Novacom Telephone Company, Inc. Business Telephone Systems Installation & Service Since 1981

Voice over Internet Protocol (VOIP)-Novacom Business Telephone Systems and Data Networks

As adoption of IP (VOIP) telephony accelerates, more organizations are realizing the competitive
advantages of integrating advanced communications applications into their business processes.
Don Peterson, Chairman & CEO, Avaya Communications. More About Avaya IP

Network convergence is quickly becoming a reality as data, voice and multimedia come together (via VIOP)
on one network. Bill Owens, President and Chief Executive Officer, Nortel Networks Corporation.

Technology Issues-Understanding Voice Over Internet Protocol (VOIP). The standard analog and digital telephone systems
produced by most manufacturers have provided reliable telephone connections. However, these systems lack the simultaneous
transmission of voice and data resources; voice was transmitted by the telephone system's wires, and data was transmitted via
leased or dial up data lines. Technically, voice and data can be simultaneously transmitted over the same line in voice and data packets.
Each voice and data packet is identified and is transmitted over the same, multipurpose line. VOIP systems which transmit voice and data
have become very popular in recent months. VOIP is the process of digitalization of a voice telephone conversation and digital data
transmission into fixed formatted digital packets over a broad band connections ( for example: Frame Relay, Cable, T-1 or Internet dial up)
transmitted from an originating location to a received location. The processing of the voice and data requires that the voice prioritization
takes precedence over the digital data to allow the voice to be uninterrupted and clear.

Voice Over Internet Protocol (VOIP) is a very fast developing call transmission product that will eventually replace the current
public switched telephone network (the PSTN). PSTN establishes discrete speech paths and switch these calls over the network.
Although the VOIP network is improving on a daily basis, the switch service backbone we know remains the most predictable and reliable.
However, when your work at home and remote sites you require connection for voice and data, VOIP is economical and reliable. Ask Novacom.

Novacom's Experience With VOIP Networks Conforms with Predominant Recent Papers by Telecommunications Engineers in Trade Publications
Novacom's deployment of VOIP platforms within a single location has proven to be reliable and flexible. However,
in multiple site applications, we have noted that the in place Local Area Network (LAN) proved to be an impediment
to a good quality conversation over VOIP networks. The most optimal VOIP configuration includes: the recommended Novacom IP voice
and data switch; a Layer 3 LAN data switch for your workstations and printers and Quality of Service (QOS) protocols; and a Layer 3
router with QOS. Novacom recommends that these components be installed along with all Novacom VOIP networks for optimal operation.

An Engineer's Approach-IP Quality of Service for Voice Applications
Part 1 - IP QoS Protocols for Voice over IP
By Phil Thompson, Texas Instruments

Voice of Internet Protocol
Quality of service across the IP network is an essential prerequisite for deploying converged voice and data services. However, when
assessing QoS for voice applications, the subjective listener experience must also be considered. If the voice processing technology
that converts speech signals to packet data is inadequate, guaranteeing timely and jitter free delivery of voice packets simply isn't
going to occur. In the first part of this article we will examine how common IP quality-of-service protocols can be deployed across IP
networks to ensure voice traffic does not suffer when data loads are high and networks become congested. We will also explore how
voice-over-IP (VoIP) technology is implemented in gateways and to what degree this can affect the perceived user experience of
a VoIP telephone call. IP Quality-of-Service Protocols. There are several network protocols that are commonly used to bring quality
of service to an IP network. IP and associated TCP and UDP transport layers do not guarantee any level of quality of service other than
best effort. Even the Real-Time Transport Protocol (RTP) is so named only because it carries information about the timing of its payload
yet does nothing to ensure its timely delivery. Using best-effort only, it is possible to construct an ordinary IP network to carry high-quality
voice traffic effectively. Employing low overhead UDP transport protocol to carry voice frames, and RTP to provide information about the
relative timing of voice packets, users of VoIP can get good results, provided the IP network bandwidth is not exceeded and all data travels
with the same priority. However, this tends to preclude mixing voice and data on a single physical network, or using the network near the peak
of its practical capacity. For many, this defeats the point of deploying VoIP. For voice traffic to pass successfully over an IP network, the data
link and the network and transport layers must work together to provide quality of service between end points across all network layers. This
requires a variety of protocols to be present in different parts of the network, some of which are explored below.

Quality of Service-Introduction of DiffServ
Differentiated Services (DiffServ) is an example of a quality-of-service protocol that relies on prioritization of traffic in the network (routing)
layer. It describes how packets behave at network ingress and egress points by assigning them a per-hop behavior. It also assumes that there is a
service-level agreement between networks that border each other. In a small LAN environment, sufficient quality of service for shared voice and
data traffic can be ensured if all network end points and routers are equipped to provide DiffServ packet prioritization.

