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Understanding the Basics of RoIP and VoIP

By better understanding the terminology & considerations involved, organizations can ensure the IP network meets their requirements. This article shares some IP data basics for using internet protocols to transmit radio voice and data.

Radio over IP (RoIP) is a generic term that describes the application of Voice over IP (VoIP) on two-way radio networks. So all digital radio networks with IP connectivity use RoIP. For radio fleet operators, RoIP can lower costs, improve reliability and increase interoperability
in the following applications:

  • access to remote radios,
  • point-to-point, point-to-multipoint and cross-band connection,
  • radio-to-dispatch console links,
  • radio-to-telephone gateway,
  • by reducing cabling.

All of these applications utilize IP infrastructure – the IP backbones that make up both local and wide area networks.

VoIP
Originally designed as a telephone replacement, VoIP allowed large service providers to connect many users for voice communications on demand. VoIP and RoIP use the same mechanism to transmit voice or audio data. However, radio has additional capabilities when it comes to signalling and control. These include:

  • Push-to-Talk (PTT) indicates the radio should transmit and must be synchronized with the audio,
  • Carrier Operated Switch (COS) from the radio generates a busy signal or keys up another transceiver,
  • Type of call in progress (Broadcast, Group or Individual calls),
  • Radio identification,
  • Advanced radio features (Stun, Revive, Remote Monitoring, Emergency),
  • Channel profile change (can include frequency and other operating parameters).
Image courtesy of iStock Images

Image courtesy of iStock Images

VoIP Standards
Using VoIP standards to implement RoIP provides several advantages. They improve vendor-independent interoperability and integrate radio networks with existing phone and voice systems. They are compatible with off-the-shelf voice recorders, routers, firewalls and network tools.

There are many standards that are relevant to voice and multimedia communications, none of which are interoperable. However, SIP and RTP are common protocols. DMR (AIS) and APCO P25 (DFSI, ISSI and CSSI) standards use both RTP and SIP.

Session Initiated Protocol (SIP)
SIP is a standard protocol used to set up and disconnect VoIP calls. It determines IP addresses of remote devices and UDP Port Numbers for RTP and negotiates which features can be used. No audio is transmitted via SIP. SIP devices include:

  • devices that create and manage a SIP session, such as a radio-to-VoIP gateway or a SIP VoIP phone,
  • servers, such as a Registrar Server that stores device registration information in a database, and Proxy Server that re-routes requests or messages. (A SIP server may be both a registrar and a proxy server at the same time.)

Real Time Protocol (RTP)
RTP provides data transport for audio and video over IP in telephony, video teleconferences and television. Designed for end-to-end, real-time transfer of stream data, it prioritizes “real time” over reliability, often compromising audio/video quality for immediate delivery.

RTP can send voice and radio information, but requires more system configuration and is less flexible to operate. However it can be used for unicast or multicast links, and can manage issues that occur when sending data over packet-switching networks. For example RTP can detect lost data packets, and can rectify packets arriving in the wrong order, or with variable packet delivery delays (packet jitter).

Common VoIP Issues
Delay. Two types of delay occur within packet switched networks:

  • constant delays or “latency”
  • variable delays or “jitter”

To compensate for jitter, the end device keeps a buffer of audio data so that it can continue to play audio even if the next packet is late. This in itself can be another source of delay.

Radio PTT means that these systems tolerate delay better than telephone systems, provided PTT signalling is synchronized with audio signalling so that no leading or trailing syllables are cut off.

Echo
Without delay, echo is not usually an issue for system users. Once a delay is added, the echo may be audible. Echo can be eliminated using Digital Signal Processors (DSP) to process the audio, where echo cancellation algorithms filter the echo signal from the received audio.

Packet Loss
Sometimes data packets don’t make it to the receiver. Fortunately, digital voice is generally intelligible with quite high levels of packet loss as VoIP systems incorporate Packet Loss Concealment (PLC) algorithms to compensate. Well-designed RoIP systems can provide acceptable, intelligible audio with packet loss of 10%.

On wired networks (LANs or WANs), packet loss generally only occurs as a result of overload or congestion. However, WIFI networks or Microwave links are subject to loss of individual packets.

System designers and operators need to be aware that their communications may depend on other devices that may be less tolerant of packet loss.

A word about CODECs
Codecs (Coder/Decoders) transport voice through packet switched data networks. Software codecs take digitized audio and encode it to send over the network, compressing the audio, so less data is required to transmit it. Codecs for VoIP compression by discarding some audio information, potentially reducing audio quality.

For real time communications like VoIP, codecs need to process audio in real time, so it only has 20 milliseconds to encode or decode 20 milliseconds of audio. By comparison, Codecs such as MP3 don’t have to encode audio in real time so they achieve better compression and audio quality.

Unfortunately, current digital radio protocols and standards do not specify widely-implemented or freely-available Codecs (even those that use RTP and SIP).

One solution is to continue to use gateways that are designed to transmit the added digital radio data and functionality. This approach provides additional benefits:

  • radios with Point-to-Point protocols can be shared amongst a number of operator positions,
  • only one vocoder (located in the gateway) is required, shared amongst a number of users,
  • radios can be accessed simultaneously by different dispatch consoles,
  • gateways provide flexibility to change protocols to meet future demands.

The above is an extract from the Omnitronics White Paper ‘Understanding RoIP Networks’. This white paper delves further into the above considerations and also discusses and compares the range of IP Connectivity methods organizations can choose from to best suit their needs.

For further information, go to www.taitradio.com

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Paul Whitfield is Team Leader of Software Engineering at Omnitronics.