New Gear: Grandstream’s HT812 Analog Terminal Adapter

In the earliest days of consumer VoIP services the venerable Cisco ATA-186 was the way to connect a traditional analog telephone to one of those new-fangled Vonage accounts and save some loot. It didn’t take too long before our strategy evolved from an analog terminal adapter (ATA) plus a an analog phone or a cordless phone, to SIP desk phones and SIP/DECT cordless phones.

As such, it’s been years since an ATA held any interest at all…until last week. Last week I received a couple of notices about a new pair of ATA’s from Grandstream, the HT802 and HT812.

Grandstream HT812

The first thing I saw was a promotional email from VoIP Supply for the HT812. It described the HT812 as a two-port FXS with a built-in router and Gigabit Ethernet.

Combining the edge router with the ATA is a convenient way to ensure that there’s an adequate quality-of-service mechanism for the voice traffic. It reduces the support burden for an ITSP.

I’m not certain the real value of Gigabit Ethernet on an ATA. Perhaps some people have real Gigabit ISPs. In any case, more is better, right?

VoIP Supply HT812 Promot Snippet

Setting aside the basic stuff, what caught and held my attention were two specific claims about the new gadget;

  • Support for wideband audio (HDVoice.)
  • Support for the Opus audio codec.

While the VoIP Supply promo conflated these facts, they are in fact separate items.

A further promotional email, from Grandstream themselves, highlighted, “Exceptional voice quality with wide-band HD codec.”

Grandstream HT812 Promo snippet

I see this very same language being parroted elsewhere.

This is interesting. To my knowledge there has never been a wideband-capable ATA.

Opus is a Great Codec

Opus can be used in narrowband, where it achieves very good voice quality at very low bit rates. It’s narrowband modes were, in large measure, derived from aspects of Skype’s SILK codec.

It also has wonderful forward error correction (FEC) capability, making it especially useful on lossy or congested networks.

Of course, Opus is widely known for its splendid wideband (or better) capabilities, which almost every WebRTC-based service trumpets. It’s the intersection of HDVoice and an ATA that has me confused.

Is the FXS a Bottleneck?

An ATA by definition presents one or more FXS ports for connection to an analog telephone. FXS ports and analog telephones are not, generally speaking, HDVoice capable. There are, I believe, long-standing technical standards that stipulate analog devices attached to the network must limit the audio bandwidth presented to the legacy PSTN standards. That means 300 Hz – 3.4 KHz. Definitely not HDVoice.

When the telephone networks transitioned from analog to digital it was realized that it was truly necessary to protect the network from out-of-band analog signals making their way into an A/D convertor. An unfiltered analog signal passed into a A/D convertor creates aliasing. This aliasing noise occurs in the audible range, and can badly degrade the sound quality.

The natural consequence of this fact is that all electronic telephones that have analog line interfaces are designed to bandwidth limit the signal path to PSTN standards. Just take a quick look at the spec sheet for the Texas Instruments TCM29C13A, a chip that is a “COMBINED SINGLE-CHIP PCM CODEC AND FILTER.” Therein you will find frequency plots that show filtering with strict adherence to the best legacy PSTN audio standards.


Many thanks to Dave Knell, founder of TelNG and a member of the VoiceOps Mailing List, who provided the tip about that TI document, as well as a link to a related ITU G.712 specification document.

Prior Efforts

I am aware of a past project that involved  passing HDVoice via an FXS connection. In that case, cable STB-maker Arris was working with Gigaset. Since the US CableCo’s are major telcos in their own right, there was the idea that a single, hybrid STB could provide cable TV, broadband and voice service.

The phone service component of the bundle could be delivered via a DECT/Catiq base radio included in the STB, but that would require the customer to purchase a DECT/Catiq cordless handset. A cheaper alternative was to add an FXS jack to the STB. That way there was nothing more to buy. The end-user could connect a traditional telephone.

The further rumor was that Gigaset, who could easily provide the DECT/Catiq solution, could also provide an analog telephone that supported HDVoice via an analog line connection. It would essentially be an analog telephone specially built to not be connected directly to the PSTN.


Way back in the past telephones were not electronic, but electro-mechanical. They were electrically very simple devices. Lacking for cheap electronics, there were no fancy filters.

Some think that, at least in theory, such antique phones might be able to deliver and audio experience beyond legacy PSTN standards. Perhaps not HDVoice, as we enjoy with better IP phones, but a better analog telephony experience.

I, for one, am a bit skeptical about this idea. When I said so it somewhat animated a discussion thread over at the DSL Reports VoIP Tech Chat forum.

While skeptical, I remain curious. So to satisfy my curiosity I have ordered an HT812. I intend to find out exactly what it can, or cannot do.

Now I wonder where I can find an old phone to use in some experiments?


    While I understand the criticism expressed here about wideband codecs, I purposely was looking for an ATA that supports narrow-band opus. In fact, that’s how I got to the blog, I searched for ATA with opus support. Waiting for it to arrive to see how it works… Hope to use it over dodgy data connections to call home, and my wife insists on an easy to use reliable cordless phone, hence the ATA requirement.

    • mjgraves

      In the realm of audio codecs Opus remains Johnny-code-lately. You’d probably have better results using G.729, which remains the most widely deployed low-bitrate codec. Literally every ATA in existence supports G.729.

      While I have an HT812 in-house, I have yet to put it into use. A quick initial experiment with interop to other end- points proved fruitless. It would not connect to my Polycom VVX or Bria soft phone, both of which “support Opus.” My next step is to Wireshark the SIP traffic and see what’s happening.

      If you get one perhaps we can try to to connect. Interop between like devices should be the easiest.