26
Land Mobile Radio and Professional Mobile Radio: Emergency First Responder Communications
Jerry D. Gibson
26.2 Unique Requirements of Emergency First Responder Communications
26.3 Conventional Land Mobile Radio in the United States
26.4 Project 25 Land Mobile Radio in the United States
26.5 Professional Mobile Radio in Europe
Broadband Backbone in the United States: LTE
26.1 Introduction
While mobile communications have become ubiquitous throughout the World due to the widespread availability of cellular networks, wireless access points, and satellite links, there is another set of wireless communications systems that is critical to our well-being, particularly in the event of natural and man-made disasters. This chapter is about the wireless communications systems used by Emergency First Responders, such as firefighters, police, and other governmental agency personnel. Perhaps surprisingly, these systems are based on much more rudimentary communications technologies than are the commercial systems that we use in our everyday lives. Similar to military communications systems, communications systems for emergency first responders have some very specific special requirements that are not available in most commercial systems and we develop these special requirements here. These emergency communications systems also have a unique evolutionary path that is far from what many consider as the quite revolutionary path blazed by digital cellular and other commercial wireless technologies in the past 25 years [1–3].
It is well known to the public that communications systems relied upon by the public often fail during natural disasters, as in, for example, Hurricane Katrina, the Haiti earthquake, and wildfires near urban areas, wherein switches are flooded, cell sites are toppled, or cellular traffic outstrips cell site capacities. What may be disconcerting is that emergency first responder communication systems have exhibited their own failure modes, such as lack of interoperability, loss of relay stations, and poor voice quality due to background impairments, and that these systems have very limited capabilities for data, video, and other multimedia traffic. The various agencies of the World are attempting to correct these very serious shortcomings, but research and development, planning, training, equipment options, and budgets are all sorely lacking.
We begin our treatment with an overview of the unique requirements of first responder communications systems and the different scenarios that are envisioned when designing such systems. We then provide discussions on land mobile radio (LMR) Systems in the U.S. and professional mobile radio (PMR) Systems in Europe and elsewhere. We next turn our attention to the planned expansions of emergency responder networks in the United States, particularly with respect to the broadband nationwide connectivity via digital cellular LTE. We also discuss how some standards bodies are attempting to address the needed capabilities. Finally, we outline the shortcomings and the ongoing challenges in first responder communications.
26.2 Unique Requirements of Emergency First Responder Communications
An important point to realize at the outset is that all public safety communication problems are not equal and can greatly differ. In recent years, public safety personnel have responded to terrorist acts, hurricanes, prairie fires, mountain wildfires, tsunamis, earthquakes, and infrastructure failures, such as the Minneapolis bridge collapse [4,5].
The MESA Project recognized the variety of public safety responses that can occur and created the Scenario Class Diagram in Figure 26.1 to help in their classification [6]. As examples of using this diagram, one can classify the Minneapolis bridge failure as Urban/Disaster/Single Spot, whereas a wildfire in mountainous terrain would be classified as Rural/Emergency/Wide Area. Another classification that is useful is time span. The Minneapolis bridge collapse had intense communications for a relatively short period of time (a few hours), whereas a mountainous wildfire can go on for days or weeks and require sustained, coordinated communications of all kinds.
During these events, literally hundreds of responders may be connected, and so large talk groups must be supported, plus there must be a quick-response push-to-talk (PTT) capability so that anyone on the call can comment. Another requirement is for quick response radio-to-radio communications, that is, direct communications between any pair of first responders. For voice communications, these requirements are not generally available in commercial digital cellular or WiMAX systems, without what is called “trunking.” Trunking means that the channels are shared and usually that communications must be handled by a switch, an example of which would be the base station/mobile switching center in digital cellular systems.
FIGURE 26.1 Characterization of emergency scenarios. (Adapted from Project Mesa; service specification group-services and applications; statement of requirements executive summary. MESA Organizational Partners (ETSI, TIA), 2005. [Online]. Available: www.projectmesa.org.)
Also important in these events is the support of low speed, but very responsive, data communications. The key term in characterizing first responder communications, whether voice or data, is “mission critical.”
