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Trends in Technology: AoIP Codecs for Remote Broadcasts

A look at several hardware and software options currently on the market

The ability to generate unique content for your station has never been as important. A simple way to do it is by taking “live shots” from a news or sporting event. With the advent of hardware that uses the public internet for transmission, followed by software apps that emulate that hardware, “remotes” are easier than ever. In this article we’ll look at hardware and software solutions.


Comrex is a long-established manufacturer of codecs, including portable units, along with other telephone-related products. It was an early developer of AoIP codecs. Opal is a recently developed device that enables remote guests to connect back to the studio by clicking on a link delivered in a message originating from the radio station.

Opal works by activating the Opus encoder built into commonly used browsers, including Chrome, Firefox and Opera. Support for Safari, on iPhones with iOS 11, along with newer versions of the Mac OS. As a result, the remote user can connect to Opal from any computer or mobile device with one of those browsers installed. Opus transmits high-fidelity, low-delay audio in both directions. All the remote guest needs to transmit audio is a browser and a microphone.

At the radio station, the half-rack width Opal device makes the actual send and receive connections to the studio facility via balanced XLR connectors (analog or AES). Aside from its Ethernet connector, it also has a nine-pin DIN connector for remote contact closures.

In order to use Opal you will need a static, public-facing IP address and a domain name associated with that IP. “In order to keep the web browsers from complaining, we need to provide SSL/TLS security. In order to have Opal provide this security, it needs a URL with a real domain name instead of an IP address,” according to the company.

The Telos Z/IP One is a 1 RU rackmount IP codec designed for remote broadcasting. It includes a range of codecs including AAC-ELD, AAC-HE, AAC-LD, MPEG 4 AAC, MPEG 2 AAC, MPEG Layer II, G.711, G.722 codecs, plus linear audio and optional aptX Enhanced coding. Z/IP One supports SIP 2.0 protocol and conforms to N/ACIP standards; it also works with VoIP devices and connects to compatible SIP PBXs. A complement of I/O, including Livewire AoIP, analog and AES/EBU, is standard. Other salient features:

• Works with wired and wireless IP connections including Wi-Fi, WLAN (with matching Wi-Fi stick);
• Telos’ Agile Connection Technology (ACT) automatically senses network conditions and adapts codec performance to provide the best possible audio;
• Dual Ethernet ports for separate streaming and control;
• Livewire, analog and AES/EBU I/O standard;
• Easy browser setup via built-in web server;
• “Push Mode” for one-way network transmission; “Multiple Push Mode” for audio distribution to multiple destinations;
• Distributed Z/IP Server directory service, with multiple geolocations, lets you connect to other Z/IP One devices without the need for an IP address and also provides NAT traversal support;
• Transparent, time-aligned RS-232 channel for remote control or metadata, e.g., RDS;
• Time-aligned 8-bit parallel GPIO port for signaling and control.

[Related: “Summer of Products 2017 Surfs On “]

Tieline’s ViA is a remote codec that supports IP, ISDN and POTS, plus data aggregation and redundant IP streaming. ViA can connect over IP with dual Ethernet LAN ports, or two USB modems, or an internal LTE module, or use built-in Wi-Fi. The user can insert an optional POTS or ISDN module to allow the codec to connect over alternative network transports, thus supporting the configuration of primary and backup connections over different network transports as required, or simply using them as an IFB circuit.

ViA integrates with Tieline’s Merlin and Merlin Plus audio codecs to transmit high-fidelity, full-duplex stereo program audio with a separate bidirectional IFB circuit. ViA’s features include:

• Three analog mic/line inputs (Input 1 supports stereo AES3 digital audio, or mono AES42 microphone) and a stereo analog line input and stereo digital in/out via micro-USB or S/PDIF;
• Touchscreen matrix editor routes any input to any output; customize headphone mixes via touchscreen for three headphone outputs
• EQ, compression and limiting on all inputs; output AGC available on all outputs;
• Bidirectional mono, stereo or dual mono connections;
• Uncompressed PCM audio plus the low-delay, cascade-resilient aptX Enhanced algorithm; LC-AAC, HE-AAC v1 and v2, AAC-LD, AAC-ELD v1 and v2, Opus, MPEG II, MPEG Layer III, Tieline Music and MusicPLUS, G.722 and G.711.

The IP Codec from WorldCast Systems codec developer APT is a single rack unit, stereo, full-duplex AoIP codec, featuring dual XLR inputs and outputs (analog and AES), with dual IP ports (for configuration and management, or redundant streaming). Multiple coding algorithms are standard, including linear PCM 16-/24-bit, apt-X Enhanced 16-/24-bit, ÊMPEG1/2 Layer II, MPEG 1 Layer III (MP3 for decoding only), MPEG2/4 AAC-LC -LD -ELD, ÊMPEG2/4 HE-AAC v1/2 and digital MPX@128/192FS (optional).