RSVP
ReSerVation Protocol (RSVP) enables the reservation of resources on the network. It operates at the transport layer and provides the signaling
that allows network elements to communicate their resource requirements. It is deployed both within the network infrastructure and at application end
points, and is often used in conjunction with a prioritization protocol, such as DiffServ. RSVP is a very complex protocol and represents a significant
departure from best-effort quality of service. 802.11pq. The IEEE 802.1p, 802.1Q and 802.1D protocol describes a way in which switches can classify
frames at the Ethernet layer. As such, it is a LAN protocol that allows end points and routers to assign a priority with which LAN frames get treated.
Along with DiffServ, 802.11pq can be deployed in small LAN arrangements to allow sufficient quality of service for shared services.

Introducing MPLS
Multiprotocol Label Switching is protocol-independent, and resides only in network routers. It uses traffic-engineering techniques to establish
fixed bandwidth pipes through the network. MPLS is an important protocol in carrier networks providing high levels of quality of service.

Introducing ATM/AALx
Asynchronous Transfer Mode (ATM) and its Adaptation Layers (AAL) have an inherent quality of service capable of supporting true circuit
emulation for delay-sensitive real-time data. Quality-of-service classes in ATM range from constant bit rate (CBR) to available and unspecified
bit rates (ARB and UBR). By selecting the appropriate quality-of-service class, ATM can provide
a bandwidth-efficient, superior quality link for packet voice services. However, ATM is far more costly than IP and is therefore deployed only
by carriers in the lower network layers.

Other Quality of Service
Other quality-of-service protocols in use today and worth investigating further include SBM, Weighted Fair Queuing, ML-PPP, Random Early Detection
and Expatiated Forwarding. DQoS (Dynamic Quality-of-Service) is a technique employed when delivering VoIP services via cable modem and the cable
industry's DOCSIS standard.

Voice Quality and Mean Opinion Scores
Ensuring quality of service for voice-over-IP applications is essentially about making callers believe they are talking to someone in a room next door.
Line effects, such as echo, jitter and packet loss all contribute to the perception of a poor phone connection, yet these are precisely what result if
technology of sufficient sophistication is not used for VoIP. Echo arises when hybrids in the telephone handset or telephony network cause
reflections of the signal in the analog domain. This becomes a problem when delay introduced by a packet network causes the echo to return
to the speaker with a significant time delay (>50ms). Successful VoIP implementations rely on substantial digital signal processing techniques
to compensate for echo, delay, jitter and packet loss introduced by packet networks. Measuring the quality of an audio stream digitally processed
to remove echo, jitter and delay is possible by mathematical objective methods such as Perceptual Speech Quality Measurement (PSQM) or
Perceptual Evaluation of Speech Quality (PESQ). These methods compare samples of processed audio to reference models. However, the diversity
of human speech means that the only effective way to measure the quality of a voice call is to ask someone how it sounds! A Mean Opinion Score (MOS),
which is basically a complex and costly way of getting expert listeners to give their opinions on the quality of a call, formalizes this process. Implementing
processing techniques to ensure a high MOS is always a trade-off. Voice compression will compensate for a lack of bandwidth, and large memory buffers
can compensate for jitter (at the expense of delay), but these have implementation costs associated with them as well. High compression of voice
streams costs processing power. Often, saved bandwidth is lost again when these compression techniques are applied since the resulting compressed
voice frame can be smaller than the packet header required to carry it around the network. An obvious way around this might seem to be to include
more voice frames in the packet, but, while the effect of losing 5 milliseconds of voice into the "bit bucket" may be imperceptible, the loss of 30 milliseconds
of voice will not be. The overhead is acceptable if loss is a problem on the network. However, it may not be worth it if quality of service can
guarantee packet delivery, but bandwidth for other applications is scarce. Another voice-quality issue of concern is caused by the distribution of
compression schemes within the network. Low-bit-rate vocoders, by necessity, remove information from the voice stream, which results in reduced quality.
Multiple compression, decompression and recompression (tandem transcoding) events reduce voice quality even further. The arrangement of VoIP
gateways and end points in the network should aim to compress and decompress the voice stream only once, and use tandem-free techniques
when necessary (i.e., carry compressed voice over a synchronous link instead of decompressing only to recompress at the other end). This can
also be an issue when interworking with calls that originate from mobile telephony networks, where the voice stream will have already been compressed
and decompressed.Different Types of VoIP Networks Deploying carrier or enterprise VoIP networks results in different challenges for the cost vs.
quality equation. As a result, VoIP technology deployed in these two areas varies considerably. In Part Two, we will consider specific quality-of-service
issues faced by designers of enterprise and carrier VoIP networks, and how they make use of IP quality-of-service
protocols to address these challenges.

For More Information Contact:
Novacom Telephone Company, Inc
Hillsborough, NJ 08844
Tel: 908-431-9600 Extension 217
FAX: 908-359-0961
Internet: pmilano@novacomtelephone.com