A different type of requirement that equipment vendors emphasize is “hardening” of the handsets and other first responder communications equipment. That is, the equipment must withstand substantial amounts of physical abuse due to the environments encountered by first responders. This hardening requirement spills over to the systems themselves, in that a certain level of coverage needs to be maintained even in a catastrophic event. In addition to the required level of geographical coverage that may be needed anywhere an emergency occurs, the coverage must also include the capability to support unexpected spikes of traffic. Of course, these requirements are very different that digital cellular, which can easily suffer from lack of coverage in some areas and which can also experience dropped or blocked calls when cell sites are saturated with traffic. While these latter capabilities of coverage wherever needed and the ability to handle sudden spikes of traffic are often touted for public safety communication systems, mission critical communications by first responders has suffered substantial loss of coverage and blockage of calls in many recent emergencies.
The overall view is that public safety communication networks need to support worst-case situations, while commercial systems often operate on the best effort premise.
26.3 Conventional Land Mobile Radio in the United States
The largest-scale approach to improving first responder communications in the United States is Project 25 (P25), begun in 1988, and designed around the then existing analog FM communications systems. P25 was established to develop and standardize digital radio methods for LMR emergency response applications with the cooperation of the Association of Public Safety Communications Officials (APCO) and the Telecommunications Industry Association (TIA) [7,8], although the driving force is APCO, which is a group of vendors without a government or organized user group providing independent oversight and guidance.
Land mobile radio in the United States has evolved from a system based on narrowband analog channels to systems based on digital modulation and voice compression in the last 20 years. However, the channelization and basic problem has not changed dramatically, and in fact, it is very important to emergency personnel and government agencies that certain functionalities remain intact. We provide an overview of the operation of an LMR analog system from which today's systems are evolving. We use Figure 26.2, adapted from Reference 3, as the focal point for discussion.
The transmissions shown in Figure 26.2 represent voice conversations transmitted using analog frequency modulation (FM) of carrier signals at the frequencies indicated. The consoles shown represent dispatchers for the two frequency groups. Direct mode operation (DMO) is also indicated at the top of the figure at a different carrier frequency to avoid interference. DMO does not go through the base station or switching and therefore delays are minimal, plus the individuals communicating using these units only need to have a good transmission path between them, and do not have to have a good communication link with the base stations. In many first responder communications scenarios, such as in tunnels or buildings or in remote locations, communications with a base station cannot be maintained but communications between units is critically important.
In contrast to the multiple access schemes used in digital cellular or wireless access points for the Internet, the basic channel access scheme for conventional LMR is called “radio discipline”; that is, communications should be limited to short voice calls of 6 s duration, and only information that is essential for personnel safety and emergency response coordination is supposed to be communicated. This type of communications using large talk groups allows everyone using that carrier frequency to “be on the same page” but congestion can occur as well. Users gain access to the channel by push to talk (PTT) [9], and PTT response times are desired to be on the order of less than a second, but in emergency situations with large talk groups, it may be difficult for a user to gain access to the channel. It is thus clear that latency should be minimal, and the analog modulation method does not add significant latency in the processing of the voice communications since the voice signal directly frequency modulates the carrier.
FIGURE 26.2 Conventional analog first responder communications. (Adapted from R. I. Desourdis et al., Emerging Public Safety Wireless Communication Systems, Artech House, Inc., Norwood, Massachusetts, 2002.)
Analog LMR systems can have inefficient use of allocated spectrum and so the concept of trunking can also be employed. Trunking is where access to a channel using PTT is no longer direct, but the request is submitted through a control channel to a controller for the allocation of a frequency for the communications. If all goes well, the delays are not prohibitive, although the call initialization has additional latency compared to direct PTT access.
To address more demand for communications channels and to incorporate expanded data services as well, efforts were undertaken to move toward digital communication technologies. There are a number of approaches used for digital communications in public safety communications, but two have come to dominate the first responder communications space, usually called “mission critical” applications. These technologies are associated with two standards, Project 25 and TETRA [10,11]. We develop both of these systems in the following sections.
26.4 Project 25 Land Mobile Radio in the United States
Conventional analog LMR systems do not necessarily require much of a standard since they consist of simply FM transmission of voice on a known collection of frequencies. However, the move to digital communications implies a host of possibilities for modulation in a given band, for ways to represent the voice signal digitally, for data transmission, and for system access. Project 25 is a standard established in the United States to specify the needed requirements. Among the goals of Project 25 are to maintain the same radio channels and allocated bandwidth and also to retain the user experience as much as possible. To make more efficient use of the spectrum, Project 25 also intends to double the number of channels in 25 kHz for Phase I, that is, 12.5 kHz per channel, and then implement 6.25 kHz bandwidth channels in Phase II. Different modulation methods are used as these channel allocations evolve.
