This library written by Minitool defines an advanced and popular IP codec that is used by many advanced countries, ipDTL. After reading the below content, you will have a thorough understanding of it and know which to choose between ipDTL and Source Connect when broadcasting and recording.

What Does ipDTL Mean?


  • IP codecs are used to send video and audio signals over an IP network like the Internet. “IP” here stands for “Internet Protocol” while “codec” is short for “encoder/decoder” or “compressor/decompressor”.
  • In broadcasting engineering, a remote broadcast is broadcasting done from a location away from a formal television studio and is considered an EFP (Electronic Field Production).
  • Voice-over is a production technique where a voice, which is not part of the narrative (non-diegetic), is used in radio, television, theatre, filmmaking, and many other presentations.

ipDTL is a replacement for the classic Integrated Services Digital Network (ISDN) while backward compatible with ISDN.

The Origin of ipDTL

ipDTL was invented by Kevin Leach, a former BBC sound engineer. He makes use of the open-source codec Opus to create since it first became available in Google Chrome to create ipDTL. By accessing via a web browser, Opus enables higher audio quality than ISDN. Kevin released the tech in 2013 as an initially free service for radio stations but shifted a subscription model shortly after.

Relying on Opus of ipDTL, it is possible to apply audio quality of 72 kbit/s mono for voice contributors, 320 kbit/s for outside broadcasts with music, and 3 Mbit/s (video at 1080p) for contributions on TV programs.

Tip: Opus is a lossy audio coding format developed by the Xiph.Org Foundation and standardized by the Internet Engineering Task Force (IEFT). It is designed to general audio and efficiently code speech in a single format while remaining low-complexity enough for low-end embedded processors and low-latency enough for real-time interactive communication.

ipDTL Technologies

ipDTL adopts both Web audio and WebRTC technologies and is designed mainly for Blink-based browsers, such as Chrome and Opera. ipDTL runs on all platforms except for iOS where those browsers are supported.

ipDTL uses Opus codec for audio and VP8 for video. The supported bandwidth is up to 320 kbit/s (stereo) for audio and up to 3 Mbit/s (1080p) for video.


  • WebRTC is a free and open-source project providing web browsers and mobile applications with real-time communication (RTC) through simple application programming interfaces (APIs).
  • Web audio, also known as HTML5 Audio, is a subject of the HTML5 specification that incorporates audio input, playback, synthesis, and speech to text in the browser.
  • Blink is a browser engine developed as part of the Chromium project with contributions from Google, Opera Software, Adobe System, Microsoft, Facebook, Intel, Samsung, IBM, etc.
  • VP8 is an open, royalty-free video compression format created by On2 Technologies as a successor to VP7 and owned by Google since 2010.

Connections are established point-to-point and DTLS-encrypted. Where a point-to-point connection is not possible, Traversal Using Relays around NAT (TURN) relay servers are used to route the audio. TURN servers are available in the US, the UK, Australia, Japan, and Brazil, with an independent backup system being maintained at connections can also be made via a special URL that allows users to use another account to connect with it.

ipDTL uses a proprietary signaling method. Yet, it still supports SIP for interoperability with other applications and devices like Media5-fone and Comrex Access. ipDTL can also transcode between any two of Opus, G.711, and G.722. ipDTL supports interoperability with legacy ISDN hardware through cloud-based bridging servers.


  • Media5-fone was a VoIP softphone that uses the Session Initiation Protocol (SIP).
  • G.711 is a narrowband audio codec originally designed for use in telephony that offers toll-quality audio at 64 kbit/s.
  • G.722 is an ITU-T standard 7 kHz wideband audio codec operating at 48, 56, and 64 kbit/s. Its corresponding narrow-band codec based on the same technology is G.726.
  • ipDTL powers hybrIP, a talk show system that allows screening calls using another computer.

How Does ipDTL Work?

In most home recording setups, there is a Microphone > Audio Interface > Computer style arrangement. While in ipDTL’s web page, your Audio Interface turns out to be the input source. At the other end, the opposite is reflected and you pipe the ipDTL interface into your recording software. Isn’t it simple to use?

ipDTL vs Source Connect

Source Connect was released in the 2000s and came as a plugin to the Pro Tools system. It worked in the same role as ISDN effectively but over the Internet, in the same way adopted by Skype. Network speeds were slow then, so it incorporated a catch-up algorithm that meant that you may hear the live quality as quite low. Yet, the data could catch up in the background. Therefore, it ended up with good quality.

Though Internet speeds are much faster today, Source Connect is still a choice for voice artists. Source Connect’s top-end packages are still very expensive and you are probably only found in high-end studios. Yet, Source Connect also has an entry-level package called Source Connect Now, which is free though with certain limitations.

While ipDTL has similar offers to Source Connect with the only difference that everything is browser-based. Compared with Source Connect, ipDTL has some advantages. Firstly, equipped with an ISDN bridge, if you want to do a remote session with a large studio, you only have to pay for the cheap ipDTL interface. Then, it will bridge automatically from the source studio to the ISDN interface of the destination studio.

Also, with ipDTL, you can broadcast video at the same time so both parties can see each other whilst recording. It is ideal for directorial sessions and can be used as an internet phone system.

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