That’s nice. Alan certainly knows his stuff. He’s been a VUCfrequent guest in recent years.
It’s a pity that the podcast was produced via a plain vanilla PSTN telephone call. Narrowband in the best tradition of Ma Bell, circa 1945.
The failure to tap a new age, HDVoice-capable means of podcast production just feels wrong. Most especially given the widespread emphasis on WebRTC as a key aspect of the new age of telecom creativity.
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.
Earlier this week Matt Brunk penned a post over at CMP’s No Jitter blog entitled “SIP Means Change.” It’s a short piece detailing the contrast between SIP phones, Asterisk servers and legacy proprietary equipment. In particular it dwells on the boot times of the various items. He highlights how anything that takes longer, even just a little longer, ultimately has a higher cost.
Matt points out that the older, proprietary digital phones were effectively instant-on devices compared to SIP phones. This is a little obtuse in that SIP isn’t really the culprit. It’s just a protocol. Cisco phones running SCCP would have similar boot times to SIP handsets. I presume that Nortel phones running UNISTIM would also have similar boot times.