The overall block diagram of P25 radios is shown in Figure 26.3 [12]. Following the diagram, we see that the input speech undergoes analog-to-digital conversion and is then passed to a speech coder. This codec has to operate at a bit rate that takes into account the overall allocated bandwidth per channel, the additional bits due to error control coding, and the modulation methods used over the channel. The speech encoder is discussed in detail in a subsequent section. Error control coding (channel coding) bits are then appended and all of the resulting bits are sent to the modulator for transmission. The demodulator has the usual components needed to recover the speech signal.
For Phase I, the allocated channel bandwidths are 12.5 kHz, so the final modulated signal must fit in this bandwidth. There are other desirable attributes as well, such as ease of demodulation, constant envelope transmitted signal, and backward compatibility with analog FM radios. The modulation method for Phase I is C4FM (Compatible 4-Level Frequency Modulation) [2], wherein each pair of bits at the input to the modulator have an assigned frequency deviation from the carrier, with 01 and 11 having +1.8 and −1.8 kHz deviation, respectively, and 00 and 10 having +0.6 kHz and −0.6 kHz deviation, respectively. C4FM has a constant envelope and therefore does not require linear amplifiers, which is a saving in cost and complexity. A block diagram of a C4FM modulator is shown in Figure 26.4. Not illustrated in Figures 26.3 or 26.4 is the multiple access method used for the 12.5 kHz channels within the 25 kHz band. For P25 Phase I, FDMA (frequency division multiple access) is used, thus splitting the frequency band in half. The voice codec used in P25 Phase I is IMBE at 4.4 kbits/s. The voice codecs are discussed in detail in a later section.
For P25 Phase II, the modulation methods are HCPM (hybrid continuous phase modulation) for the uplink (mobile to base) and HDQPSK (harmonized differential quadrature phase shift keying) for the downlink (base to mobile). There is another modulation method that can be used in the downlink for simulcast systems, and it is not developed here. A block diagram representing the modulator for these two methods is shown in Figure 26.5 [12], where CQPSK is used to denote Compatible Quadrature Phase Shift Keying. The differences between HCPM and HDQPSK are that HCPM has a constant envelope and HDQPSK is not constant envelope (thus requiring linear amplifiers) but it is also easier to demodulate. HCPM is vulnerable to intersymbol interference in addition to requiring a more complex demodulator.
FIGURE 26.3 Block diagram of P25 land mobile radio units. (Adapted from P25 Radio Systems, Training Guide, Daniels Electronics Ltd., 2004.)
FIGURE 26.4 The C4FM modulator. (Adapted from P25 Radio Systems, Training Guide, Daniels Electronics Ltd., 2004.)
FIGURE 26.5 Modulation method for P25 Phase II. (Adapted from P25 Radio Systems, Training Guide, Daniels Electronics Ltd., 2004.)
The multiple access method used for Phase II is a combination of FDMA to obtain the 12.5 kHz channels and then 2 slot TDMA on each of these bands to obtain the 6.25 kHz channelization. The voice codec used in Phase II can be AMBE or AMBE+2, both of which can operate at dual rates to be compatible with the lower bandwidth systems.
26.4.1 Voice Codec
The primary need for first responders on Land Mobile Radio is voice communications, and that was the driver for conventional LMR systems. Given the narrow frequency band allocated to each channel in LMR for digital operation, the incoming speech must be sampled and compressed or coded at a low bit rate. For P25 in the United States, the codecs used are the IMBE, AMBE, and AMBE+2 codecs, all of which are based upon the Multiband Excitation (MBE) coding method [12,13]. In P25 Phase I, the Improved MBE, or IMBE, codec at 4.4 kbits/s is used for speech coding and then an additional 2.8 kbits/s is added for error control (channel) coding. This 7.2 kbits/s total then has other synchronization and low-speed data bits incorporated to obtain the final 9.6 kbits/s presented to the modulator. For P25 Phase II, the total rate available for speech and channel coding is half of 7.2 kbits/s or 3.6 kbits/s, which is split as 2.45 kbits/s for voice and 1.15 kbits/s for channel coding.
