Telephones: a weak link in remote broadcasting

Have you been listening to the quality of telephone calls on radio during this time of lockdown? Most of them are pretty bad.

Steve Ahern asks a range of technical experts about the causes and solutions to bad phone lines during this time. This is a long article, so we’ve provided a table of contents so you can skip around if you want to

I have been hearing lots of very bad telephone lines on radio since the lockdowns began and it started me thinking about the causes and possible solutions.

The more I thought about it, the more I identified multiple points of failure during these changed broadcasting circumstances.

These are some of the things I have heard:

  • Echo on the line
  • Crosstalk
  • Poor signal strength style mobile phone dropouts
  • Digital drop outs
  • Speed changes
  • Profanity Delay errors

This is what I think is happening:

  • Echo on the line – the presenters are using zoom or some other video/audio conferencing system but there is not enough echo cancelling being applied. Plus there is likely no mix-minus mic clean-feed capability in remote studios.
  • Crosstalk – this seems to me like the skype, zoom or other digital conference system is overloaded and there are artefacts from other calls on the line.
  • Poor signal strength style mobile phone dropouts – this is standard, but because there are more people using their mobiles from home, I suspect the cell towers in residential areas are overloaded because they are not equipped for peak load capacity during business hours, like towers in the CBD.
  • Digital drop outs – this could be the phone line, or more likely it is a data capacity issue on the line between the studio and the presenter’s house.
  • Speed changes – this is common when using skype, zoom, messenger, viber etc as the system tries to balance poor internet capacity by slowing or freezing the data stream momentarily, then catching it up once capacity improves to get back into real time. Bluetooth headphones can also be a weak point here are they sometimes distort the audio if there is interference.
  • Profanity Delay errors – After all the other factors are taken into consideration, there’s the profanity delay system to think about as well. Who can dump? Where from? Is the remote caller dumped too? How is the ramp back into delay handled? How does delay work on multiple transmission/stream paths with at least one more  source studio added to the mix?

All this made my brain hurt. So I asked some experts to explain their experiences and sought advice about these questions:

  • Are voice landlines still be best quality?
  • Is there really any such things as a landline any more in this era of NBN? Aren’t they all just data lines?
  • The next best option after landlines should be voice calls via mobile phones, but if mobile voice calls use the data stream of each cell, and the data capacity for voice calls is compromised by overloading of cell tower capacity, is there any benefit in mobile voice calls any more?
  • Do voice calls still get dedicated bandwidth on cell networks, or are they all just data now?
  • Non professional data call options include zoom, skype, whats app, messenger, etc. These are normally good because they can deliver better frequency response, but during this time there is so much data traffic that they are almost always interrupted in some way.
  • There are also some professional level options too, how do they work? If you have a professional system on one end, but the caller just has a plain old normal phone, is it any value?
  • Or, maybe it is not the phones at all, maybe it is unstable bandwidth from home studios being used be presenters remotely. For internet access the weak point is usually the last mile to the home. Providers had little incentive to fix this issue in the past, because most traffic was through big pipes in CBDs. Now it is dispersed.

Here’s what the experts told me.

ARN’s Technology Director, Joe Sexton
Well spotted, the mobile network congestion in particular was pretty bad in the early weeks of COVID and did provide a few challenges. We had two challenges: Internal and External
ARN upgraded the internal PABX system in late 2019 which gave us a flexible IP solution for users. For example, desktop phones could be diverted to personal phones or to a soft client on laptops. For the Executive and Technology teams we extended the ARN network to personal ISPs and ARN desktop phones could be used at home. Personally, this was a life saver as my mobile was having constant issues.
We also had live callers who were having similar issues and quickly moved to IP which took several forms.
We took an agnostic approach where we would take any Digital format that was of acceptable quality for radio broadcasting. Technology was the enabler for great content.

In the case of the Kyle and Jackie O show, we used the enhanced audio algorithms (optional) of Zoom for an interview with Chris Hemsworth. In many ways,  it was the path of least resistance for our content teams and this included MS Teams, Google Meets or Hangouts, Zoom, Skype and Cisco VC tools. A lot of these have options for superior audio algorithms over GSM and 4G.
We also utilised Digigram’s 16 stereo channel IQOYA *SERV/LINK system, with the HTML5 Digigram routing system.  This solution meant that we could easily connect listeners and talent to studios using IP with Broadcast acceptable algorithms like OPUS.
We found the national Internet Service Providers (ISPs data) to be a lot more reliable than the mobile 3G/4G telco data (POTS/ISDN are not viable in the near future).
Audio quality is important but so is content and we’ll do whatever it takes to get great content to air.

