Voice over IP Basics for IT Technicians - Part One

This Article has been divided into two parts. This is the part one of the article.

The IP phone is coming – or has arrived – on desk near you. The IP phone is not a PC, but does have a number of hardware and software elements also resident in PCs. Voice over IP (VoIP) brings a new environment to the network technician that requires expanded knowledge and tools to deploy and troubleshoot IP phones. The LAN diagnostic tools have to analyze Ethernet, but must now support.

VoIP signaling protocols and voice transmission. Understanding VoIP will become mandatory for the network technician. This paper provides an introduction to VoIP technology and operation.

What is VoIP?

Voice over Internet Protocol (VoIP), also called IP Telephony, is rapidly becoming a familiar term and technology that is invading enterprise, education and government organizations. VoIP is designed to replace the legacy TDM technologies and networks with an IP-based data network. Digitized voice will be carried in IP data packets over a LAN and/or WAN network. Installing and testing the VoIP network of IP phones, gateways and servers requires new tools and expanded knowledge. This paper discusses VoIP operation and testing.

The legacy telephone network has provided reliable and high-quality voice communications for many years. It delivers voice and speech over a standardized 64 Kbps channel. The 64 Kbps bandwidth is guaranteed for each call and the speech path carries the voice as a continuous digital stream. Digital voice is NOT carried in packets. Enterprise and residential callers use DTMF (Touch Tone), TI channel and ISDN D channel signaling to set up and manage the call. Within ISDN, signaling is carried in packets on a separate signaling channel over the Basic Rate Interface (BRI) and Primary Rate Interface (PRI) connections to the carrier. The carrier then translates the signaling (all forms) in an internal signaling protocol called Signaling System #7. Signaling protocols are primarily active at the beginning and end of a call.

In a VoIP network, there is a signaling protocol and a speech transmission protocol. Both protocols require all information be carried in IP packets. Several standards-based choices are available for signaling protocols, including H.323, SIP, MGCP and H.248 (MEGACO).

Most IP PBX vendors have developed their own proprietary signaling protocol, the most common of which is Cisco’s SCCP (Skinny) protocol.

RTP is the standard speech transmission protocol used with VoIP networks. The speech is digitized, placed in packets, and transmitted through the IP network. Multiple packets are required to carry a single spoken word. The voice is digitized using one of the G.7xx standards. Each of these standards will be discussed in detail later in this paper.

Why, then, do organizations move from a TDM-based telephone to an IP-based telephone? This is a major change in the underlying technology, yet we are still making phone calls as we have for years. One or more of the following may justify the move to VoIP:

• Reducing long distance charges, especially international long distance

• Reducing staff by combining voice-network and data-network management and eliminating redundant functions

• Adding expanded applications that are not offered by TDM-based systems

• Having one common network for different forms of communication

• TDM vendors are not offering new systems, thus forcing customers to eventually adopt IP-based telephone systems.

There are two forms of a VoIP call. If you have Microsoft’s NetMeeting, you can set up a PC-to-PC call without working with a call server. This is typically how the early users of VoIP made calls. However, the prevalent enterprise VoIP solution requires a call server (the standards community calls this a “gatekeeper”) to be part of the network configuration. Although it is called a server, the server does not operate like a traditional server. An e-mail server and a PC are in constant contact for the e-mail operations. In VoIP, the call server (see Figure 1) controls all the services offered, provides control over the call, supports the telephone features, authenticates and authorizes the caller and implements security. The call server is NOT the telephone switch. Once the call server sets up a phone (peer-to-peer) call, the server becomes dormant during the speech transmission unless the phones contact the server to indicate a change in status or the call server wants to change the call configuration, such as indicating there is a call waiting. The server is there to process the signaling, but does not switch the speech. The speech packets are passed directly from phone to phone.

There are two major categories of IP phone implementations: hard phone and soft phone. The hard phone contains all the hardware and software to implement VoIP. It is not a PC, but is specifically designed as a phone. Hard phones can be simple in their functions,

but can also have color displays with touch sensitive screens and may even support web browsing. There is no typical hard phone on the market. The second category, the softphone, is a headset connected to a PC with all the telephone features implemented by the sound card and software resident in the PC.

Another piece of hardware, the gateway, is usually part of the VoIP network. Most organizations will have legacy phones, fax machines, modems, connections to the PSTN, and other devices that originally connected to the organization’s telephone switch,called a PBX. When migrating to VoIP, these devices and interfaces will have to be connected to a conversion system that supports the legacy devices and interfaces on one side and connects to the IP network on the other. The legacy devices will be connected to an access/gateway and the PSTN interface connection will be terminated on a trunk gateway.

Standards for VoIP

“Standards are great, I have so many to choose from” is a quote that amply describes VoIP. There are multiple signaling standards.

• H.323, the ITU standard that was published in 1995, started the development of VoIP products and services. There are four versions available. V.1 is obsolete and has been discontinued in virtually all products. Versions 2, 3 and 4 are used in today’s products.

These three versions are similar in design and are upwardly compatible. This is the dominant installed signaling protocol for use with hard and softphones.

• The Session Initiation Protocol (SIP) was produced by the IETF as an IP standard. Although SIP is gaining considerable attention, it will not become the dominant installed protocol for a few years. The attractions of SIP are better interoperability among vendors, easier application development, operation that is close to other existing IP protocols, and easier operation through firewalls.

It is usually part of hard and softphones, but it may also be used with gateways. SIP is a completely different design when compared to H.323.

• MGCP is a protocol used primarily with gateways, although an occasional hard phone may support MGCP.

• MEGACO/H.248, another standard protocol, is a combined effort of the ITU and IETF. It can be used with gateways and server-toserver communications. It is not found in hard or softphones.

In addition to the standards, nearly every IP PBX vendor has produced a proprietary signaling protocol. The most commonly found protocol is Cisco’s SCCP, or “Skinny,” protocol. These proprietary protocols may be variations of the standards or may be uniquely designed. They each provide the call control found in the standard protocols. An IP PBX vendor usually supports one or more of the standard signaling protocols plus their proprietary protocol. All the signaling protocols follow the same path for control as shown in Figure 2. The H.323, and most proprietary signaling, is carried over TCP, while SIP operates over UDP.

<!--[if !supportEmptyParas]--> 

Speech is carried in packets that use the Real Time Protocol (RTP) standard. Each RTP packet contains a piece of a digitized word.

Multiple RTP packets, when combined at the receiving IP phone, produce a spoken word. The RTP IP PBX vendors commonly implement RTP. Proprietary protocols that operate like RTP are uncommon. The speech paths connect directly between phones and gateways; speech does not pass through the server, nor is speech carried by the signaling protocols as shown in Figure 3.

There are several voice digitization standards and some proprietary techniques in use. Most vendors support one or more of the following

ITU standards and avoid proprietary solutions:

• G.711 is the default standard for all vendors, as well as for the PSTN. This standard digitizes voice into 64 Kbps and does not compress the voice

• G.729 is supported by many vendors for compressed voice operating at 8 Kps. With quality just below that of G.711, it is the second most commonly implemented standard.

• G.723.1 was once the recommended compression standard. It operates at 6.3 and 5.3 Kbps. Although this standard reduces bandwidth consumption, voice is noticeably poorer than with G.729 and is not very popular for VoIP.

• G.722 operates at 64 Kbps but offers high-fidelity speech. Whereas, the three previously described standards deliver an analog sound range of 3.4 kHz, G.722 delivers 7kHz. This version of digitized speech will become common in the future

In all cases, the IP phones and gateways collect about 10 to 30ms of digital speech and place it in the RTP packet for transmission.