Block diagrams of the IMBE encoder and decoder are shown in Figures 26.6a and b, and a flow chart showing all the steps in the calculations is shown in Figure 26.7. We describe the basic IMBE codec in the following.
The IMBE vocoder models each segment of speech as a frequency-dependent combination of voiced (more periodic) and unvoiced (more noise-like) speech. This ability to mix voiced and unvoiced energy is a major advantage over traditional speech models that require each segment of speech to be entirely voiced or unvoiced. This flexibility gives the IMBE vocoder higher voice quality and more robustness to background noise.
The IMBE encoder estimates a set of model parameters for each segment of the incoming speech signal, which consists of the speaker pitch or fundamental frequency, a set of Voiced/Unvoiced (V/UV) decisions, which are used to generate the mixture of voiced and unvoiced excitation energy, and a set of spectral magnitudes, to represent the frequency response of the vocal tract. The encoder computes a discrete Fourier transform (DFT) for each segment of speech and then analyzes the frequency content to extract the model parameters for that segment. These model parameters are then quantized into 88 bits, and the resulting voice bits are then output as part of the 4.4 kbps of voice information produced by the IMBE encoder.
FIGURE 26.6 Block diagrams of IMBE voice codec (a) encoder and (b) decoder.
After error control decoding to try and correct any bit errors that were introduced during the wireless transmission, the IMBE decoder reproduces analog speech from the decoded 4.4 kbps digital bit stream that is remaining. In particular, the model parameters for each segment are decoded and these parameters are used to synthesize both a voiced signal and an unvoiced signal. The voiced signal represents the periodic portions of the speech and is synthesized using a bank of harmonic oscillators. The unvoiced signal represents the noise-like portions of the speech and is produced by filtering white noise. The decoder then combines these two signals and passes the result through a digital-to-analog converter to produce the analog speech output.
DVSI has developed a Half-Rate (3.6 kbps) vocoder that has been proposed for use in P25 Phase 2. Designed as an extension of the current 7.2 kbps IMBE vocoder used in P25, DVSI's new Half-Rate vocoder operates at a net bit rate of 2.45 kbps for voice information and a gross bit rate of 3.6 kbps after error control coding. This represents a 50% reduction in bit rate as compared with the current 7.2 kbps IMBE vocoder used in P25 Phase 1.
DVSI has also introduced new Enhanced Vocoders for P25 based on DVSI's latest AMBE+2 Vocoder technology. These Enhanced Vocoders are backward compatible with both the standard P25 Full-Rate and proposed Half-Rate vocoders, while providing improved voice quality, better noise immunity, tone capability, and other new features. The Enhanced Vocoders significantly improve the voice performance of the P25 system, while facilitating the migration and interoperability between new and existing P25 equipment.
FIGURE 26.7 Flow chart of the IMBE coder and decoder.
For the particular application of interest, four characteristics of the voice codec must be considered carefully: (1) Coded speech quality and intelligibility at the codec operating rate, (2) Codec performance when the input speech is contaminated by noise or the codec is operating in a very noisy environment, (3) codec interoperability with other networks, which directly involves how the voice codec performs for tandem coding with the codecs in the other networks, and (4) complexity of the voice codec, since a lower complexity translates into reduced power consumption for the mobile device and also reduced cost of the mobile device.
First, there is the issue of coded speech quality and intelligibility at the chosen bit rate. For a bit rate in the range of 4–4.8 kbit/s, the IMBE/AMBE set of codecs achieve an MOS of 3.5–3.8 for clean speech input [14]. Since the IMBE/AMBE codecs are proprietary to Digital Voice Systems, Inc. (DVSI), these codecs are much less studied than other voice codecs, and so firmly establishing the MOS values achieved by these codecs is somewhat difficult. DVSI only tests these codecs infrequently and the results are not widely nor clearly documented and disseminated. The scores for the IMBE/AMBE codec as quoted by the vendor are quite good for the operating bit rate; however, the MOS values shown for other voice coding methods in comparison with the IMBE codecs appear low compared to the values obtained in many other independent tests.
