HD VoIP? Oh, Brother…
VoIPGirl is asking a question about whether or not people should consider HD VoIP when they are choosing a VoIP service. I’ve never heard anyone advertise that, so it’s not like a consumer would even know how to look for it. Meanwhile, I will attempt to answer her question.
Without getting too technical here, when a VoIP call is initiated, it is possible to utilize one of a number of codecs. The ‘Co’ means compression, the ‘dec’ means decompression. More succinctly, it is the method by which your analog voice is compressed and sent across the wire as ones and zeros. The codec used will often depend on the available bandwidth and endpoint capabilities.
Let’s use Skype as an example. When you make a Skype-to-Skype call, the call quality is wonderful. Why? They use something called ISAC (warning: PDF link), which is a proprietary codec by Global IP Sound. It is a “wideband” codec that sounds pretty damn good. When you make a SkypeOut (PSTN) call, however, the codec used is G729, which is a narrowband codec that doesn’t sound quite as good.
Presumably, if you are using a service that supports “HD” VoIP, then the “HD” codec, whatever it is, will only come into play on-net, meaning within the relevant VoIP provider’s network. Once it goes off-net (e.g. to the PSTN), a different codec will be used (G711 or G729). This is all done on the fly and completely transparently to the end user, except for the obvious sound quality difference.
Should a consumer care about HD VoIP? Unless you call people within the same provider a lot, it’s not going to matter. It won’t hurt anything, but it won’t help either.
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[...] asks what HD VoIP is, and PhoneBoy responds that it probably doesn’t mean much. Wideband CODECS have been around for ages, but because [...]
Comment by John Federico
Consumers may not care, but podcasters care a LOT since many of them record their podcasts using VOIP services like Skype and Gizmo Project.
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-jf.
–
John Federico
http://www.newrules.com
http://www.linkedin.com/in/johnfederico
http://johnfederico.brandbrains.net
http://www.odmcast.com
Comment by PhoneBoy
Podcasters mostly use Skype for this very reason, John. Non-podcasting consumers are, on the whole, not likely to care. Of course, there are some that are extra sensitive to the use of codecs (some hate G729, for instance).
Comment by Mark R. Lindsey
PolyCom is starting to advertise better-quality codecs built into their hard phones. They sound nice, talking to one another; but I wonder if the tiny marginal difference in a 16-bit codec is going to matter to anyone.
Of course, PolyCom’s are big sellers among VoIP service providers lately. And carrier-to-carrier VoIP peering is moving along, slowly. Maybe some day the nicer codecs will matter.
Comment by Vic
I had no clue HD VoIP was even around. Apprently it’s so limited that I’m not suprised I havent.
http://nationwideLD.com
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Network Performance Links: HD VoIP, iPanic, and one man’s four-core trash is another man’s three-core treasure….
The Phoneboy Blog: HD VoIP? Oh, Brother… Phoneboy talks about HD VoIP, or VoIP that sounds like it’s coming from a CD Player, not a phone. That’s entirely plausible, but, as he points out, it’s a major increased bandwidth cost……
Comment by Stephen Brown
I’m sure part of the talk of HD VoIP comes from the recent articles that appeared in eWeek. http://www.eweek.com/article2/0,1759,2181656,00.asp
It would seem like until we get the performance and bandwidth issues worked out with good “old-fashioned” VoIP that moving to HD-VoIP doesn’t make sense. The thought that now I’ll be able to hear tele-marketers crystal clear in Hi-Def really doesn’t do a lot for me.
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Comment by john
HD VOIP?? LOL, that is a joke! Do you really think anyone would purchase this? Unless you have money to spend down the drain.. Come on people, what are high speed internet providers thinking? http://t1survey.com
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Comment by Peter
If you want to trial a “HD VoIP” environment (16K, 22K or higher audio for the rest of us) then do give FreeSWITCH a try. It’s an Open Source telephony platform and should release this week their first Release Candidate anticipating their upcoming 1.0 release. Get a server, install Red Hat Enterpise Linux (or it’s free alternative called CentOS) and FreeSWITCH and hook up a couple of Polycom IP430 or IP560 phones. It even does end-to-end encrypted calls too (TLS/SRTP). That’s even safer than Skype with that German company selling Skype interception equipment to their government (like the NSA does not have that…). Enjoy the real stuff: Open Source & Open Standard HD VoIP. Not that PC bound proprietary Skype crap who’s feature is questionable to say the least. The project lives at http://www.freeswitch.org Support is available in the irc channel #freeswitch on irc.freenode.net or the mailinglist (subscribe at freeswitch.org).
Comment by A.T.
no, high-quality VoIP is not a joke - it is quite feasible, and often available right there… but there are few factors against of it. Firstly, your operator (last mile, be it DSL or cable or even IP-over-avian cheapest carrier) is _not_ interested to give you high quality, it is interested to push as more herrings into… ergh, I mean cheap VoIP calls into available bandwidth. And current battle “Bittorrent-humans vs Comcast-and-all-big-telco-cable-brotherhood” is clearest example - telcos are in business of “yes, we sold you 8mb/sec downstream but you may not use it for full”. Secondly, the other side of coin is purely psychological - once you get used to HD, even slightest distortions will bother you like sharp knife: is this what you are looking for?
Comment by PhoneBoy
@peter even if the transport is encrypted, a-la Skype, the endpoints themselves are susceptible to being hacked. That’s how most of the Skype “interception” equipment works for the most part.
@A.T. I have a hard time using other VoIP tools because of the quality of Skype.
Comment by A.T.
@Phoneboy I do not give up that easy to all cloaked nicety of Skype
i prefer to know what i run on my machine and where it sends what 
and i do not see any big difference on “Skype-to-Skype with iSAC on 16kHz” and “SIP-to-SIP on Speex with 16kHz”… should I try differentiate them harder? lol
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