The IP Codec automatically detects the correct algorithm on the receive end. Some of its other features include WorldCast’s SureStream (which supports multiple redundant streams to one destination); network security features for firewall compatibility; four opto-coupled inputs and corresponding relay-isolated outputs for remote signaling as well as SNMP and VLAN support; NTP based content time alignment; Stream Forwarding; ScriptEasy application developer, and, configuration via a built-in web interface, along with alarm and event logging. The system has recently added support for directly connected 3G/4G modems to ensure easy connectivity.


Some common applications that work with both iPhones and Android phones are Linphone, Luci and QGolive. Remote talent can do live reports, cut-ins, or even good old-fashioned car dealer remotes, using nothing but their smartphone on the cellular telephone network. Let’s take a closer look at these applications.

Linphone is an open-source SIP Phone, available on these mobile platforms: Apple iOS 8 to 10 (ARM v7, ARM 64); Google Android 4.1 to 7 (ARM v5 to v7, x86); BlackBerry OS10 (ARM v7); and Windows 10 UWP: mobile and desktop (ARM v7).

Linphone will also work in a desktop environment (GNU/Linux, Mac OSX and Windows).

Among its features we have the following: audio (and HD video) calls; multiple calls management (pause and resume); call transfer; audio conferencing (merge calls into a conference); instant messaging; display of advanced call statistics; echo cancellation; call quality indicator; and, support for secure communications (zRTP, TLS, SRTP). Advanced features for Linphone include support for the following audio codecs: Opus, SILK, Speex, G.722, AMR-WB (G.722.2), AMR-NB, GSM 6.10, ILBC, G.729, ISAC, BV16, G.711, and Codec2; integration with push notification (requires compatible SIP server); ICE support (RFC5245) to allow peer to peer audio and video connections without media relay server; call handover across network access type change (start a call in Wi-Fi and continue in 3G); the ability to configure multiple proxy accounts with different transports (UDP, TCP, TLS); and finally, IP v6 (dual stack and v6-only support). LinPhone is available for free on Android and iTunes.

Luci will work on the following platforms: All iOS devices with iOS8 or newer; Apple MAC computer with Mac OS 10.7 minimum; PC/laptop/netbook with Windows XP, Windows Vista, Windows 7, Windows 8, Windows 10; Linux computers PC/laptop/netbook; and Android phones and tablets.

Among its features are use of RTP over UDP low-delay streaming, in a duplex fashion, so that it includes a return channel; N/ACIP compatibility; one-way Shoutcast/Icecast streaming; the ability to record while broadcasting; the ability to play prerecorded material while broadcasting; stream cloning Ñ that is, sending redundant streams via 3G, Wi-Fi and Ethernet simultaneously; support for codecs MP2, AAC, AAC-HE, AAC-LD, AAC-ELD, AAC-HE v2, G.711, G.722, ULCC, and linear, as well as a 24-bit ULCC audio codec, 44.1384 kHz sample rate; and, ASIO support on Windows.

Current Luci versions include Luci Live for iPhone and Android, priced around $300. If you want to start out spending less money, consider Luci Live Lite Ñ that version doesn’t include the record, edit and FTP functions and limits the codec choice to G.722 or Luci’s ULCC codec. Other than that, it retains the same functionality as the more expensive version. Cost of this version is around $30.

QGoLive is a software-to-software solution that does not require the purchase of hardware at the studio (receive) end. The transmit app runs on iOS or Android devices; the receiver application runs on PC or Mac. An Android-based hardware receiver, with balanced XLR inputs and outputs, can be used on the receive end so that there is no need to tie up a computer.

The primary purpose of QGoLive is to replace the live shots that are typically broadcast in phone quality because the reporter has just arrived on the scene and has not had an opportunity to set up equipment for a broadcast quality live shot.

QGoLive has three major functional aspects: live, playlist and scripts. The “live” mode allows the user to connect to the receiver at the radio station after logging in. To connect the user hits the “play” button in the center of the screen, which will send audio to the station receiver and send cue audio to the app. The app will run in the background so you can use other nonaudio apps while broadcasting with QGoLive.

The “playlist” function allows the user to play out cuts recorded and edited in external apps (such as the TwistedWave audio editor or any audio program that can open its output MP3) in another program (or a browser that can open downloaded files in another app).

QGoLive allows you to write or dictate scripts which the talent can read while live. The user can also insert edited audio directly into the script for playback during a live report. And, imported cuts can also be added to and played from the scripts tab.

“Live shots” from a news or sporting event make unique content for your station. Say what you will about the cellular telephone system, but it has made an enormous difference in the way we carry out remotes. The amount of time, expertise and certainly hardware costs are all greatly reduced when compared to what they were just a few years ago.

Doug Irwin, CPBE AMD DRB, is vice president of engineering at iHeartMedia in Los Angeles and a technical advisor to Radio World. His Trends in Technology columns will appear here regularly.