Codec performance in a noisy environment is quite different than for clean speech, and this is where the IMBE/AMBE codec shave been shown to be very poor performers by tests done by the Institute for Telecommunications Sciences under the auspices of the U.S. Department of Commerce and by the experience of firefighters in the field [14]. The IMBE/AMBE vocoder has good clear speech performance for its low bit rate, but it is known to perform extremely poorly for noisy (PASS alarms, chainsaws, etc.) input speech and speech from inside a mask (Self-Contained Breathing Apparatus (SCBA) that is essential in firefighting) [14,15]. No fixes have been put forward to address this issue except to require additional training for users. Further, the IMBE/AMBE vocoder has not been fully tested in tandem voice communications connections that require transcoding at network interfaces, such as will be necessary when voice communications enters the LTE network [16–18].
The LMR codecs are standardized for the LMR application, and INMARSAT, but not for commercial VoIP or digital cellular systems [17,19]. Therefore, when voice communications is set up between an individual on an LMR connection and someone on a digital cellular system such as LTE, the voice must be decoded at the network interface and then recoded with the codec available in the other handset. This is called tandem coding and tandem coding degrades quality substantially, adds latency, and increases system complexity. Since the IMBE/AMBE codecs do not tandem well with other standardized codecs, this is a serious issue for voice communications outside of a public safety LMR network.
Voice codec complexity is important in the LMR environment because of cost, battery power usage, and battery weight. The IMBE/AMBE codecs are somewhat complex, and it would be preferable to employ a voice codec that exhibits minimal complexity both in terms of battery power usage and cost.
26.5 Professional Mobile Radio in Europe
The standard used for first responder communications in Europe and the United Kingdom is TETRA, originally Trans European Trunked Radio but now Terrestrial Trunked Radio, and TETRA includes a comprehensive set of standards for the network and the air interface. TETRA was created as a standard for a range of applications in addition to public safety. TETRA operates on 25 kHz channel spacing and is a fully digital system based on four timeslot TDMA to yield the 6.25 kHz channels. No analog mode of operation is available and there is no backward compatibility with conventional analog systems. The modulation method used is π/4 shifted Differential Quaternary Phase Shift Keying (π/4 DQPSK), which does not have a constant envelope. As a result, linearity in the transmitters and receivers is necessary. The symbol phase transitions are shown in Figure 26.8. The symbol shaping at the transmitter is square root raised cosine shaping with a 0.35 roll off factor [10,11].
For TETRA, the voice codec is based on code excited linear prediction (CELP) and the speech is coded at 4.567 kbits/s, or alternatively, if the speech is coded in the network or in a mobile handset, the AMR codec at 4.75 kbits/s is used [20]. Block diagrams of the TETRA encoder and decoder are shown in Figures 26.9a and b, respectively. Although it has been stated in some publications that the TETRA codec is older than the MBE codecs, the CELP structure is much newer and more widely studied and evaluated than the MBE class of codecs. Comparing the block diagrams of the IMBE codec in Figure 26.6 and the TETRA codec in Figure 26.9, it is immediately evident that these are two very different approaches to speech coding. The TETRA codecs achieve MOS values in the 3.25–3.5 range for clean speech. The TETRA codecs, although they use a very different coding method than IMBE, are also expected to be vulnerable in high-noise environments, especially at the low rates used in TETRA.
FIGURE 26.8 Symbol transitions in π/4 DQPSK.
The algorithmic delay of the TETRA voice codec is 30 ms plus an additional 5 ms look ahead. Such a delay is not prohibitive, but a more thorough calculation in the standard estimates an end-to-end delay of 207.2 ms, which is at the edge of what may be acceptable for high-quality voice communications. A round trip delay near 500 ms is known to cause talkers to talk over the user at the other end, thus causing difficulty in communications, especially in emergency environments [10,11].
The complexity of the TETRA codecs has been carefully studied, with the encoder complexity, including operations, ROM, and RAM factors, calculated as 11.923 MOPS and with the decode complexity calculated as 5.383 MOPS.
26.5.1 Broadband Backbone in the United States: LTE
To address the perceived need for high-speed data and video services, a National Broadband Plan (NBP) was released by the FCC to provide for the creation and establishment of a nation-wide interoperable public safety network. The NBP utilizes the 3 GPP Long-Term Evolution (LTE) technology [16].
