Additionally other significant H.323 advantages include no restriction relating to the type of platform H.323 will operate on simply because H.323 has been developed by several manufacturers that made certain interoperability was a feature of the system. Through H.323 users can select the most appropriate codecs that support their computers and network selections. Microsoft’s NetMeeting utilizes audio and video conferencing processes conceived in the H.323 system that allows NetMeeting to function with other H.323 standards-based products.

H.323 product interoperability is calculated using the following categories of scale: a) Call signaling and control — a mechanism that establishes third-party acceptance using the same codecs or that an appropriate set of codecs can be used; b) Audio and Video streaming — a mechanism that identifies when interoperability issues may arise; c) Audio and video codec compatibility — a mechanism that examines third-party products to ensure compatible codecs are installed. Audio and video codecs present the audio and video information as a compressed package to be sent over the network. H.323 support amounts to the options it makes available to handle the audio and video coding.

In relation to data communications H.323 makes a provision for using T.120 as the mechanism by which it packages and sends data. File transfer and program sharing data use T. 120 support to operate in conjunction with H.323 connections.

Through the use of the H.323 protocols NetMeeting users are able to establish and maintain both audio and video connections to other compatible audio or video clients. In addition it seems highly probable that improved security and interoperability in relation to streaming media servers will appear in the near future. H.323 does not appear to have peaked as it continues to provide the necessary support for audio and video communications while at the same time possessing further development potential.

ISUP or ISDN is a part of the complex signaling system SS7 uses to set up calls through the PSTN and is also able to provide information relating to timeslots. ISUP will ensure that should no outbound CIC be available a blocking message be routed back to the previous switch in order that a new direction be attempted. Also the circuit based protocol to establish, maintain and end connection for calls is resolved by 1SUP. In North America ITU-T set the criteria to be followed by ISUP while in Europe that task was carried out by ETSI. Both ISUP specifications provide the blueprints for national ISUP variants.

Today the SS7 network, which separates the signaling planes from the voice circuits, provides the link between VOIP traffic and the PSTN network and also plays a role in cellular networks such as GSM and UMTS for circuit switched (voice) and Packet switched (data) applications. The SS7 network manages calls by identifying end-to-end addressing and from there controls all the routing judgments necessary for a seamless operation. In addition SS7 supports all the telephony services such as 800 numbers, call forwarding, caller ID and local number portability. The SS7 Protocol consists of four Message transfer divisions with ISUP handling the circuit based protocol in order to establish, maintain and end connection for calls.

STUN was designed to act as a client-server protocol and can be found included in a Voice Over Internet Protocol (VoIP) phone or software package. A STUN client will send a request to a STUN server which will respond by providing the STUN client with the identity of a public IP address used by the NAT router together with the port opened by the NAT to allow incoming traffic into the network and the category of NAT in use. Identifying the category of NAT is necessary because of different handling protocols followed by the various NATs currently operating. With generic Internet applications vulnerable to disruption when end-to-end significance of an IP packet is interrupted it is significant that NAT can create problems both at the protocol level and with application data.

STUN is able to operate with three of the four main NAT types but is unable to operate with Symmetric NAT, which are commonly located on networks found in large companies. The problem with between Symmetric NAT and STUN relates to mapping with the NAT server and STUN not able to mirror identical endpoints.

However Symmetric Nat apart STUN is capable of healthy communications with Full Cone, Restricted Cone and Port restricted Cone types. With Full Cone either side is able to set up communications. With both Restricted Cone and Restricted Port Cone it is necessary that both sides begin simultaneous transmission.

The usefulness of STUN extends to include handling UDP packets that require the transfer of signalling traffic containing sound/video/text signalling across the Internet that is located behind a NAT not able to operate a traditional connection. STUN provides the appropriate mechanisms to ensure a connection can be initiated and maintained.

Problems that occur with STUN reflect a lack of standardization in both behaviour and management systems contained within the different types of NAT devices in current usage. STUN should be viewed as a short-term solution to a long-term problem that will inevitably need to be addressed by the introduction of standardized NAT devices that provide global interoperability. With the pressure from users mounting the need for equipment able to comfortably interact with sound/video/text signalling across the Internet relevant manufacturers would be unwise to ignore the scale this very significant compatibility requirement possesses.

Thanks to SS7 network maintenance, messaging, interfacing and the provision of a universal telephony network structure are made available. Setting up a call, swapping user information, call direction, billing structures and the support of Intelligent Network (IN) services are fundamental to the functions expected of SS7. [N technology brought the concept of splitting the signaling planes to the environment with significant uses of IN technology directed to simple number translation services as well as complex CLASS and prepaid telephone calls. The SS7 network comprises link types together with three signaling echoes — Service Switching Point (SSPs), Signal Transfer Point (STPs) and Service Control Point with each code uniquely identified and links between nodes being full-duplex 56kbit/s and or 64kbit/s.

