ITEXPO begins in:   New Coverage :  Asterisk  |  Fax Software  |  SIP Phones  |  Small Cells
  November 2006
Volume 1 / Number 6   

SIP & Mobility:
Bringing Radio into the 21st Century

By Doug Hall, Feature Articles


The adoption of SIP in the mainstream telephony world has numerous and proven benefits: open standards, a rich feature set, and readily available equipment from multiple vendors. Now SIP brings all of these benefits to land mobile radio (LMR) users plus entirely new interfacing and interoperability options. This article explores how to interface radio equipment with SIP infrastructure and explains the features radio users can expect to enjoy.

Analog telephone users have been a part of the SIP world for a while now, thanks to the Analog Telephone Adapter (ATA) which interfaces their analog telephone equipment to a SIP network. Intrinsic within POTS phones are a number of characteristics that don’t correlate with the IP network realm, such as loop current, ring voltage, and inband signaling. The ATA resolves these peculiarities, allowing analog telephone users to make and receive SIP calls unaware of the activity transpiring under the ATA’s hood. Likewise, radio brings its own set of idiosyncrasies to the SIP world.

Most radios, such as those used by public safety agencies, are half-duplex devices, meaning that they can’t simultaneously transmit and receive. These radios require user intervention to change from receive to transmit, usually via a Push-to-Talk (PTT) switch. Telephones, being full-duplex by nature, have no such requirement. Furthermore, with radios, any signaling via DTMF or other tones is usually done in-band. What’s needed is an ATA-like device that gracefully handles all the nuances of radio communications.

An example of such a device is the ARA-1, part of a family of ARA™ (Analog Radio Adapter) devices from Raytheon’s JPS Communications. The ARA-1 Radio-to-SIP interface allows radios to become SIP endpoints. Like an ATA, the ARA-1 includes a 10/100BaseT Ethernet interface to the SIP network. However, on the analog side, its only similarity to the ATA is that its users can communicate, blissfully unaware of how the interface does its magic.

The ARA-1’s analog port provides an interface for the signals commonly found on radio equipment: transmit and receive audio and PTT and Squelch signals. (Editor’s Note: Squelch involves turning off the radio’s speaker when there is no signal on the tuned frequency, and turning on the speaker again when the received signal is sufficiently strong.) Adjustments are provided for the audio levels and the type of squelch indication (valid received signal present) that will be used. These adjustments handle differences in interface requirements between radio makes and models. Some radios expect mic-level input (TX) audio, while others expect linelevel audio.

A radio’s squelch indication output (COR) signal can be active-high, active-low, or just plain non-existent. When no signal is provided, the ARA-1 can derive one by examining the incoming receive audio with a voice-activated switching (VOX) algorithm. Knowing when a valid signal is being received by the radio allows the ARA-1 to preserve network bandwidth by not streaming audio packets when there is nothing valid to send. Or, if required, the unit can also stream silence.

The final radio interface signal is its PTT input. Let’s assume that a SIP phone is connected over the network to an ARA-1 and associated radio. SIP phones don’t have PTT switches, so again there’s nothing for the ARA-1 to work with other than the incoming network audio. Whenever the SIP phone user talks, the radio will transmit this speech, and when the phone user stops, the radio switches back to the receive mode. The ARA-1 provides an adjustable “hangtime” which keeps the transmitter active during momentary pauses between words and syllables, as it’s not good to try to move too quickly between the receive and transmit states.

If only it were that simple. . . but as any telephone user knows, there are other noises on the phone line besides speech. Background noise, static, and a variety of “pops” and “clicks” of unknown origin — all are audible and could inadvertently and momentarily activate the transmitter. The ARA-1 resolves this problem by providing a Voice Modulation Recognition (VMR) algorithm. VMR examines the frequency content of the incoming audio, seeking concentrations of energy that are characteristic of human speech. It then goes a step further by ignoring those concentrations of energy that don’t occur at the same rate as syllables in speech. This allows the algorithm to reject noise and other non-speech signals, so the radio transmitter is activated only when someone is actually talking.

