HD VoIP solutions for a new home
SeanTek:
I am looking at rewiring a new home so that it is only-IP (furthermore: IPv6 preferred!). That means no need for "legacy" communications technologies such as coaxial cable and RJ11 phone lines. If possible, I want everything to run over RJ45/Cat6 Ethernet (with PoE), and 802.11n/802.11ac Wi-Fi. (Television/video services can be distributed over IP, such as with HDHomeRun boxes.)
As a technical user but a relative newbie to the VoIP space, I am looking for solutions that use standards-based technologies where possible, are not terribly expensive or hacked together, offer some kind of encryption, and support high call quality across devices. Closed-loop systems like Ooma are attractive for ease-of-use and claim to offer "encryption", but it is unclear what Ooma means by "HD" in their marketing, or how they get "HD" to work outside of the Ooma ecosystem. HD voice can be used by other closed-loop systems such as Skype or Apple FaceTime: they control the software at both ends! I want a path to interoperability.
What VoIP solutions do people here recommend? I like the premise of using Obi1062 phones with Google Voice and other SIP providers (recommendations?), but it's unclear just how much better call quality will be. I would also like recommendations for ensuring HD voice (Opus codec or G.722) across different devices.
SteveInWA:
Hi:
Wideband audio over VoIP ("HD Voice") is still emerging technology -- there is very little interoperability between carriers at this time. Over time, as carriers make more progress, you'll eventually be able to call between, say, a Verizon or Sprint HD Voice-enabled handset and a SIP VoIP carrier, but it's not here yet.
Any of the OBi 10x2 IP phones would be fine for your application. The all support HD Voice (OPUS and G.722 WB CODECs). Calls placed between OBi IP phones over Obihai's own OBiTALK network use the OPUS WB CODEC, and they sound great -- the phones were specifically engineered with high-quality audio circuitry to support wideband audio. I also tested calls between the two IP phones using two Callcentric SIP numbers (in other words, staying on the VoIP infrastructure, and not going over the PSTN), and they also used OPUS. OPUS support will depend on your choice of service provider. Calls that traverse the PSTN cannot support wideband audio, since the PSTN itself is limited to narrowband.
It's a similar story for encryption. The OBi IP phones support it, but the service provider and other end would also have to support it.
OBi IP phones also support standards-based PoE, so use a PoE-equipped Ethernet switch, and you'll only need one cable to connect each phone!
SeanTek:
Thanks Steve for your informative response!
I guess that when two IP phones are using PSTN numbers on the same provider (both Callcentric, both Ooma, etc.), the provider does not have to transcode the audio traffic to a different format, so it can stay Opus throughout, is that right?
Also if one IP phone calls another IP phone using a SIP URI, no transcoding should be needed?
So the issue is traversing the PSTN, which seems to imply some decoding and re-encoding is necessary. I read up a bit (Googled it) on how the PSTN works, but can you or someone explain at what point in the PSTN narrowband is enforced, and how?
The whole PSTN thing seems very hand-wavy. If the PSTN is a vast network of junction points and communications links, and all those elements are digital, then I do not see why transcoding has to occur (and if so, transcoding to what encoding formats, exactly, if it's all digital). Even if the nodes do not support a particular codec, the nodes should just pass the digital data through.
If a link in the PSTN ciruit is an analog copper wire pair, then clearly some decoding on one end and re-encoding on the other end has to occur.
SteveInWA:
Hi:
"Transcoding" is not the issue. Transcoding is when an already-digitized signal, which was digitized using one standard format, is decoded and re-encoded into another digitized signal. For example, CD-Audio transcoded into MP3 files.
Short history lesson: The telephone network, from the time of Alexander Graham Bell, was purely analog, consisting of a limited range of audio frequencies being carried over copper wire "loops". For a local call, the loop was between your premises and the central office. Between central offices were "long lines" (simply more pairs of wires). At first, telephone operators (picture Lily Tomlin) plugged cords into jacks to connect the calls. Later, electromechanical stepping relays switched the calls for direct-dialed calls. Thus, the name Public Switched Telephone Network, PSTN. Each call was "circuit-switched": an end-to-end connection patched together.
Over time, this meant more and more bundles of wires, until the system couldn't handle it. Bell Labs eventually developed digital telephony. Each conversation was run through an analog-to-digital converter, changing it into a pulse-code-modulated stream of bits (similar to the bitstream on those Compact Discs). Those bits were multiplexed (combined) with other conversations' bits and carried over the wires, and later, microwave towers, satellites and fiber optic cables. To conserve expensive and limited bandwidth, and, importantly, to deal with the physics of signal loss and distortion, using the technology available at that time, scientists looked at the average frequency range of human speech, and decided how much of the bottom and top-end they could throw away, leaving an intelligible mid-range. This became the standard used to this day over the PSTN, the PCM G.711 CODEC, which is a 64Kbps digitized bitstream, encoding frequencies at 8,000 samples per second, between 300Hz and 3400Hz
All the equipment in the PSTN is designed only to handle 64Kbps narrowband -- every electronic component, and other piece of hardware and software is only engineered to carry that level of frequency range. It simply isn't compatible with wideband audio. Think of it as the least-common-denominator, similar to AM radio -- AM radio has a very narrow bandwidth, by design, and FM radio has a wider, but still-not HD bandwidth, until digital HD radio came along.
VoIP doesn't have that restriction, since it uses packet-switched (not circuit-switched) RTP protocol over UDP (one of the simpler protocols in TCP/IP), which doesn't care what bits are being sent (it's just buckets 'o' bits, as far as UDP is concerned). So, any CODEC supported by the equipment at both ends can be used to transfer packetized data. It doesn't care if it's Morse Code, or a HD YouTube video, or a HD Voice phone call, as long as you have adequate bandwidth on your internet connections so the connection doesn't become saturated (like you see when trying to watch HD Netflix video when everyone else is doing the same thing at once).
Thus: IP phones, or any other VoIP hardware or software, can communicate using whichever CODEC they both support, as long as the call stays purely as TCP/IP data packets. This means, using SIP, for example, one endpoint sends a "HELLO" to the other's URI, and then the endpoints negotiate and set up the call. Once this is done, the bitstream just flies over the network.
Commercial SIP VoIP service providers (Such as Callcentric) support calls either being sent purely as SIP, URI to URI, or via the PSTN. Only the former method supports HD CODECs. In the latter case, calls are limited by the bandwidth specification of the PSTN. Services like OOMA, Skype and Google Hangouts similarly can either use direct TCP/IP or can step down to the PSTN as needed.
As a final note: the large telcos that have to support the FCC-regulated PSTN aren't really making much if any money on it anymore. Verizon and AT&T have dumped a lot of their landline business, and most of the legacy "Baby Bell" companies have merged to survive. CenturyLink and Frontier are two good examples of companies that are scraping along, with piles of landlines, much of which they inherited from bankruptcies and dumping by other telcos. The industry would like to get rid of the PSTN entirely, because it is so much more expensive to maintain than VoIP, but due to the hundreds of millions of people who depend on it, especially in rural areas, it's not going away anytime soon.
SeanTek:
All makes a lot more sense now: namely, that even in the digital parts of the PSTN (which is to say, pretty much all of it these days), all the equipment is engineered to handle the PCM G.711 codec. Is it that the G.711 codec or its technical features (64Kbps, 300Hz-3400Hz, etc.) are essentially hardwired into the protocols, or rather, that there are codec selection mechanisms (e.g., in SS7) but G.711 is the only common codec that all parties can be expected to support?
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