Chapter 4: Passthrough

When the public switched telephone network (PSTN) was initially constructed, voice communication was the primary goal. However, as data communications such as fax, modem, and text became more important, they also were made to work over the PSTN using special protocols and transport methods. Today, with VoIP taking the place of the PSTN, voice communication is still the primary objective, and specific protocols and procedures are again needed to transport fax, modem, and text communications.

One such feature that voice gateways can implement to transport modem, fax, or text telephony traffic is passthrough. This transport mechanism is the easiest and simplest way for a voice gateway to pass modulated data.

For the most part, passthrough works just like a normal voice call. The voice gateway receives an analog waveform from the modem, fax, or text device and encodes it using an appropriate coder/decoder (codec). These encoded samples are then encapsulated and transported over the packet network using the Real-Time Protocol (RTP).

You will also commonly hear passthrough referred to as voice-band data (VBD) by more recent literature and many of the specifications. These two terms are used interchangeably for the remainder of this book.

This chapter provides an in-depth look at how passthrough operates and the different ways that it is implemented to transport modem, fax, and text data. Specifically, this chapter discusses the following topics:

• Passthrough Fundamentals

• NSE-Based Passthrough

• Protocol-Based Pass-Through for Fax

• Text over G.711

• A Future Look at ITU-T V.152

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

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 higher-amplitude 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

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: µ-law and a-law. They are similar in many ways, but µ-law has a bit less distortion for lower-amplitude signals, whereas a-law has a greater dynamic range than µ-law. The biggest difference is that µ-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

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

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

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

When using passthrough, a voice gateway identifies the contents it is transmitting as simply PCM (PT=0 for G.711 µ-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

When the passthrough feature is initiated on a Cisco voice gateway, additional events take place to ensure that the modulated data is successfully transported across IP. The most important event is known as codec upspeed.

Codec upspeed makes sure that the passthrough call uses a low-compression codec such as G.711 µ-law or G.711 a-law. Passthrough calls start out in the beginning as regular voice calls. This means that the call could be using a high-compression codec such as G.729. However, when the passthrough feature is initiated, this codec is changed to G.711 in what is termed codec upspeed.

In addition to codec upspeed, another change also takes place in the Cisco voice gateway when it switches into VBD mode and prepares for a passthrough call. To make the IP path as transparent as possible, the DSP disables Voice Activity Detection (VAD). VAD is a bandwidth-saving feature that sends packets only when there is voice detected during the call. If VAD were to remain enabled for the passthrough calls, signals could be clipped, negatively affecting the data being transported.

Slight changes are also made to the DSP’s jitter or playout buffer. While in voice mode, the playout buffer is adaptive and constantly adjusts to changing network conditions. However, during passthrough mode, the playout buffer becomes fixed to an optimum value for the call. For a more comprehensive discussion of what a jitter buffer is and the specifics of how it behaves during a passthrough call, see the “IP Troubleshooting” section of Chapter 12, “Troubleshooting Passthrough and Relay.”

After the detection of certain tones by the DSP, the switchover to passthrough is signaled in one of two ways. One is NSE-based passthrough signaling, which involves the exchange of Named Signaling Events (NSE) packets between the gateways. The other is protocol-based passthrough signaling, in which a direct negotiation occurs in the protocol stack of the call signaling protocol.

NSE-Based Passthrough

When passthrough is configured on a voice gateway, it takes the modulated data from a fax, modem, or text device and transparently transports it in the media stream as PCM samples encapsulated in RTP.

The terminating gateway (TGW) always switches to NSE-based passthrough mode first by detecting the appropriate tone from the answering modem or fax machine. This tone is the 2100 Hz CED from a standard fax machine or the 2100 Hz ANSam tone from a modem or SG3 fax machine.

When the TGW detects this tone, it undergoes a passthrough switchover, including a codec upspeed to the VBD codec (G.711). In conjunction with this switchover to NSE-based passthrough, the TGW also transmits an in-band signal in the media stream to the originating gateway (OGW). In this message, the TGW signals to the OGW to switch into passthrough mode. This signal is communicated using NSE packets.

NSEs are a Cisco proprietary message that are sent as part of the RTP stream and are identified using a payload type of 100 in the RTP header by default. Despite being a proprietary message, the NSE packet format is the same as for standards-based Named Telephony Events (NTE), described in RFC 2833. Figure 4-6 shows the NSE/NTE packet format.

Figure 4-6

NSE Packet Format

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Note – The NSE payload type is configurable on a Cisco IOS voice gateway to be any value between 98 and 119. The default value is 100.

