Switching to VoIP

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Voice Channels

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Transport

The T1 carrier versus VoIP

The T1’s 24 DS0 channels are each never-ending streams of digitized voice informa-

tion. In reality though, the T1 circuit itself is one big stream of binary digits that uses

TDM to divide the T1 into those 24 DS0 channels. Each is assigned a time slice of

the big stream, and each time slice is further divided into frames, as shown in

Figure 6-1. All voice T1s use the same amount of bandwidth no matter how many

calls are in progress—roughly 1.54 mbps. Trunking with T1s is very stable and pre-

dictable as a result.

VoIP frees the system builder from requirements traditionally imposed on the lower

OSI layers of the network. In a T1, the transport and data link layers are defined

together as a bundled carrier, and you have to use the G.711 PCM codec on all the

channels, yielding 24 simultaneous voice channels in the available bandwidth. VoIP

lets you pick and choose the codec, packet interval, and transport technologies you

want and thus gives you ultimate control. Using the G.729A codec and a T1, you

could conceivably trunk hundreds of calls at once.

Unlike G.711 traffic on a T1, VoIP’s “carrier” is TCP/IP. So VoIP can traverse Ether-

net, T1s, DSL lines, cable internet lines, POTS lines, frame relay networks, virtual

private networks (VPNs), microwave radio, satellite connections, ATM, and just

about any other link. If IP can go there, VoIP can go there—just with varying levels

of quality.

Table 6-1. VoIP codec characteristics

Codec Algorithm

Bandwidth used

for sound

Packet interval

Voice bits per

packet

Processing

intensity

G.711 PCM 64 kbps 20 ms 1,280 bits Low

G.726 ADPCM 32 kbps 20 ms 640 bits Medium

G.728 CELP 16 kbps 10 ms 160 bits High

G.729A CELP 8 kbps 10 ms 160 bits High

GSM RPE

a

or CELP

a

Regular pulse excitement, yet another sound encoding/compression technique.

13 kbps 20 ms 160 bits Depends upon

algorithm used;

CELP is higher

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120

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Chapter 6: Replacing the Voice Circuit with VoIP

All VoIP systems to date use equal-sized frames of the same fidelity

during each call. You’ll never have a sample rate of 8 kHz change to 12

kHz during a single session the way you could with a variable-rate

MP3, for example.

Voice packet structure

The layered appearance of a VoIP packet is similar to that of other types of net-

worked applications that run within the TCP/IP protocol: the lower layers encapsu-

late the higher layers recursively.

The lowest layer, shown leftmost in Figure 6-3, is the Internet Protocol (IP) packet

header. It contains routing information so that the packet can be handled correctly

by the devices responsible for carrying it across the network. It also contains a flag

that indicates which protocol of the TCP/IP suite this packet is carrying: TCP, UDP,

or something else. Voice packets are almost always UDP. Among other things, the IP

header may include a Type of Service flag that allows routers and switches to treat it

with a certain priority based on its sensitivity to delay. At a minimum, the IP packet

header is 160 bits in length.

The payload of the IP packet is the UDP packet, whose header is 64 bits long. Its first

32 bits contain the source and destination ports of the UDP traffic it carries in its

payload, along with 8 bits for optional error checking and 8 bits for describing the

length of its payload in multiples of 8 bits.

The payload of the UDP packet is the RTP packet, whose header is 96 bits long. It

contains information about the sequence and timing of the packet within the greater

data stream.

Real-Time Transport Protocol

The Real-Time Transport Protocol (RTP) defines a simple way of sending and receiv-

ing encoded media streams

*

in connectionless sessions. It provides headers that

afford VoIP systems an easy way of discriminating between multiple sessions on the

same host. Remember that the codec merely describes how the digitized sample is

encoded, compressed, and decoded. RTP is responsible for transporting the encoded

sound data within a UDP datagram. RTP was designed for use outside the realm of

telephony, too: streaming audio and video for entertainment and education are com-

mon with RTP.

RTP supports mixing several streams into a single session in order to support appli-

cations like conference calling. It doesn’t, however, provide adequate controls for

defining multiplexed voice pathways that are normally associated with telephony,

like trunks. This is the responsibility of the softPBX and its signaling protocols.

* When certain signaling protocols, such as IAX, are used, RTP may not be employed to packetize the voice

stream. IAX has its own built-in voice-packetization capability.