SS7 separates the signaling planes from the voice circuits provide the Voice Over Internet Protocol (VoIP) and Public Switched Telephone Network (PSTN) network connection while also playing a role in the cellular networks such as GSM and UMTS for circuit switched (voice) and Packet Switched (data) applications. By identifying end-to-end addresses SS7 is able to control all the routing judgments necessary for a seamless communications. Additionally SS7 also supports all the telephony services including 800 numbers, call forwarding, caller ID and local number portability. The SS7 Protocol consists of four Message Transfer Parts (MTP) categorized as MTP 1, MTP2 and MTP3 with 4 comprising several different user parts such as TUP, ISUP, TCAP with INAP, MAP and SCCP. MTP controls the transfer protocols associated with the network interface, information transfer, message handling and the routing to higher levels. MTP3 handles the management services and end-to-end contact and routing connection less messages whilst ISUP completes the circuit based protocol to connect, maintain and disconnect the call. MAP is performed by TCAP which resolves database queries and handles advanced network functionalities or links to Intelligent Networks.

H.323 compatibility extends to include other ITU-T protocols such as those summarized below:
• H.225.O details call signaling, audio and video media, stream packet construction, media stream synchronization and control message formats.
• H.245 details controls the protocols that govern the descriptions of messages and procedures used to open and close logical channels for audio, video and data.
• H.450 details an explanation of Supplementary Services.
• H.235 details security contained in H.323.
• H.239 details the use of dual stream in videoconferencing.
• H.460.17-19 details firewall navigation within the H.323 environment.

The breadth of H.323 usability extends to include components such as the Multipoint Control Unit (MCU) which in relation to H.323 makes possible a connection between three or more H.323 terminals and participation in a multipoint conference. In addition H.323 gateways which are the translation apparatus related to call signaling, data transmission, and audio and video transcoding fulfills the role that addresses interoperability issues. Also Gatekeepers interact with H.323 devices by controlling the number and type of connections entering a network; directing a call to the correct destination whilst managing the network address for incoming calls.

With compatibility issues significantly reduced through the adoption of H.323 products and the fact that H.323 has not yet reached its full potential with improved security and interoperability in relation to streaming media servers appearing a strong possibility for the future. H.323 continues to offer a useful communications apparatus and one that should endure for some years yet.

Thus, the advent of Internet telephony and voice over Internet protocols has risen to become part of peoples’ daily lives.

The Internet telephony and the VoIP are referring to the transport or communication of telephone calls coursed through the Internet.

The SIP

The session initiation protocol, or more popularly known as SIP, is a kind of protocol developed to set a standard for modifying, initiating and terminating interactive and real-time user sessions using different multimedia like videos, virtual reality, online games, instant messaging and voice.

SIP was developed years ago, but it was only in November 2000 that the protocol was approved and accepted by the industry as a formal signalling protocol and as a permanent and regular element of the ‘next generation networks’ protocol.

To put it simply, SIP is the corresponding protocol intended and custom-made for VoIP and other multimedia sessions across the Internet. It is somehow similar to the standards used in widely popular online services and features like videos, online gaming and instant messaging.

Experts put it simply. SIP, according to them is somehow similar to the HTTP, which is the World Wide Web protocol. That is why through SIP, it can be noticed that messages contain headers and message bodies.

SIP provides the much needed facilitation protocol for voice over Internet protocol and Internet telephony. There are numerous valid reasons why SIP is deemed necessary in the transaction over VoIP and Internet telephony.

One, SIP can enable an outgoing call to reach the intended party wherever it is located in the world. That is because SIP has a user location and name translation feature. A detail of the nature of the particular phone call is also supported by SIP.

Second, SIP can allow the people involved in the call to reach a consensus or agree on a support feature that can recognize all parties, since not all are assumed to always have the same support levels.

Third, users are enabled to modify call characteristics during the entire course of a single call. For instance, a user is using a voice-only IP call. During the call, that user can modify the setting and transfer the call into a video function. A third party is also welcome to enter or join a call, and that party is also enabled to change call characteristics. Amazing, right?

Overall, the SIP makes up for one amazing and awesome Internet and VoIP calling experience. Today’s generation is so lucky to be able to enjoy all of these features, which in the past have only been part of dreams and fantasies.

Through H.323, users are able to connect virtually with other online users through many other devices and products that support the H.323 system.

H.323’s Support System

The standard specification system known as H.323 was developed by the International Telecommunications Union. H.323 is based on the protocols set by the Internet Engineering Task Force or IETF, Real-Time Protocol or RTP, and the Real-Time Control Protocol or RTCP.