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The Event ID field uses Cisco-defined events to signal in-band the coordination of a variety of tasks. Table 4-3 shows the NSE event numbers used for passthrough. Notice that NSE-192 is used by the TGW to signal to the OGW to go into VBD mode.

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Note – The Volume and Duration fields in the NSE packet will always be set to 0s for the discussions in this chapter. Only the event ID is pertinent.

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Table 4-3 Cisco NSE Event IDs Used for Passthrough

When VoIP calls are transitioned to fax and modem passthrough calls using NSEs, the NSE signaling occurs within the RTP media stream. The signaling protocol is generally unaware that this out-of-band messaging is even occurring. So, NSE-based passthrough is supported in just about all Cisco voice gateways implementing the common signaling protocols of H.323, Session Initiation Protocol (SIP), Media Gateway Control Protocol (MGCP), and Skinny Client Control Protocol (SCCP).

The procedures that a fax call and a modem call use to set up NSE-based passthrough are slightly different. Therefore, the cases of fax passthrough and modem passthrough are analyzed independently and in more detail.

Fax Passthrough with NSE

In this section, the information exchange and negotiations between a TGW and an OGW for an NSE-signaled fax passthrough call are covered. Figure 4-7 illustrates this message flow for a typical fax call.

Figure 4-7

Fax Passthrough Call with NSE Signaling

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Note – The NSE signaling flow shown in Figure 4-7 does not indicate a specific voice signaling protocol. NSE-based fax passthrough is compatible with H.323, SIP, MGCP, and SCCP, and the call flow from an NSE perspective is identical.

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When the calling fax machine places a call to the answering fax machine, the call is initially set up as a simple voice call between the voice gateways. At this point, with both gateways having been configured for NSE-based passthrough, the DSP on the TGW is listening for a 2100 Hz answer tone.

When the terminating fax machine answers the call, it transmits its 2100 Hz CED tone. The DSP on the TGW detects and recognizes this tone. This triggers a codec upspeed and a switch to passthrough or VBD mode on the TGW.

In addition, detection of the fax CED by the TGW triggers a notification by the DSP to IOS that a fax call has been detected. In response, IOS sends an instruction to the DSP to send an in-band RTP signal with payload type 100 to the OGW instructing it to upspeed its codec and switch to VBD mode, too. This signal is an NSE with event ID 192 (for example, NSE-192) and is shown in Figure 4-7. The OGW replies by sending an NSE-192 back to the TGW.

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Tip – In reality, more than one NSE message is passed between the gateways to protect against packet loss or other network problems. Typically, the NSE message is repeated three times. For the NSE-192 messages in Figure 4-7, three NSE messages are sent from the TGW to the OGW, and then three messages are sent in the reverse direction as a response. This is true of practically all NSE signaling messages, but for simplicity this is just shown as one NSE message in the NSE call-flow diagrams.

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At this point, both gateways have completely transitioned from voice mode to VBD mode. The fax negotiation will proceed normally as PCM packets over the IP network. The modulated data is end to end between the two fax machines.

Although not shown in Figure 4-7 because it is rarely seen, an NSE-194 may occur at the completion of the fax call if the fax machines do not hang up. An NSE-194 will occur after 4 seconds of silence and will switch the passthrough call back to voice mode. However, because fax machines usually immediately disconnect upon call completion, it is not necessary for the voice gateways to transmit NSE-194 messages.

Modem Passthrough with NSE

The basic difference between the fax passthrough feature and the modem passthrough feature is whether the 2100 Hz answer tone from the answering fax/modem contains phase reversals. If the answer tone contains phase reversals, the modem passthrough feature is engaged but an additional NSE message will be triggered. Devices with transmission rates faster than a typical fax machine, such as high-speed modems (using V.34 modulation speeds and higher) and Super G3 (SG3) faxes, send an ANSam tone that contains phase reversals.

Figure 4-8 shows the message exchange for NSE-signaled modem passthrough. Just like in the fax passthrough case, a normal VoIP call is first established, and then an NSE-192 is sent upon the TGW detecting a 2100 Hz answer tone. In the case of a modem, however, this 2100 Hz answer tone is typically an ANSam.

Figure 4-8

Modem Passthrough Call with NSE Signaling

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Note – As with NSE-based fax passthrough, the NSE call flow above remains the same no matter if the voice signaling protocol is H.323, SIP, MGCP, or SCCP.

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The in-band signal detection by the TGW’s DSP of the answer tone from the terminating modem will trigger the exact same codec upspeed and switchover to VBD mode seen in the fax passthrough case. The only difference is that in this case there is the additional detection of phase reversals in the answer tone. This phase-reversal detection instructs the DSP in the TGW to disable the echo canceller.

