Business Data Communication Networking - bapp03.pdf - po ang - Januszek66

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APPENDIX C

IP TELEPHONY

IP Telephony refers to the use of Internet protocols to provide voice, video, and data in

one integrated service over LANs, BNs, MANs, and WANs. When most people talk about

IP Telephony, they mean Voice over IP ( VoIP ). Voice over ATM ( Vo AT M ) and Voice over

Frame Relay ( VoFR ) are also fairly common, and, as the names suggests, are close

cousins to VoIP.

VoIP provides three key benefits compared to traditional voice telephone services.

First, it minimizes the need for extra wiring in new buildings (there is one cable for both

voice and data, not two). Second, it provides easy movement of telephones and the ability

of phone numbers to move with the individual (the number is installed in the telephone,

much like an IP address, so anywhere the phone is connected or the phone number pro-

grammed, the phone will connect, even if it is halfway around the world). Finally, VoIP is

generally cheaper to operate because it requires less network capacity to transmit the

same voice telephone call over an increasingly digital telephone network.

VoIP requires a VoIP Private Branch Exchange ( PBX ), which connects the organi-

zation’s internal telephone network into the public switched telephone network ( PSTN ).

The PBX can also be connected into the organization’s WAN and be configured to route

calls through the WAN to other organization locations (or even over the Internet), thereby

bypassing any long-distance charges in the PSTN (although the call does use up network

capacity on the organization’s WAN or Internet connection).

The VoIP PBX can be considered a gateway that connects the internal IP telephones

to the PSTN. 1 When the IP telephone user lifts the receiver off the hook to place a call, the

IP telephone sends a message to the PBX, which responds by sending a dial tone. Once

the number has been dialed, the IP telephone then sends a message to the PBX with the

telephone number, and the PBX connects the telephone into the PSTN or over the organi-

zation’s WAN to the VoIP PBX at the other location.

VoIP often uses H.323 at the application layer (see Chapter 2), although Session Ini-

tiation Protocol ( SIP ) is also common. Media Gateway Control Protocol ( MGCP ) and

Skinny Call Control Protocol ( SCCP ) are other competing application layer protocols al-

though they offer fewer features than H.323 and SIP. All four protocols (H.323, SIP,

MCGP, and SCCP) operate at the application layer and contain all the functions needed to

start and end telephone calls, as well as to transmit the call data. MCGP and SCCP require

1 Another type of VoIP PBX uses regular telephones and regular analog circuits inside the organization, and the

PBX does all the conversion between the analog telephone and the digital network. This is an older, temporary

approach that is quickly disappearing.

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the PBX to act as a server and communicate with other MCGP and SCCP devices only

through the PBX server; H.323 and SIP can both communicate with other H.323 and SIP

clients without needing to go through the PBX (except for telephone number resolution,

which is a process much like using a DNS to resolve an application layer name into an IP

address [see Chapter 5]). SIP is a newer and more efficient protocol than H.323, and was

developed using HTTP as its core, which means that it is simpler to debug and can be eas-

ily integrated into Web-based applications and SMTP e-mail applications.

In order to use VoIP, a device such as an IP telephone must support H.323, SIP,

MCGP, or SCCP, and also contain a CODEC to convert the incoming analog voice signal

into a digital bit stream (see Chapter 3). The CODEC is also used at the receiving end to

convert the digital data back into the analog voice data. The most commonly used digital

voice protocols are 64 Kbps PCM , 32 Kbps ADPCM (see Chapter 3), and more recent

variants on them, such as 8 Kbps G.729 (also called CELP) or 6.3 Kbps G.723 (also

called MPMLQ). As might be expected from their lower bandwidth requirements, sound

quality can become an issue.

A technique called Mean Opinion Score (MOS) has been developed by ITU-T to

subjectively rate the voice quality of different CODEC standards. A MOS of 5 is the theo-

retical maximum (meaning excellent) while a 1 is the lowest score (very poor quality). A

4 is generally regarded as an acceptable level of quality, with barely perceptible levels of

quantizing error. PCM has a MOS of 4.1; ADPCM, 3.85; G.729, 3.92; and G.723 (at

6.3 Kbps), 3.9.

Once the CODEC has produced the digital data, the data is surrounded by an appli-

cation layer packet. H.323 often uses RTP (see Chapter 5) for transmission through an

IP-based network. The RTP packet in turn is surrounded by UDP, IP, and data link layer

packets (e.g., Ethernet, frame relay) for transmission through the network. Voice data

packets tend to be very small, and thus the packets added at layers 5, 4, 3, and 2 can add

considerable overhead to the transmission. A new version of RTP called Compressed Real

Time Protocol (CRTP) has been developed that enables the set of RTP, UDP, and IP pack-

ets to be compressed to 2 bytes, thus significantly reducing the overhead.

Voice Activity Detection ( VAD ) is another way of reducing the network capacity re-

quired to send VoIP calls. With VAD, the end VoIP device monitors the analog voice sig-

nal and if the signal drops below a certain amplitude, then the device assumes that no one

is speaking and stops sending packets. Since most conversations are silent at least half the

time, this can significantly reduce the bandwidth required. VAD must be done carefully to

avoiding clipping the speech (cutting off the beginning and end).

The other problem with VAD is that because no data packets are transmitted, the

line is completely silent. We are so used to experiencing background noise from interfer-

ence on traditional analog telephone calls that silence is usually interpreted as meaning

the call has been disconnected. Therefore, several vendors have begun to add in back-

ground comfort noise (sometimes call pink noise) when no data packets are flowing. The

VAD device at the sender’s location initially sends some ambient background noise pack-

ets, which are recognized by the receiving device. After a few seconds, the VAD sending

device begins operating normally and no longer sends background noise packets when

speech ends. When the receiver detects the slowdown in packets, it repeatedly reproduces

the ambient background noise packets to fill the “dead air” so that the receiver does not

think the call has been disconnected.

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As a result of VAD, the network capacity requirements for VoIP on Ethernet net-

works ranges from about 8 Kbps using G.729 and CRTP to 45 Kbps using PCM with

RTP. On ATM networks, it ranges from about 6 Kbps using G.723 and CRTP to 53 Kbps

using PCM.

One of the key issues in VoIP is ensuring that the network has sufficient capacity to

send the VoIP packets. While users will accept a few second delays in Web traffic or e-mail,

most people cannot tolerate delays in voice conversations over about 250 milliseconds (a

quarter of a second). Therefore, VoIP is only practical in networks that enable Quality of

Service ( QoS ) routing at the IP layer (see Chapter 5) and ideally matching QoS at the data

link layer. Most organizations that deploy VoIP, therefore, use IP with QoS and also have

policy-based VLANs at the data link layer (see Chapter 8).

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