If you know nothing about ISDN (Integrated Services Digital Network) it is (was) a communication standard for the digital transmission of data, video and voice over the public telephone network conceived in 1986. Eight years later both Comrex (US) and Glensound (UK) launched ‘user friendly’ ISDN audio codec boxes that paved the way for voice actors to work from home; albeit with a hefty price tag. Three decades on and tele-communications standards have advanced greatly, and an average broadband connection is now 400 times faster than ISDN. BT (British Telecom) is ‘retiring’ ISDN in 2025 if not sooner in some parts of the U.K.
There is much ‘talk’ too in voiceover forums about what is going to replace ISDN when the ‘plug is finally pulled’, I wish I had a crystal ball to predict what will come to the fore from existing IP based solutions vying for supremacy or, for what technical advances might be on the horizon to make what we know about now totally redundant in the future.
Presently, as I type this, the contenders are internet browser based WebRTC (Web Real-Time Codec) virtual devices, proprietary (bespoke) software, the re-purposing of existing equipment and SIP (Session Initiation Protocol) - The latter being a commonly used telecommunications signalling software protocol that is being embedded into a new generation of audio codecs operating systems by well-established hardware manufacturers.
The internet browser-based codecs are ipDTL, Bodalgo Call, Source-Connect NOW, Cleanfeed, SessionLink Pro and Connection Open. These all make use of Google Chrome, or its many variants, to create a connection between two or more endpoints anywhere around the World. Most have additional features and paid-for premium services like access to ISDN. These products are relatively easy for the user yet, there is a reluctance from established production houses and industry professionals to migrate to or even try this tech as they fear the audio quality will not be as good as ISDN. The reality is the quality of the audio is as good if not better thanks to the use of an OPUS codec that offers low latency high quality audio.
The proprietary market is dominated by Source Elements with ‘Source Connect’ popular stateside in post-production film studios for the added functionality of syncing audio to video but it is also fast becoming the default go-to by smaller production studios working remotely with VO. The downside is that both ends require this premium priced software and unlike some of the browser-based offerings above there is no option to ‘send-a-link’ to the distant studio or talent to launch a connection. Again, it’s simple to use and delivers quality audio.
Source Elements also has a niche product called vISDN which repurposes redundant ISDN codec hardware with an additional ‘black box’ at the customer end and requires two independent internet connections. This service adds up to a whopping $3600 for a year’s subscription excluding the cost of calls and the broadband. It is currently only available in the U.S.
The most exciting development for voice talents is the use of SIP (Session Initiation Protocol) that was developed twenty years ago and has been the backbone of the U.K. and many other countries telephone networks enabling hundreds of thousands of phone calls to be made daily - seamlessly and quickly. Modern businesses with their network connected phones use the same SIP protocol to make and receive calls, and now broadcasters are migrating to local and wide area network (LAN/WAN) connected infrastructure the VO community and others can take advantage of using SIP if you have access to the internet; and who doesn’t these days?
So how does it work?
If you want to communicate through SIP, you need the following:
An Internet connection with sufficient bandwidth for voice communication - the average broadband connection is more than capable of providing this.
A SIP address/account from SIP.Audio for an enhanced service tailored to audio professionals; other providers are available too but are geared to enabling telephone calls. If you’re an existing ipDTL user you already have a SIP address - check your ‘Quick Start’ settings for the details - this address can be used by any SIP enabled studio, production house or fellow VO to call you. You will require a separate/new account if you purchase or already have a hardware codec that can be configured to use a SIP address.
A SIP client. For the purpose of this blog aimed at voiceover artists, audio production houses and studios this would be an audio codec from any of the following hardware manufacturers - Aeta Audio; AEQ; AVT; Barix; Comrex; Digigram; JK Audio; Prodys; Telos; Tieline; Wheatstone; Worldcast; Vortex; 2WCom. Look for a device that supports the OPUS codec for low latency high quality audio.
An audio production or pro-home studio. If you’re swapping an aged/redundant ISDN box with a new SIP enabled codec it’s like for like, albeit connected to your broadband router - a must as the new codec may be controlled via a webpage or bespoke software. If you have a basic home voiceover studio you might need to purchase additional equipment or seek help in reconfiguring your setup.
