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Technology & The Art Of The Podcast

Incidentally, ZipDX is not the only company offering wideband conferencing, just my personal favorite. Other options include:

Gigaset-S79HAlso in his first point Alec recommends:

Avoid cordless handsets. Cordless handsets often have a noticeable background hum.

I don’t believe in condemning an entire class of devices in this manner. In fact, I’ve had great success using the Gigaset SIP/DECT systems. Those being both cordless and VoIP-based they probably shouldn’t work at all…at least based upon the recommendations as given.

There are about a zillion different kinds of phones, including cordless systems on the market. You get exactly what you pay for so don’t buy the cheapest thing that you can find.

If you really want to go cheap then use a wideband-capable freeware soft phone like PhonerLite, but buy a good headset. I like the Plantronics .Audio 615m USB headset which sells for under $40.

What About Skype?

I can see the attraction of using Skype. The Skype client software is free and nearly ubiquitous. However, I never use Skype to join a conference call.

If you’re making a Skype call to a normal phone number then Skype is going to use G.729a to pass the call to someone for PSTN termination. G.729a is the industry standard low-bitrate codec.

G.729a is generally regarded capable of offering good audio quality, with a best case Mean Opinion Score of 3.9. This should be compared to a land-line call using G.711, which has a best case MOS score of 4.3. A Skype call to a normal conference bridge won’t be as good as a land-line call, even under best case conditions.

Simply put, none of the traditional codecs of the legacy public telephone network (G.711, G.729a, G.723, G.726, AMR) can hold a candle to a wideband codec! And there are so many wideband codecs to choose from, including: G.722, G.722.1, iSac, AMR-WB, SPEEX, CELT & SILK!

As a practical matter the only wideband codec of concern today is the great-grand-daddy of them all, G.722. It’s the one codec that offers great audio quality and is implemented in a wide range of software and hardware.

You’d think that Skype would be a logical choice for podcasters since it implements SILK, the very latest wideband codec. In practice it really depends upon exactly what you need to do.

If you’re using Skype at both ends of a one-to-one interview then it can provide excellent audio quality, much better than a normal phone call. Amongst the things that I have listened to Dan York’s BlueBox Podcast on VoIP Security was often done using Skype in this manner.

Recent releases of Skype allow multi-way as well as one-to-one calling. If all participants are using Skype and a good headset then you should be able to enjoy a superior quality call.

However, if you’re using Skype to call a traditional conference bridge via a normal phone number you will not realize any improvement in audio quality.

Remember, the goal is to achieve the best audio quality possible…not just sound like a good phone call.

3. If possible, use a conference calling service that allows you to record the call from the conference bridge, rather than from one of the handsets. By recording the call from the bridge, you minimize the drop-off in volume that occurs as phone calls traverse multiple networks. In addition, if you record from the bridge, no additional equipment is required to make the recording.

It’s true that using the conference service to record the call is very convenient, and the quality is often very good. However, there can be advantages to making a local recording as well.

The conference bridge typically is going to make a recording that is compressed into MP3 format at some nominal bit-rate. That’s ok as you likely want to distribute in that format. But if you wish to do some audio post-production, as Alec suggests,  you might be better off starting with a higher-quality, uncompressed recording.

In my case, I often want to go into the recording and edit sections for content or perhaps replace my intro or extro if I think that I could have done better. I achieve a better quality final recording by starting out with an uncompressed WAV recording made locally.

I take one of three approaches to this:

1. The Polycom SoundPoint IP650 & IP670 desktop SIP phones can record a call in uncompressed WAV format to a USB stick when equipped with the optional Polycom Productivity Suite. That option costs under $10 per phone. If the call is a wideband call then you are assured that the captured call quality is as good as it can possibly be.

This is my preferred approach since the Polycom phone is essentially an appliance. It just works. It leaves my PC free to do anything else that I might need during the call.

2. Many soft phones, including my favorite Eyebeam, can record a call to an uncompressed WAV file. I tend to use this less since it means that I’m relying upon my PC which I may need to use for other tasks during the call.


3. V-Emotion software can be used in conjunction with a soft phone to record the call audio. While I had to buy this software it has the advantage of allowing you to keep your voice split from the rest of the call. It saves a stereo track with your voice on one channel and the other participants on the second channel. This can be invaluable for editing sections where, in a heated debate, the host and the guest are “stepping on” each others audio.

V-Emotion can also be used to playback pre-recorded audio files into a call. This is handy for injecting a musical opening & close into a live call, or adding game-show-like sound effects.

In reality, I have a belt-and-suspenders approach. That is, I record locally but I also keep the MP3 recording from the conference bridge to use as a backup. I can work from it if something happens to make my local recording unusable.

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