Phone Systems New Jersey - Business VoIP Basics

There are several areas that VoIP can be applied through your phone system to benefit business owners. Much of this is determined by the size and structure of the operation. It is worth putting in the time to evaluate how communications play a role within your organization and look for ways to operate more efficient and cost effectively.

Lets look at a few common implementation ideas of benefit:

Hosted VoIP phone SOLUTIONS have become much more popular as the technology becomes more stable and cost effective. In the right application the pricing of a hosted solution can be very attractive. In some cases a business owner will pay around the same monthly cost using hosted phone service in comparison to the existing traditional phone service. In essence it's like getting a phone system for nothing. Think of it like trading in your old phone system and phone lines for new technology. This doesn't relate in every case but it is definitely worth a look.

On the larger scale where an organization has multiple offices and/or remote employees, the right IP solution can have a dramatic impact related to monthly cost savings and efficiently tying together offices and employees. From a networking standpoint, hardware costs within the phone system along with networking circuits to pull this off in a traditional model would most likely be cost prohibitive. However utilizing an IP based phone system, offices and remote employees could be tied in via "the cloud" eliminating expensive hardware and circuits between sites.

The future of communications and new technology holds endless possibilities for business owners who have the imagination and innovation to drive their business to levels within the imagination. As this technology develops further it will indeed enable businesses to creatively operate their organizations from all over the world.

On the opposite end of the spectrum many companies will be lost and, out of fear, not move forward because they feel threatened by a lack of knowledge and understanding.

Much of this technology is available today. I would encourage you to contact the resources within this lens to explore the possibilities.

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Here are some common VoIP terms along with a basic understanding of how they perform:

Compression - This is technology that 'squeezes' data together by removing any unused or inefficient portions, in order to minimize the bandwidth taken up when a call is sent to its destination. There are various forms of compression - it is generally regarded that the higher the compression ratio, ie how many times the data is squeezed, the more loss of quality occurs.

Codec - This is a device capable of performing Coding and Decoding on a data stream or signal. In the case of a VoIP call, the Codec will encode the call packets and then decode it at the other end in order that a stream of voice can be heard.

IP Gateway - The purpose of an IP Gateway (or Media Gateway) is to translate data from one telecommunications network in order for it to pass on a different type of network. For example, an IP Gateway sits between the PSTN and an IP network in order to ensure that traffic can pass between these networks, ie. Converting a call from a PSTN phone into a format so it can be passed on IP.

Jitter - An abrupt or unwanted variation in the characteristics of a signal. For example, the gap between successive pulses. This will cause a distortion in the sound noticeable by the person being called.

Media Gateway - A device that is used on the edge of an IP network, which converts data from one network into a format so it can pass on another different network, eg from the PSTN to an IP network.

NAT (Network Address Translation) - This is the process of re-writing both the origin and destination IP Addresses for IP packets as they pass through a router or a firewall. In most cases, this is to allow multiple users on a LAN or other network access to use the Internet through a single public IP address.

Packetised voice - Any voice call which is converted into discrete packets in order to be transported over an IP network.

Packet loss - A situation where one or more packets does not arrive at its final destination. This causes a noticeable gap in the flow of voice that may seem like the person talking is missing parts of their speech!

Router (IP) - A router sits in the Network Layer of the OSI Interconnect model and serves to pass data packets across the network to their end destination. Intelligent routers will choose the most efficient path to send that traffic.

SIP (Session Initiation Protocol) - A protocol designed to initiate, maintain and terminate a session involving multimedia streams such as voice, data and video. All SIP messages are either a request or a response to a request.

VoIP - Voice Over IP is the technology that provides the capability to break voice streams down into small pieces, group them together in an IP Packet, and then send them over an IP network to the far end caller.

There are a number of VoIP protocols that have been developed to establish, maintain and terminate multimedia sessions such as telephony, video-conference and data exchange.

As carriers seek new sources of revenue, they are forced to adopt more flexible business models that include peering with other carriers and direct IP connectivity to the enterprise. Within this environment, carriers must meet several technical challenges to fulfill their business objectives. Specifically, while the use of VoIP technologies reduces operating costs and provides new revenue bearing applications, it also creates new issues of network security, signaling interworking and multivendor interoperability.