Like an ATA, the ARA-1 can accept an incoming call (a SIP Invite) and be programmed to answer automatically after a specified time. A much more difficult problem to address is how does a radio user initiate a call? It’s easy to visualize a call being put through by a radio with a DTMF keypad, but the majority of LMR radios don’t have one. The only user controls available on most of the handheld radios commonly used by public safety agencies are a PTT button, a speaker, and volume and channel knobs. Fortunately the ARA-1 can accommodate outgoing calls from radios limited to these few simple control mechanisms.

Before we go any further, we need to properly visualize a radio-to-SIP call. An analog telephone user who makes a call through an ATA simply picks up the phone, dials the extension, and is directly connected. In contrast, a radio user with DTMF capability transmits DTMF “digits” over the air to another radio. This is the one that’s cabled to the ARA-1 that provides the SIP network interface. Any return network audio is transmitted over the air and received by any number of radio users “in the field” who have radios of the right frequency. So an ARA-1 interfaces an entire radio system, not a single user.

The ARA-1 can be programmed to initiate calls based on pre-programmed DTMF sequences received from those radios equipped with DTMF capability. For the rest, the ARA- 1 provides a feature called COR Cadence signaling. A radio’s COR output signal goes to an active state whenever a signal is being received by the radio. Radios in the field without DTMF capability can activate their PTT switches at a preset rate; the resulting RF pulses are detected by the radio cabled to the ARA-1. This radio’s COR output pulses in time with the incoming RF and the ARA-1 detects and compares these pulses with an internal “speed dial” that matches these COR cadence sequences with stored SIP PBX (News - Alert) extensions or IP addresses.

For example, pressing and releasing the PTT button 5 times within a 2 second period could instruct the ARA-1 to dial a pre-programmed extension. The numbers to be dialed are programmable, and the timing of the COR cadence can be adjusted to suit specific radio requirements. DTMF and COR Cadence sequences can also be used to disconnect a call.

Installing and provisioning an ARA-1 is similar to installing an ATA, though the radio interface process is more involved than plugging in a telephone’s RJ11 cable. There are a number of network, SIP, and radio settings which can be viewed and optimized with a web browser. Ready-made ARAto- radio cables are available from JPS Communications for over 100 different radio models, or the user can construct a cable based on information in the ARA-1 manual.

The ARA-1 includes some additional user-friendly features such as the ability to “speak” its IP address and other network settings over the radio interface. Initiated by a preset DTMF or COR Cadence sequence, this feature helps determine the address when it has been assigned by DHCP. On the network side, STUN support is provided for operation behind a NAT device.

While the ability to interface individual radios as SIP endpoint devices has tremendous utility, features already existing within SIP allow a valuable bonus: Radio interoperability! Multiple radios can be interconnected through a SIP conference call just as regular telephones could be. A SIP conference call might include a VHF radio, a UHF radio, and an 800MHz trunked radio, all connected to the SIP phone system through ARA-1 devices. Any radio traffic received on the 800 MHz radio would be retransmitted to the VHF and UHF radios. The radios can join the conference through the use of DTMF or COR Cadence signaling, or they can be added to a conference by an operator. In short, they enjoy all the benefits that a SIP phone system brings to ordinary SIP phones, such as call forwarding, call logging, and any existing or yet-to-beimagined feature that’s applicable to radio communications (or can be adapted to it).

As more and more organizations build out their SIP infrastructure the applicability of this technology to radio communications will become more and more apparent. The old saying “If you can’t beat ‘em, join ‘em,” is certainly appropriate. Why try to make telephones that act like radios, when with SIP and devices such as the ARA-1 we have the best of all worlds: Since everyone already knows how to use a telephone, why not make radios act like telephones?

Doug Hall is the Senior Scientist at JPS Communications (www.jps.com), a wholly-owned subsidiary of Raytheon Company.



Today @ TMC
Upcoming Events
ITEXPO West 2012
October 2- 5, 2012
The Austin Convention Center
Austin, Texas
The World's Premier Managed Services and Cloud Computing Event
Click for Dates and Locations
Mobility Tech Conference & Expo
October 3- 5, 2012
The Austin Convention Center
Austin, Texas
Cloud Communications Summit
October 3- 5, 2012
The Austin Convention Center
Austin, Texas