Passthrough Fundamentals

With only a few minor variations that are discussed at the end of this section, a passthrough call is treated the same as a VoIP call from a voice gateway perspective. The human voice sample that is processed by the gateway on a VoIP call is simply replaced with the modulated data used by faxes and modems.

For both voice and passthrough calls, a process known as pulse code modulation (PCM) converts an analog signal to an equivalent digital representation. This digital signal is what is packetized and transported over the IP network. Figure 4-1 illustrates how PCM works.

Figure 4-1 Pulse Code Modulation

Amplitude

Time

Analog Signal Carrying Modulated Data

Time

Analog Signal Carrying Modulated Data

OOOOOOO1. OOOOOOOO

PCM Encoded Signal

PCM first filters out all frequencies greater than 4000 Hz because the majority of human speech occurs in the 300 Hz to 3200 Hz range. Nyquist's theorem specifies that to accurately reconstruct a signal, it must be sampled at twice the highest frequency of that signal. Because a band-limited 4000 Hz filter is used, the original analog signal must therefore be sampled at 8000 times a second.

Sampling is merely taking an amplitude reading of the original signal. This process is known as pulse amplitude modulation (PAM). PCM takes it one step further than PAM and quantizes the signal.

Quantization is the process of breaking up the continuous amplitude spectrum into discrete intervals. Each quantization level is assigned an 8-bit codeword. Therefore, there are 256 distinct amplitude levels with a unique 8-bit codeword assigned to each one. Figure 4-1 illustrates an analog signal encoded as digital PCM through the process detailed in the preceding paragraphs.

For a VoIP call, there are a number of codecs to choose from. A codec integrates with PCM and defines a particular encoding scheme to be used in the conversion of an analog signal into its digitally encoded version. Codecs vary in bandwidth requirements, voice quality, and computational requirements.

For example, voice is commonly transported over the WAN using high compression codecs, such as G.729 (8 Kbps) or G.723 (5.3 Kbps/6.3 Kbps). Because these codecs are optimized for human speech, they do a great job in preserving speech quality while at the same time offering a high compression rate that saves bandwidth.

However, the tones used for modem and fax negotiation are very different in nature from human speech and in many instances not even in the same frequency range. This makes it difficult to optimize a high-compression codec for both voice and fax/modem tones. These high-compression, speech-optimized codecs distort modulated data signals to the point where modems and fax machines are unable to communicate successfully.

Although codecs such as clear-channel codec or 32 Kbps compressed G.726 may transport modem or fax tones in-band, this discussion will be limited to using G.711 as the VBD codec. This is because it is overwhelmingly the most frequently used and the only one officially supported for Cisco passthrough features. G.711 is a 64 Kbps uncompressed voice codec that implements a PCM scheme that is compatible with modulated data.

Rather than the uniform quantization seen in Figure 4-1, the G.711 codec uses a nonuniform quantization scheme, known as companding. This has the effect of a greater concentration of quantization levels at the lower amplitudes, and conversely the higheramplitude values have quantization levels assigned more sparsely. Figure 4-2 shows this uneven distribution of quantization levels for the amplitude.

Figure 4-2 G.711 Companding of a PCM Signal

Amplitude 11111111 -i 11111110 -

00000000

11111111 11111110

00000001 00000000

1 Tt

11=1

1 Tt

11=1

'}

1 4' 4

[•*•

Uniformly Quantized PCM Signal

Uniformly Quantized PCM Signal

Quantization Noise liiii iiiil

Non-linearly Quantized (Companded) PCM Signal

Companding is appropriate for voice because the majority of human speech occurs at the lower end of the amplitude spectrum. This allows for greater fidelity and improved voice quality for the lower-amplitude signal, which is the bulk of human speech.

Two types of companding are used in G.711: ^-law and a-law. They are similar in many ways, but ^-law has a bit less distortion for lower-amplitude signals, whereas a-law has a greater dynamic range than ^-law. The biggest difference is that ^-law is used by North America and Japan, whereas a-law is used by the rest of the world. It is important to note that these two companding schemes are not compatible, and any calls between countries that use different companding types have to convert between the two.

The major impairment that results from analog-to-digital conversions, such as PCM, is the introduction of noise. Any difference between the actual amplitude value of the original signal and its assigned value of the closest discrete quantization level will introduce quantization noise.

As Figure 4-2 highlights, the nonlinear distribution of quantization levels used in companding will produce less quantization noise at the lower-amplitude signals and more quantization noise at the higher-amplitude signals. This keeps the signal-to-noise ratio (SNR) relatively constant over the entire signal amplitude range.

Now that the process of digitally encoding an analog signal has been discussed, it is important to understand how these PCM samples of modem, fax, and text data are packetized for transport over the IP network. Like in any data communication, the payload is independently encapsulated by the corresponding protocol of each of the OSI layers. For example, Figure 4-3 is an illustration of how PCM modulated data samples would be encapsulated for transmission over an IP configured Ethernet interface.