The two principal VoIP protocols, however, and those that you will hear referred to regularly, are H.323 and SIP (Session Initiation Protocol).

The proliferation of VoIP endpoints within the enterprise has forced the carrier to confront VoIP traffic on the access network. Likewise, carriers must adapt to IP when peering with the early adopters of VoIP that use H.323 technologies for the trunking or backhaul of their voice traffic. Because SIP and H.323 messages carry end point addresses within the payload, these carriers are forced to expose the elements of their core network to other carriers and enterprises. This presents a threat to the security and service availability of the carrier's network.

Of the VoIP call setup protocols, SIP and H.323 have gained prominence.

Here we take a slightly closer look at both of these protocols, their uses, and how they are shaping the future of Voice Over IP.

H.323

H.323 is a protocol that was developed by the International Telecommunications Union Telecoms Standardisation Sector (ITU-T). Its original use was for transmitting audio, video and data across an IP network, such as the Internet. Any two user devices may communicate with each other, as long as they comply with H.323.

The various components of H.323 address call signaling and control as well as multimedia transport and control and bandwidth control for point to point and multipoint conferences. H.323 was originally designed with the purpose of transporting multimedia applications across a Local Area Network (LAN), but more recently it has evolved to address the needs of a VoIP network. The standard was designed with the following objectives :

- To be based on the existing standards

- To add some of the advantages that packet switching networks offer to transport real-time data

- To solve the problems of real-time data on packet switching networks

One of the advantages of H.323 is that the designers realised that over time requirements for communication change, and therefore allowed that the equipment manufacturers could add their own specifications and features to H.323 whilst still allowing these devices to communicate with each other.

As early adopters of VoIP, many carriers tended to implement networks using a single H.323 vendor such as Cisco, Clarent or VocalTec. Given that each vendor had differentiating features that eased their deployment, interoperability issues arose with other H.323 implementations. As these carriers migrate to next generation infrastructure, their fiscal constraints require them to preserve their existing infrastructure investment and find a way to consolidate these islands of H.323 into a single operational network.

However, more recently H.323 is being substituted with another of the VoIP protocols, more widely being used in VoIP networks - SIP.

SIP (Session Initiation Protocol)

SIP is one of the application layer signaling VoIP protocols, which is used to establish, maintain and terminate multimedia sessions. SIP invitations can be used to establish a session, and also to carry session descriptions.

It is a text based VoIP protocol that is part of the overall Internet Engineering Task Force (IETF) multimedia architecture. SIP is used more and more as the protocol for Voice Over IP, and in 2000 it was also chosen as the 3G protocol used by Mobile Network Operators (MNO's) within the IP Multimedia Sub-system (IMS).

All SIP messages are either requests or responses to a request. The general format is:

- A start line

- One or more header fields

- An empty line

- A message body (optional)

SIP uses six types of requests:

1. INVITE - Indicates a user or service is being invited to participate in a call session.

2. ACK - Confirms that the client has received a final response to an INVITE request.

3. BYE - Terminates a call and can be sent by either the caller or the callee.

4. CANCEL - Cancels any pending searches but does not terminate a call that has already been accepted.

5. OPTIONS - Queries the capabilities of servers.

6. REGISTER - Registers the address listed in the "To" header field with a SIP server.

A motivating goal in the creation of SIP as another of the VoIP protocols was to provide a signaling and call set-up VoIP protocol for IP-based communications that can support the call-processing functions of the PSTN. Ie it would make the interaction of the PSTN and IP-based networks more feasible.

Hardware endpoints (SIP phones) which have the look and feel of a traditional phone, are widely available from several vendors. They use Electronic Numbering to translate phone numbers to SIP addresses so calls to other SIP users bypass the traditional telephone network. Examples of SIP applications today are Windows Messenger and AOL's Instant Messenger.

SIP based networks are emerging as service providers are deploying SIP based infrastructure to prepare themselves for the flexible and collaborative applications being developed for SIP. Microsoft XP provides a great impetus for new applications to take advantage of this converged next generation network infrastructure. As a result, carriers must have a technology strategy for both VoIP protocols (H.323 to SIP) interworking which ensures maximum service reach to any endpoint and also merges their H.323 and SIP networks into a single network for greater operational efficiencies.