Thus, H.323 is highly reliable in quick-transferring data, very ideal for Internet telephony and VoIP. The system was initially created to mechanize transporting of multimedia applications over local area networks.

But technology is fast-evolving and modifying itself. H.323 has further developed to address the fast-growing needs and requirements of various VoIP networks around the world.

Moreover, H.323 is one among the series of standards for communications that facilitate for video conferencing within a wide range of networks. In some parts of the globe, it is also known as H.32X.

The Importance of H.323

There are a number of significant reasons why H.323 is deemed important and significant in the modern Internet communications setting.

The H.323 is flexible and comprehensive that fit the requirements and settings of voice-only handsets and multi-media stations for full-scale video conferencing. In a few years time, experts forecast that H.323 would come mainstream and would be more popularly known and in demand as telecommunications services continue to evolve.

There are a number of advantages and perks attributed to H.323. Among them are as follows:

1. The H.323 is able to provide standards needed for interoperability between different networks and local area networks.

2. H.323 enhances modern multimedia platforms as processors get even faster and computer chips are enhanced and made more efficient. Thus, VoIP and Internet telephony are significantly improved, to the benefit of the consumers and users.

3. H.323 provide application-to-application, vendor-to-vendor and device-to-device interoperability, enabling users and customers to interoperate especially with other products that are compliant to H.323.

4. Multimedia standards for current online infrastructure are set by H.323. Thus, a highly variable and effective local area network with ideal latency is provided.

With different types of external equipment appearing in the market to accommodate people’s communications needs, VoIP telephony was born. Voice over Internet protocol or VOIP telephony is today’s most popular service that allows users to make and terminate calls over the Internet to landline units. Apart from lower costs, voip telephony offers the same quality service as the traditional landline calls.

Telephony uses different protocols in transmitting calls between two different sets of networks. While the IP telephony uses the TCP/IP protocols of the Internet, old telephone service uses public switched telephone network (PSTN). This can bring a lot of problems for the users like loss of data, inability to transmit good voice quality and frequent termination of calls.

To solve the problem of the delay in voice transmissions, two standards have been used for VOIP telephony: ITU-T H.323 and the IETF Session Initiated Protocol (SP). These protocols give the same end-user service but differ in their approach to signaling functions. H.323 is the current standard for sending voice and video using the IP on the public Internet and within an intranet. However, SP is becoming a threat to H. 323 protocol users.

SP uses the benefits of IP by using a HTTP-like protocol that makes it less complex. It uses textual encoding so it is much easier to extend in terms of applications, debug and process through text-processing programs. This, in turn, provides users and organizations with better control over their voice and data communications relay.

Apart from these two protocols, there is another one used online and it is called the Media Gateway Control Protocol (MGCP). MGCP is a protocol being used to control (VoIP) Gateways from call agents. This means, MGCP acts as a control tower for the existing SP or H.323 protocol. As the last two protocols give call control functionality, MGCP manages data establishment in the media gateways.

What makes MGCP more superior to SP protocols is its adaptability to other gateways and Ips. SP is a peer-to-peer protocol, which means it only works best if the external equipment also uses SP. MGCP can work within the same network and beyond the limitations of other protocols online. The result is a faster and clearer delivery of voice transmission over the Internet.

However, MGCP is not meant to replace SP or H.323 protocols. SP and H.323 provide symmetrical setup or control. MGCP then divides the call setup or control and the media establishment functions, making for an easier and clearer data flow.

SP has been lauded as the next best thing for VoIP telephony. With the possibilities of web integration, instant messaging capabilities and the flexibility of new applications, it promises to have great potential in replacing the H.323 standard. For now, SP and MGCP are better off working together to serve the VoIP telephone users.

Signaling System #7 (SS7) is just a set of telephony, signaling protocols used in setting up the vast majority of PSTN telephone calls. Though abbreviated as SS7, it is known differently in different places. It is referred to as CCS7 in North America whereas in UK, it is at times known as C7 (CCITT number 7) or number 7 and CCIS7. AT&T had developed the SS7 Protocols in 1975 and was defined as a standard in 1981 by ITU-T in its ITU-T’s Q7XX-series recommendations. SS7 are a replacement for the SS5, SS6 and R2 standards. The out-of-band system of signaling, used by the SS7 is a better option to the in-band signaling system used by SS5 and SS6. The separate signaling channel system used by the SS7 also did away with the security problems faced by the earlier signaling Systems. ITU has standardized an international version of SS7. However, in the US, ANSI governs the standard for SS7.