Also, this detection of an answer tone with phase reversals triggers a notification by the DSP to IOS that an answer tone with phase reversals was received. IOS responds with a command to the DSP to send an in-band RTP message to the OGW’s DSP to also disable its echo canceller. The message is an NSE packet with event ID 193 (NSE-193). As illustrated in Figure 4-8, the TGW sends an NSE-193 and that is replied to by the OGW with the transmission of another NSE-193 message.

If low-speed modems are used that do not implement an ANSam tone, the NSE call flow will resemble fax passthrough in Figure 4-7 rather than modem passthrough in Figure 4-8. Low-speed modems are typically defined as 14.4 Kbps, and below and like the modulations used for fax, any 2100 Hz tones that are used do not contain phase reversals. So, an NSE-193 would never be triggered in the absence of phase reversals, and the call would proceed with just an NSE-192 as in Figure 4-7.

Protocol-Based Pass-Through for Fax

The other method used to trigger a transition from voice mode to passthrough mode is in the messaging of the VoIP call signaling protocol. Although the call signaling protocol’s primary responsibility is to set up and tear down the VoIP call, it can also be used to transition to passthrough mode.

At this point, a distinction needs to be made between the term passthrough and pass-through. The term passthrough is what has been used until now when referring to the VBD feature in general and the mode where the passthrough switchover is signaled by NSE packets. This terminology will not change. However, to indicate a transition to VBD mode via messages in the voice call control protocol, the term pass-through is used going forward.

This terminology difference arises from a command-line interface (CLI) configuration convention used on Cisco IOS voice gateways. The modem passthrough nse command enables the VBD feature for fax and modem calls using NSE signaling. Thus when handling a fax call using NSE signaled VBD, this is commonly referred to as “fax passthrough” or “NSE-based passthrough.”

The command fax protocol pass-through indicates that a fax call is being handled by the signaling protocol. Therefore, if the VBD mode is signaled via protocol signaling messages, this is commonly referred to as “fax pass-through” or “protocol-based pass-through.”

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Note – The two configuration methods for passthrough are covered in detail in Chapter 9, “Configuring Passthrough.”

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There is no protocol-based pass-through for modems. Only fax calls can take advantage of pass-through using the call signaling protocol. Unlike NSE-based passthrough, which is triggered by a 2100 Hz tone, protocol-based pass-through can be triggered only by a fax V.21 preamble.

The principal reason to use the protocol-based signaling rather than the Cisco proprietary NSE-based signaling for a fax passthrough call is that it allows for interoperability with third-party voice gateways. NSE-based signaling is proprietary, so protocol-based signaling is the only solution for Cisco voice gateways to interoperate with voice gateways from other vendors.

The two fax pass-through implementations that are discussed involve the H.323 and SIP protocol stacks. H.323 and SIP are the only supported call signaling protocols for pass-through signaling on Cisco voice gateways.

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Tip – There is not a protocol-based pass-through solution for the MGCP protocol stack on Cisco voice gateways due to protocol conflicts and reliability concerns. NSE-based passthrough should be used instead.

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Fax Pass-Through with H.323 Signaling

H.323 fax pass-through occurs when a VoIP call is set up using the H.323 protocol stack, and then H.323 messages are used to transition the call to pass-through mode. Figure 4-9 illustrates pass-through using the H.323 call signaling protocol.

Initially, the VoIP call is established using the H.323 signaling protocol. When the terminating fax machine answers the call it plays a 2100 Hz CED tone. However, unlike NSE-based passthrough, this tone does not trigger a pass-through switchover.

Following the CED tone, the terminating fax machine begins the transmission of the DIS and optional NSF and CSI fax messages. These messages are flagged with a V.21 preamble. The V.21 preamble is the signal that triggers the transition to pass-through.

Upon detection of the V.21 preamble by the TGW, a passthrough switchover is initiated with the H.323 protocol stack. H.245 request mode messages signal the switchover to the G.711 codec while H.245 logical channel messages are responsible for closing the previous voice mode logical channels (typically using a high-compression codec such as G.729), opening the new logical channels for the G.711 codec, and acknowledging the same.

Upon completion of the H.245 message exchange, the fax pass-through session is established. The fax call should now be able to complete successfully.

Figure 4-9

Fax Pass-Through Call with H.323 Signaling Protocol

Fax Pass-Through with SIP Signaling

With SIP as the call signaling protocol, fax pass-through is very similar to H.323. SIP sets up a normal VoIP call, and then when the V.21 preamble is detected, SIP handles the transition to pass-through mode. Figure 4-10 illustrates fax pass-through with SIP as the call signaling protocol.