Figure 4-3 Encapsulation of an RTP Packet over Ethernet

Ethernet

IP

UDP

RTP

RTP Payload

Header

Header

Header

Header

(PCM Modulated Data)

Layer 5 (Session Layer)

Layer 5 (Session Layer)

Layer 4 (Transport Layer)

Layer 3 (Network Layer)

Layer 2 (Link Layer)

Because of the real-time nature of the transport of the PCM-encoded modulated data, it is important to take a closer look at the RTP header. From Figure 4-3, you can see that the G.711 encoded samples of voice-band modulated data become the payload of an RTP encapsulated packet. Figure 4-4 illustrates the RTP header, which is defined in RFC 3550.

All real-time traffic that is encapsulated in RTP maintains the timing characteristics of the original analog signal via the Timestamp field in the RTP header. Likewise, the PCM encoded samples can be played out in the same order as they were received because of the Sequence Number field. For this discussion, the most important field is the Payload Type.

Figure 4-4 RTP Packet Header

Fixed Header Fields

Optional Header Fields (not typically seen with voice codec payloads)

V: Version, identifies the version of RTP.

P: Padding, if set the packet contains one or more additional padding octets.

X: Extension, indicates the presence of a header extension field.

CC: CSRC Count, specifies the number of CSRC fields that follow the fixed header.

M: Marker, defined by a profile with the intention of allowing significant events such as frame boundaries to be marked in the packet stream.

Payload Type: Payload Type, defines the format of the RTP payload and determines its interpretation by the application.

Sequence Number: A counter that increments by one for each RTP packet sent while the receiver uses it to detect packet loss and out of sequence packets.

Timestamp: Reflects the sampling instant of the first octet in the RTP packet.

Synchronization Source (SSRC) Identifier: Identifies the source with a unique, random identifier.

Contributing Source (CSRC) Identifier: Identifies up to 15 contributing sources for the payload contained in the RTP packet.

Extension Header: A variable length extension to the RTP header that allows individual implementations to experiment with new payload-format-independent functions.

1 byte_|_1 byte_i_2 bytes

V

P

X

CC

M

Payload Type

Sequence Number

Timestamp

Synchronization Source (SSRC) Identifier

Synchronization Source (SSRC) Identifier

Contributing Source (CSRC) Identifier

Extension Header

The Payload Type field identifies the type of data being carried in the RTP packet. This defines how the packet will be interpreted and dealt with by the remote side. Table 4-1 shows the Payload Type values that are defined in RFC 3551.

Table 4-1 Payload Type Values

Payload Type

Payload Encoding

Payload Type

Payload Encoding

0

PCM |-law

25

CelB

1

reserved

26

JPEG

2

reserved

27

Unassigned

3

GSM

28

nv

4

G.723

29

Unassigned

5

DVI4

30

Unassigned

6

DVI4

31

H.261

7

LPC

32

MPV

8

PCM a-law

33

MP2T

9

G.722

34

H.263

10

L16

35-71

Unassigned

11

L16

72-76

Reserved

12

QCELP

77-95

Unassigned

13

CN

96-127

Dynamic

14

MPA

dyn

G.726 (40 kbps)

15

G.728

dyn

G.726 (32 kbps)

16

DVI4

dyn

G.726 (24 kbps)

17

DVI4

dyn

G.726 (16 kbps)

18

G.729

dyn

G.729D

19

Reserved

dyn

G.729E

20

Unassigned

dyn

GSM-EFR

21

Unassigned

dyn

L8

22

Unassigned

dyn

RED

23

Unassigned

dyn

VDVI

24

Unassigned

dyn

H.263-1998

Table 4-1 shows a number of dynamic and unassigned payload types. The dynamically assigned portion of this range is what is primarily discussed in this chapter. Unless explicitly configured on the gateway, Cisco uses the dynamic and unassigned payload type values shown in Table 4-2 by default.

Table 4-2 Dynamic and Unassigned Payload Types Commonly Used by Cisco

Table 4-1 shows a number of dynamic and unassigned payload types. The dynamically assigned portion of this range is what is primarily discussed in this chapter. Unless explicitly configured on the gateway, Cisco uses the dynamic and unassigned payload type values shown in Table 4-2 by default.

Table 4-2 Dynamic and Unassigned Payload Types Commonly Used by Cisco

Default Dynamic and

Unassigned Payload Type

Payload Encoding

90

RFC 2198 Passthrough Redundancy

96

Cisco Fax Relay Switchover

97

Cisco Fax Relay Switchover ACK

100

Named Signaling Event

101

Named Telephony Event

119

Cisco Text Relay

121

Cisco RTP DTMF Relay

122

Cisco Fax Relay

123

Cisco CAS Payload

125 Cisco Clear-Channel

125 Cisco Clear-Channel

When using passthrough, a voice gateway identifies the contents it is transmitting as simply PCM (PT=0 for G.711 -law or PT=8 for G.711 a-law). Thus, it makes no distinction within the RTP packet between a voice call and a modem/fax/text call.

As Figure 4-5 highlights, the fax/modem modulated data is transparently carried over the IP network, and the data is never demodulated within the IP infrastructure. This is the principal difference between passthrough and relay, which is covered in Chapter 5, "Relay."

Figure 4-5 Fax and Modem Passthrough

End-to-End Modulated Data

IP Network

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