Mr Prokop’s SIP Adventures blog has proven interesting, so I thought the podcast worth a listen. Sadly, while the host is presented full bandwidth, as might be expected from a local recording, the guest is presented in narrowband. Given that the subject matter is WebRTC I think that this is more than a little anachronistic. WebRTC-based services are in fact a very easy way to enjoy wideband audio for the purposes of producing a podcast.
Dragging the podcast in my trusty editor I find it to the a definitive example of full-band audio vs narrowband. The file is sampled at 44.1 kHz, so the top of the vertical axis is 22 kHz. The guests audio is a good quality PSTN call, but even that is quite a contrast from the host. This contrast is very jarring to the listener.
As to the content of Mr Prokop’s commentary on WebRTC and SIP, I’m not sure that I agree with all that he offered. He seems to harbor some serious doubts about the future of WebRTC, a technology that is only 2-3 years from its inception. I wonder how far SIP had progressed just that long after its initial creation?
There are people making use of WebRTC right now. It might not have swept the traditional telecom world yet, but it’s impact is being felt in many quarters. Were that not the case there’d be little reason to be discussing it in such a podcast.
P.S. – I grow weary of pointing out how the telecom press are not typically “dog fooding” their industries’ own better technologies in the production of podcasts. If anyone would like some guidance on how to grow beyond narrowband telephony for the specific purposes of podcasting, please get in touch. As co-producer of the VoIP Users Conference for the past few years I certainly have some experience in how it’s done. It’s getting easier all the time.