Figure 4-10

Fax Pass-Through Call with SIP Signaling Protocol

After the initial VoIP call is established using the SIP signaling protocol, a V.21 preamble detected by the TGW triggers the switchover to pass through. However, unlike NSE-based passthrough, the SIP protocol will handle the pass-through transition rather than NSE messages.

A SIP re-INVITE is issued by the TGW to transition to pass-through mode. The most important occurrence here is the codec upspeed to G.711. The pass-through parameters enclosed within the SIP re-INVITE are confirmed in the SIP 200 OK message. A SIP ACK from the TGW completes the pass-through switchover.

Text over G.711

Text telephony poses a major challenge to both NSE-based and protocol-based passthrough mechanisms. There is not a common tone implemented by the different text telephony protocols that can be focused on to trigger a passthrough switchover. So, to pass text telephone protocols in a passthrough mode, a manual version of passthrough must be configured. This is referred to as text over G.711.

With text over G.711, dedicated voice connections (using dial-peers on IOS gateways) are configured specifically for the use of text telephone protocols. These text-specific configurations on the voice gateways force all calls across this connection to use the G.711 codec while disabling silence suppression or VAD.

From a call-flow perspective, there is no need for any switchovers using NSEs or the call signaling protocol because the call is set up from the beginning to handle text telephone traffic. Figure 4-11 illustrates text over G.711.

Figure 4-11

Text over G.711 Call

Text over G.711 is signaling protocol independent. So, voice signaling protocols such as H.323, SIP, MGCP, and SCCP are compatible with G.711 over text because these protocols can be configured on voice gateways to set up simple G.711 VoIP calls with silence suppression/VAD disabled.

A Future Look at ITU-T V.152

Both NSE-based passthrough and protocol-based pass-through have their positives and negatives. NSE-based passthrough works for modem and faxes, but this solution is Cisco proprietary and excludes Cisco voice gateways from interoperating with other vendors. On the other hand, protocol-based pass-through offers third-party interoperability, but this is only an option for fax calls. Modem calls do not work with protocol-based pass-through.

The ITU-T has a specification titled V.152 that details a standards-based form of passthrough or VBD for faxes and modems over an IP infrastructure. Although relatively new, V.152 presents a passthrough solution that possesses the relative strengths of NSE-based passthrough and protocol-based pass-through without any of the previously mentioned drawbacks. Furthermore, V.152 supports text telephone protocols, too.

V.152 specifies two methods for transitioning to VBD mode from normal voice mode: payload type switching and SSE (State Signaling Event) messages. Payload type switching is similar to protocol-based pass-through. VBD parameters are negotiated in the call signaling protocol stack and a predefined VBD RTP payload type is established. V.152 recommends that the VBD payload type be a dynamic one.

VBD payload type switching can be accomplished with a number of different call signaling protocols. V.152 defines specific VBD parameters for SDP (used by SIP, MGCP, and H.248 gateways) and H.323.

SSE messages can also signal a V.152 VBD switchover. SSEs are defined in Annexes C, E, and F of V.150.1. These messages are similar to NSEs in that they notify other gateways of signaling events that are usually modem and fax related.

V.152 declares SSEs as optional for transitioning to VBD mode. If one or both gateways do not support SSEs, the payload type switching method must be implemented.

Although Cisco voice gateways do not currently support the recently ratified ITU-T V.152 standard, this specification is the next logical step for handling fax, modem, and text telephony in passthrough scenarios.

Summary

Passthrough or VBD is a transport method for passing modulated data such as fax, modem, and text telephone protocols over an IP network. Transitioning to passthrough mode involves additional changes beyond a normal voice call. In Cisco voice gateways, this passthrough switchover can be signaled in one of two ways: NSE-based passthrough or protocol-based pass-through.

NSE-based passthrough uses Cisco proprietary NSE messages between the originating and terminating voice gateways to transition to passthrough. This signaling method is compatible with the common call signaling protocols of H.323, SIP, MGCP, and SCCP, but interoperating with third-party gateways is not possible.

Protocol-based pass-through is available only for fax calls because it is triggered by a V.21 preamble that is not present for modems. Unlike NSEs that communicate over the RTP media stream, protocol-based pass-through uses the voice signaling protocol to handle the pass-through transition. Only the call signaling protocols of H.323 and SIP support this signaling method, but interoperability with third-party gateways is possible.

For handling text telephone protocols, a manually configured version of passthrough must be used that is referred to as text over G.711. Text over G.711 can be used with any call signaling protocol or third-party gateway that supports the G.711 codec.

The ITU-T V.152 specification is a standards-based form of passthrough or VBD. Although this is a relatively new specification that is not widely implemented, it offers the benefits of both NSE-based passthrough and protocol-based pass-through without the current drawbacks.

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