SBS Manager of Radio Technology & Operations, Aaron Alphonso

Poor telephone quality / home studio contribution quality could be due to many reasons, including networks perhaps being overstretched and various other factors.
Here at SBS we developed a standard remote workflow procedure for social distancing that encourages the use of a commercially available professional remote audio contribution tool called ipDTL that can be accessed on a web browser without additional codec equipment. As a backup, we rely on Tieline ReportIT to connect to our existing Tieline codecs.  We also have a backup solution that relies on hardware codec devices.
These solutions are purpose built audio software codecs with low latency (delay) figures for remote audio contribution and voiceover work.
Because of this, they do not bother themselves with echo cancellation or mix minuses – they just focus on very high quality audio delivered with minimum delay, and trust that its being used by an operator that understands and has the facility to derive a true mix minus from the studio router or console.
We additionally equip our talent with specially selected headsets and audio interfaces pre-configured with the right EQ and dynamics settings built in to cater for a variety of home broadcasting scenarios.
Our disaster recovery plan includes using some of these more widely available video conferencing solutions, but we would only use it in an absolutely desperate bid to get someone on air, as echo cancellation is often not a strong point. Also audio bandwidth is poor, and it is very aggressively compressed, leading to noticeable audio problems when our listeners are hearing us on DAB+ or streaming, which as you know adds another encoding layer.
One thing that trips up these video conferencing systems and makes their echo cancellation algorithms work harder is talent not using headphones, or perhaps headphones that are not ‘closed-back’ which allow audio to bleed back into microphones.
I understand that not all networks may have the capacity or resources to use purpose built audio contribution solutions, and as such may rely on Zoom, Google Meet etc – my advice would be to please equip your talent with microphones that have excellent off-axis rejection.

The best microphone for remote working may not be the nicest looking large ‘studio like’ condenser microphone that our talent may desire, often these don’t work well in somebody’s kitchen or bedroom where there is a lot of unwanted noise reverberation from hard surfaces.  What we need are microphones that focus on picking up audio from the talent speaking directly into it, and not the entire untreated room. Also very important to ensure that they wear headphones or earphones that don’t bleed audio back into the mic.

Nine Radio Resources and Radio Technology Operations Manager Michael Sammut

I think there are a few things at play here.

Firstly, I’ve found that all the telcos are under enormous pressure during the coronavirus pandemic.  I’ve had several dealings with them while trying to arrange connectivity for outside broadcasts and setting up our talent to present from home and it’s very apparent that they are having a difficult time keeping service levels up while the demand profile for their services has changed so dramatically and so quickly.

So I think in some cases poor line quality has been a reflection of the network under strain.

Secondly, in the early days it was a scramble to get enough equipment together in such a short time to facilitate all our remote broadcasts. I remember in particular one of our regular contributors in the first week – she was dialling in from her landline at home! I heard it go to air and was mortified – the next day I bought and shipped some equipment to her to solve that problem.

We’re using a range of methods to enable remote broadcasting – the go-to is the Tieline ViA IP codec. We’ve also distributed a number of Rode NT USB microphones that we’re using with the Tieline Report-IT iPhone app and that’s been a great success.

With everybody broadcasting remotely and the immense strain this is placing on resources it’s not surprising that you’re hearing lower quality audio sneaking through a bit more. Frankly, it’s a testament to the professionalism and skill of the engineers, producers and presenters in the Australian radio industry that in almost all cases nobody listening can even tell that it’s very much not business as usual behind the scenes!

I don’t think that the audio quality that we’re putting to air is appreciably lower at this time. It’s certainly been a challenge and we’ve not always been perfect but our systems are working and in fact we’ve taken the opportunity to learn and introduce some great new innovations that have made our remote broadcasts even more dynamic and flexible in ways that are delivering real value for our listeners.


Ian Campbell and Hayden Beetar from AVC Group
Some of our clients are getting great results with the latest ‘CallerOne’ version of PhoneBox (now called Bionic Studio).

It started off as just a phone system,  but now our remote options are in high demand during lockdown, as well as the audio and video capture and social media functions. It incorporates Skype and soon other systems such as WhatsApp.

Combining CallerOne with another of the options, called ‘Anywhere’  has given some great results for phone calls in remote studios.

With so many broadcast teams working from home studios, CallerOne can connect the team members from any location. A call screener might be at his home, the presenter is somewhere else and the panel operator is in the studio. The system works on a browser and each person can perform their functions from anywhere, then the others can see what’s happening in real time. So the call screener will answer and park a call, then use chat to tell the presenter to take the call on line 2. It is possible for the presenter to put that caller to air from their home studio, or, if there is a panel op at the studio, they can take the call through the desk.

With the Anywhere function, producers can give callers or regular guests a browser link to call straight into the system. It bypasses firewalls and works with the G.722 standard for phone codecs to produce a result which is fantastic. Sometimes announcers tell us it sounds too good, the listeners don’t think it is a real caller!

The Anywhere function makes a browser to browser phone call, for better quality audio, this is important because phones are definitely competing for data now.

It uses WebICT for a point to point IP connection. The STUN (Session Traversal Utilities for NAT) Server allows it to connect through firewalls and the Opus codec gives good quality over open lines.

Phone systems are changing, major media companies are moving over to these systems. SCA, ARN MediaWorks and NZME are some of the big networks using it, either via SaaS or private PaaS.

STUN and TURN (Traversal Using Relays around NAT) Servers work hand in hand. STUN figures out where you are and TURN relays messages via voice.