Network maintenance, messaging, interfacing and providing a universal structure for telephony network are the features of SS7. Establishing a call, exchanging user information, call routing, billing structures and supporting intelligent network (IN) services are the integral part of the SS7. IN technology introduced the concept of separate “service plane” and the most important uses of IN technology include the simplest number translation services and the complex CLASS and prepaid telephone calls.

Uses of SS7
SS7 is an important link between VoIP traffic and PSTN network. It is also used in the cellular networks like GSM and UMTS for circuit switched (voice) and Packet switched (data) applications. The GCM/UMTS CS interfaces in the MSC transported over SS7 are B-> VLR with B as interface “internal”; D-> HLR for location update and attaching to the network; E-> MSC for inter-MSC handover; F-> EIR for identity check of the equipment; H-> SMS-G for SMS over CS. Some to the GMS/UMTS Ps interfaces in the SGSN transported over SS7 include Gr-> HLR to attach to the Ps network and location update; Gd-> SMS-C for SMS over PS; Gs-> MSC for CS+PS signaling over PS; Ge-> charging for CAMEL prepaid and Gf-> EIR for equipment identity check.

SS7: Network
SS7 is a network, which splits the signaling planes from the voice circuits. The network is made up of link types (A, B, C, D, E and F) and three signaling notes – Service Switching Point (SSPs), Signal Transfer Point (STPs) and Service Control Point. There is a unique identification number corresponding to each code. Links between the nodes are full-duplex 56 kbit/s and or 64 kbit/s. They are timeslots (DSOs) within E1 or T1 trunk in Europe whereas in USA the SS7 links are usually carried over a network called non-associated signaling.

SS7 – PROTOCOL
The SS7 network sets up and tears down the call, handles all the routing decisions and supports all telephony services such as 800 numbers, call forwarding, caller ID and local number portability (LNP).

The SS7-Protocol has only 4 levels identified as Message transfer Part (MTP) 1, MTP2, and MTP3 and with level 4 consisting of a number of different user parts, such as TUP, ISUP, TCAP with INAP and MAP and SCCP. Transfer Protocols including network interface, information transfer, message handling and routing to higher levels are controlled by MTP. The end-to-end addressing and routing connection less messages (UDTs) and management services are provided by the MTP3, also known as Network Service Part (NSP). TUP is used to connect calls via link-by-link signaling system. Circuit based protocol to establish, maintain and end connection for calls is taken care of by ISUP. Data base queries and advance network functionalities or links to intelligent networks (INAP), mobile services, MAP is done by TCAP.

Around the world, VOIP systems are being unravelled and launched left and right with the principal purpose of luring more users and eventually generating higher revenues.

Because technology and market demand are changing rapidly nowadays, VOIP systems or IP telephone systems are also evolving at a pace or rate that can be comparable to how data are transferred online.

Fundamentals of the VOIP

The VOIP is an Internet-based telephony service that allows the transmission of voice and audio messages online. Thus, using VOIP, the communication between two parties parted by locations are made easier, faster and more convenient.

According to statistics and reports, about 10% of telephone subscribers in Japan and in South Korea have shifted from the traditional landline telephone to VOIP or IP telephone systems, as of January 2005. The figure is estimated to be higher in the US and in Europe, where adoption of new technologies is basically more rapid.

The wireless IP phone

VOIP basically requires broadband Internet connection, which is the prerequisite for faster, more efficient and more quality-reliable online connection.

Thus, IP telephone systems now are competing head on with wireless IP phone systems, which are rolled out digitally and more conveniently through WiFi hotspots.

Because the wireless Internet connection is currently the more preferred mode of online connection worldwide, VOIP transactions can now be also made wirelessly.

The advantage of wireless

Of course, being wireless means a lot of advantage for numerous and tech-savvy users. Nowadays, it is assumed that people are increasingly annoyed by wire connections.

That is why through the years, computer makers have developed computers that are operated wirelessly. To secure Internet connection, these computers do not need to connect through the wires of their Internet service providers.

The WiFi

The WiFi technology has made wireless Internet possible and accessible to everyone. Just like mobile phones, which rely on mobile carriers’ signals, WiFi also relies on signals.

Thus, wireless Internet can be accessible to WiFi hot spots, or those designated areas where wireless Internet signals are run through.

Since the Internet can be accessed nowadays wirelessly, so is VOIP. It can be assumed, and safely asserted, that wherever there is Internet, there is VOIP.

Cheaper service

The advantage of wireless VOIP, or wireless Internet in general (specifically WiFi), is that it is practically cheaper. You must be wondering why since it is a modern technology.

The answer would be because wireless VOIP and Internet providers want to lure more users, so they have to compete intensely with traditional providers. And how to better convince subscribers and users than to provide a cheaper and more practical alternative.

Thus, with the emergence of wireless Internet connection, VOIP systems and services, logically, are also made wireless. How technology works!