server "is an intermediary resource that acts as a server as well as a client for the sole purpose of authoritizing requests on behalf of other clients. A proxy server most importantly plays the role of routing, which literally means its job is to make sure that a request is sent to another entity which is relatively "closer" to the targeted user. Proxies are very useful for enforcing policy (for example, making sure a user is allowed to make a call). A proxy acts as an interpreter, and, if necessary, rewrites particular portions of a request message before it is forwarded."
"A registrar is a server that accepts registering requests and places the information it receives in those requests to the location for the particular domain it handles."
"A redirecting server is a user end agent server that creates 3xx responses to the requests it accepts, directing the client/user to establish contact with an alternate set of URIs. The redirect server enables SIP Proxy Servers to direct SIP session invitations to any external domains.
SIP uses transactions to handle the exchanges between participants ,in this way delivering messages reliably. The transactions sustain an internalized state and use timers. Client Transactions send out requests and Server Transactions are there to respond to those requests with one-or-more responses. The return responses may include zero-or-more Provisional (1xx) responses or possibly one-or-more final (2xx-6xx) responses.
Transactions are further put into categories as Invite or Non-Invite transactions. Invite transactions are different in that they can create a long-running conversation, which is referred to as a basic Dialog in SIP, and so involve an acknowledgment (ACK) of a non-failing resultant response (e.g. 200 OK).
Because of these transactional mechanisms, SIP can use un-reliable transport parts such as User Datagram Protocol (UDP).
Many VoIP phone companies give customers the ability to use their own SIP devices, such as SIP-capable telephone sets, or soft phones. The wide market for consumer SIP devices continues to grow, there are various devices such as SIP Terminal Adapters, SIP Gateways as well as SIP Trunking which provide replacements for ISDN telephone lines.
The free software community started to provide increasingly more of the SIP technology which is required to build both end points and proxy and registrar servers which lead to a modification of the technology, which in turn increases global adoption.For example, the open source community at SIP foundry continuously develops a diverse array of SIP stacks, client applications and SDKs, together with entire private branch exchange (IP PBX) solutions that are in a competition in the market against proprietary IP PBX implementations from well established vendors.
Differences between IP PBX and SIP are as follows:
SIP is peer-to-peer, whereas IP PBX requires centralized controlling.
It has architectural differences. As mentioned earlier
Different Products of SIP
Enterprise Products
3Com VCX V7000 Aculab IP telephony card Aculab Prosody S Aculab Prosody X Avaya Converged Communications Server (CCS) BlueNote Networks IP Session Suite Brekeke SIP Server Brekeke PBX Cisco 2600 Series Multiservice Platforms Cisco 3700 Series Multiservice Access Routers Cisco AS5350 Universal Gateway Cisco AS5400 Univeral Gateway Cisco 7940 / 7960 IP phones Cisco SIP Proxy Server CTL VoiceSIPort (VSP) IBM Lotus Instant Messaging (Sametime) Indigo InstantConferencing Server IPtel.org SIP Express Router (SER) MCI Advantage Mediatrix 1102 Mediatrix 1104 Mediatrix 1204 Mediatrix 1124 Microsoft Live Communications Server (LCS) NEC UNIVERGE Pingtel SIP softphone Pingtel SIPxchange IP PBX Polycom SoundPoint R IP 500 Polycom SoundPoint R IP 600 Siemens (Switzerland) SIP Client Sightspeed Video Messenger 2.0 SightSpeed Web Voxtream Parlay VoXip gateway Xten X-Cipher Zultys MX30 / others
Open Source
Asterisk Minisip oSIP stack (GNU) pjsip Partysip SIP Express Router (SER) (iptel.org) SIPfoundry.org (SIPxchange, reSIProcate, ...) sipsak SUN Vovida Yate - Yet Another Telephony Engine
Consumer Products
Broad Voice Vonage
Service Provider Products
Acme Packet Net-Net Session Border Controller Aculab GroomerII aTelo media server BayPackets Agility Broadsoft BroadWorks Brooktrout SnowShore IP Media Server Brooktrout SnowShore IP Media Firewall Cisco AS5850 Universal Gateway Cisco BTS 10200 Softswitch Cisco SIP Proxy Server CounterPath EyeBeam CounterPath X-PRO Empirix Hammer XMS - Voice Network & Service Monitoring System Hotsip M2CE SCE Hotsip Multimedia Communications Engine Indigo Servers IP Unity Harmony6000 Platform IP Unity Auto Attendant IP Unity Conferencing IP Unity Unified Messaging Applications IP Unity Prepaid Services OpenSIPg Service Platform Netrake Session Border Controllers Nortel Communication Server 1500 (CS 1500) Nortel Adaptive Application Engine Nortel Communication Server 2000 (CS 2000) sentitO Proxy7 SIPquest Audio Conferencing Server SIPquest Collaboration Agent SIPquest Data Conferencing Server SIPquest SIP-H.323 Gateway SIPquest Video Conferencing Server SIPquest Voice Mail & Auto Attendant Sleipner comedia asp Sleipner comedia basic Sleipner comedia High Density Gateway Sleipner comedia RealTime Ubiquity SIP Application Server
Tools
Columbia SIPstone CT Labs Testing Services (PSTN through IMS) Data Connection DC-SIP Flextronics SIP Stack and Toolkit Empirix Hammer DEX - IMS Device Emulator Empirix Hammer NetEm - IP Network Emulator Empirix Hammer FX _" PSTN through IMS Feature Tester Empirix Hammer HCA - SIP Call Analyzer Empirix Hammer NXT " PSTN to IMS Load Tester (SIP-optimized) Flextronics SIP Server Framework HCLT SIP Server (HCL) HP SIPp Pingtel / SIPFoundry SIPx Radvision ProLab Testing Tool Radvision SIP Protocol Toolkit Radvision SIP Server Platform Radvision Videophone Framework for SIP Sipient Systems SIPFlow Standard
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