Telcos are turning off the old services, so it is going to be a battle for data in the future and stations have  to be equipped for that. As well as competing with other telephone callers, phone users are also competing with the telcos themselves, for example, if you are and Optus customer using your phone in a Telstra cell area, do you think that Telstra won’t prioritise the quality of its own customers over you. That is why broadcast phone calls need a way to guarantee quality no matter what the settings or priorities of the mobile phone company are.

Media Technologist & President of Technorama, John Maizels

This is a very complex topic, because there are a squillion variables that come into play and mostly we have absolutely no control over them.

Digital services have been around for forty years or more, but pre-NBN some of the best phone quality came via good old POTS (Plain Old Telephone Service ) lines – real analogue phone lines, or the digital equivalent, designed to interconnect POTS lines and not affect the signal. 
We haven’t had true analogue signals between telephone exchanges for quite a while.  Those connections have been almost universally uncompressed 4kHz bandwidth, occupying a 64kbps channel for yonks.  However, the ‘last mile,’ the connection between the exchange and the office/studio/home has been good old copper wires.  In most cases, that gave a really good result.  When connected to a console via a decent digital echo-cancelling hybrid (good examples:  a Gentner DH20, the JK Innkeeper series, and the Telos Hx1) you get very good results to air. 

But we’ve progressed and most connections these days are fully digital, one way or another.  You’d think that would lead to a better result, wouldn’t you.  It’s digital, so what could possibly go wrong?


Voice Over IP services started appearing commonly in the 1990s, and many people had VoIP connections at home.  Unfortunately the desire to get more services into limited bandwidth, or to put lowest demand on dial-up or low bandwidth internet services, led to use of audio codecs that might have sounded OK in the lab but were pretty diabolical in practical use.  Many of the early codecs simply threw away bits rather than the more complex techniques we know today.  Distortion, muffled calls, and noisy conversations happened too often.

The challenge with those services from a user point of view is that it’s generally impossible to turn the knobs that affect quality.  You got whatever the phone company had set in the connection box (Analogue Terminal Adaptor, or ATA) and if those settings were suboptimal for your use, the call would sound like crap.  Often the call would sound good in one direction and awful in the other direction, or the call quality would depend on who called whom.  Some users have been able to log into the box and adjust the arcane parameters, but most phone companies don’t let you do that because you might actually make things worse.

For most users, the audio quality of phone services provided over NBN is still entirely dependent on the ISP and the box they give you.  It’s digital to your premises, with an analogue phone port on the back of the router.  Is that bad?  My personal experience has been very good – an analogue phone to analogue phone connection via NBN can be truly excellent.   This is the service provided to many community stations, and if the panel connection is via a good hybrid, the on-air quality could be excellent.  The best thing about the NBN service, is that the quality is unlikely to change no matter where the other end of the call is… from next door to across the country, the call will be as good as the phone at the other end.

Mobile phones add an entirely different level of complexity.  The codecs used on GSM are designed to minimise bitrate.   Depending on the codec – and again, you have no control over that choice – the quality might be good or bad.   At the worst end of the scale, different brands of phones might use different codecs, leading to a situation where the audio has been coded multiple times… and that NEVER helps.  And then there’s the signal path back to the base… we all know how a call sounds when the caller’s signal is marginal. Anyone who has ‘WiFi Calling’ fully enabled on their phone, and is connected to a decent WiFi service, is substantially bypassing the tower network and sending audio directly back to the telco’s entry point over IP.  Not only does that eliminate the bad-signal effect, it might well lead to automatic selection of better end-to-end coding.

Once you get to real data connections with client-driven audio, the game can change a heap in two ways:

    – between two digital clients (eg:  Zoom to Zoom), the audio connection is full duplex, in that the input and output signals never have to be mixed.  This makes a heap of difference at the console connection end, and the software can take the place of a hybrid.  The echo-cancelling ability of the best software can be just brilliant.  Zoom, GoTo, Skype and Jitsi all work wonderfully, and the interface might take no more than a line in/out connection between a PC and the console telco channel.

    – many services offer direct dial in connections from a phone line, with numbers available for many countries.   So instead of calling your studio’s local number, the caller can dial into the conference client.  There’s no way to tell if the call will be stable or interrupted – you take your chances, and you’re subject to many factors that you can’t control.   Conferencing systems can combine many callers on-air at once, and the software sorts out all the problems of cross connections and mix-minus, which may be an advantage for interviews with multiple remote contributors.

As always, the quality of engineering into and out of the console and the PC audio interface is paramount.   No laws of physics have changed since radio began, and it’s absolutely vital to ensure you manage gain structure, noise, distortion, impedances, and cabling to get the right result.  But nothing that is too onerous, there are plenty of USB audio interfaces that will do the job with balanced lines and plenty of headroom.

So there you have it. There are many aspects to the issue of good quality phone content, and I hope we have uncovered most of them, thanks to the experts.

We welcome more experiences and opinions on this issue in the comments box below, or to [email protected].

Main photo: Wilko and Courts from Hit 104.7 Canberra using NextGen and PhoneBox in their home studios.



About the Author

Steve is the founding editor of this website.

He is a former broadcaster, programmer, senior executive and trainer who now runs his own company Ahern Media & Training Pty Ltd.

He is a regular writer and speaker about trends in media.

More info here.




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