By adopting LTE for the nationwide interoperable broadband emergency network, the FCC hopes to leverage a widely deployed standard and to save network installation costs by exploiting the efforts of commercial entities. By giving high priority to emergency communications on commercial LTE networks, connectivity can theoretically be provided during critical emergencies. Thus, LTE provides the broadband data while LMR/PMR continues to provide voice communications for first responders. However, this dual LTE/P25 approach ignores several subtle but potentially crippling future problems. One subtle limitation is that current LTE networks are data only in that voice over IP will not be available for sometime [16]. Until VoIP is available, voice will be carried separately over 2 and 3 G digital cellular. Additionally, future LTE VoIP will require transcoding at network interfaces with any LMR connection [16], since LTE does not support the LMR specified IMBE/AMBE vocoder. Further, transcoding experiments of IMBE/AMBE with the projected (but not standardized) LTE AMR codec [16], have not been performed. Transcoding of the IMBE/AMBE vocoder with 2 and 3 G codecs has similarly not been tested. Such transcoding can yield a substantial drop in voice quality and clarity, even under ideal conditions.
LTE and LMR are also poorly matched in that LMR has been a voice centric service emphasizing low latency push-to-talk (PTT) and group talk services where the group size may be 100 or more [9]. These offerings are not easily supported by LTE today. The support of peer-to-peer voice communications, called Direct Mode Operation (DMO), without the use of a base station or other trunking infrastructure is particularly important and this has not been addressed by the LTE broadband network proposal [9].
FIGURE 26.9 Block diagrams of the TETRA voice. (a) Encoder and (b) Decoder. (Adapted from ETSI, Terrestrial trunked radio (TETRA); speech codec for full-rate traffic channel; part 2: TETRA codec, ETSI EN 300-395-2 V1.3.1, 2005.)
FIGURE 26.10 One approach to combining commercial cellular and LMR. (Adapted from J. Facella, Current state of 700 MHz Broadband for Public Safety, Powerpoint Presentation, Harris Corporation, Nov. 19, 2010.)
These LTE-based systems only provide broadband backbone services, not voice, and do not address the serious shortcomings of current LMR/PMR systems, including limited interoperability, poor voice quality in disaster scenarios, and restrictive coverage in remote locations and/or where infrastructure has been damaged [21,22]. In addition, the proposed national broadband network based on LTE is not designed to the high-reliability performance standards and rapid response requirements of mission critical applications, is not hardened to natural disasters, is not available in remote locations and is not easily portable and reconfigurable[16].
Therefore, the evolving public safety communications network consists of two silo communications networks, LMR for first responders and commercial LTE for the national broad network. The plan is to “duct tape” these two silos together at the IP network layer. One concept for combining LMR and LTE broadband is shown in Figure 26.10 [23]. The combination of these two silos is not seamless, and each silo has significant limitations for the planned tasks. First, LMR has well-known serious drawbacks, including poor voice quality in high-noise environments and with breathing masks, the number of direct mode calls (peer-to-peer) on one channel using TDMA is limited to one, the LMR codecs are not interoperable with other wireless networks or with other public utility communications such as an electrical utility, and no broadband coverage. The commercial LTE networks are designed for an entirely different set of applications and do not support the standard direct mode voice, talk group voice fast call set up using push to talk (PTT), high reliability, and rural coverage that is needed for emergency first responders.
26.5.2 Evolution
There have been plans for some time to continue to develop the LMR standards in the United States, perhaps combining them in some way with TETRA. There is also a Project 34 effort that has been in planning for a number of years. The hope is to eventually extend broadband data services and video to LMR, while still maintaining the key voice communications features of LMR such as DMO and PTT with large talk groups.
The market for LMR and PMR is no more than 10% of that for digital cellular and other commercial wireless services, so the competition of the marketplace has not worked well here. The several long time system vendors and one voice codec vendor continue to dominate the market without significant competition in the United States. The TETRA system in Europe is more extensively investigated and standardized, plus there are compatibilities with GSM digital cellular systems that make TETRA at least a “more vetted” system than LMR in the United States.
There are efforts in some countries without LMR or PMR to develop such systems based on WiMAX technologies. Indeed, highly mobile WiMAX systems are being proposed in the United States in order to address catastrophic failures in emergencies or to extend broadband coverage to remote areas.
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