Studios/VO to talk to. Maybe this is the first item in the list that’s checked. You may have lots of contacts, but they need to be using SIP too if you want to call them. So, share your SIP addresses just like you do phone numbers and encourage your regular contacts to start using SIP enabled devices.
Cost. An annual subscription for a fully supported SIP.Audio address/account is £50 + VAT per annum for an individual/single user. The hardware costs start from £669 + VAT for a simple ‘black-box’ as pictured below; the uScoop (microScoop) from Aeta Audio. This codec has a generic web browser interface, however, a box at a similar price from AEQ, the Phoenix Mercury, uses bespoke Windows only software for control and configuration. Choose a codec with more functionality and the price increases. The Comrex BRIC-Link is £1200 + VAT while their Access is £2500 +VAT.
Finally, and quite importantly the cost of calling a SIP address is... FREE, I’ll repeat that just in case it didn’t register there is no charge to call a SIP address - it’s FREE!
I’ve been using an Aeta Audio uScoop for a few weeks and in my tests I've have connected with a variety of manufactures codecs using a number of different SIP addresses from various providers and with only one exception, which was down to a basic configuration error on my part, establishing a call between devices has been trouble free and the audio quality has been considerably better than ISDN would have been. I’ve also succeeded in connecting to and receiving calls from several mobile applications including Luci Live and Linphone.
There is a video here of me using the uScoop to connect to Cameo Productions in London.
My only observation would be that in use if you’re connecting with a remote studio and they’re calling your SIP address the Aeta uScoop just auto-answers and very quickly too, if you are dialling a destination you have to use the unit’s web interface where you can enter addresses directly or setup shortcuts for quick access. This interface is also the configuration panel for the unit, so you just have to be aware of what you’re doing when selecting the page TABs. I will feedback to Aeta that they might want to consider creating just a ‘dial pad’ application to bypass access to account and audio setting for greater ease of use.
The Aeta-Audio uScoop is supplied with a setup manual, Cat5 LAN network cable and a 12v power adapter, it can also be powered by Power-Over-Ethernet (802.3af POE); this is how I’m powering the device. Plugging the uScoop into my home/office network straight out-of-the box it found a network address from my broadband router and using that I was able to configure the unit. Nothing more needs to be done to make/receive the first call, other than connect a mix-minus audio input to and an output from the XLR connectors on the rear. The front panel has a USB socket for backing-up/restoring configuration data, a Power, Ready, Decoder and Left & Light audio led status lights.
By now, this blog should have given you plenty to think about some real food-for-thought. Already major institutions like BBC Radio, Global, Bauer and other broadcasters are using SIP enabled devices to move audio around within their networks; some are now extending this connectivity beyond their ‘firewall’! Independent production houses who are regular sources of audio content are also using SIP enabled devices and services to get contributors on-air; Cameo Productions are a prime example. You may have already read this article from iHeart Media where they have ‘ditched’ ISDN for SIP saying and I quote; ‘Endpoints don’t need to know each other’s IP addresses; they just “dial” a registered SIP address to start the connection process’ they go on with; ‘…and we see Opus codec (low latency high quality audio) and SIP as the future of interoperability in our industry’. What more needs to be written to persuade you that the future is here now, and it is SIP!
We are in a very unique position as VO talent, producers, production houses, and studios to influence how we start reconnecting with one another post ISDN, although we only have to take our lead from the broadcasters above. I can see that SIP will be as ubiquitous as ISDN once was and we will be able to connect with countless more devices than ISDN ever could.
So, how do we start this revolution – get a SIP address, get a SIP enable device or use SIP enabled software. And start spreading the word that you’re contactable via SIP.
Appendix.
If you already have a SIP address and SIP enabled codec or use ipDTL you can call the Aeta Audio uScoop in my office by dialling rogerwoodsip@sip.audio when connected you’ll hear a music stream encoded at 128k 48kHz Mono using the OPUS codec (unless I’m recording or editing and you’ll hear whatever I’m doing…).