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.

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Opus Codec Appears in Broadcast STL System

Intraplex-IP-Link-100-and-200-angle-transparentToday’s news dump included an article on GatesAir, now a freestanding entity, it was once the transmission division of what was then known as Harris Broadcast. The company makes radio and TV transmitters, as well as related equipment, which includes studio-to-transmitter (aka STL) links. According to the Broadcast Beat article they have sold and installed one of their Intraplex IP STLs to WRLY-LP, a low power radio station in Raleigh, NC.

A broadcaster with a transmitter that is not located right at their main building (not co-sited) needs an extremely reliable means of sending their broadcast signal from the studio to the transmitter location. They also need some way to get some transmitter telemetry back from the remote location so that they can monitor the health of the transmitter. Their on-air presence via the transmitter is, after all, their bread and butter.

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Obi Hai Ships OBi1032 IP Phone

OBi1032-SIP-PhoneObi Hai has been around a long while. Their niche has been ATA-like devices that were sufficiently sophisticated to provide hardware access to Google Voice. As was discussed when they appeared on VUC, the founders of the company were in involved in the earliest days of VoIP. More specifically, they were behind the development of Cisco ATA 186, the very first ATA.

In years past I’ve watched as others have expressed their enthusiasm for the OBi Hai ATAs, especially those who were trying to leverage Google Voice. GV has never been a significant factor in my working life.

Further, I’ve long held that ATA’s  fail to offer many of the advantages of a bone fide IP phone. I do admit that OBi Hai was quite aggressive about extending the capabilities of the humble ATA beyond the demands of the typical ITSP. Nonetheless, ATAs have held little interest for me in recent years.

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VUC500: A Special Event Featuring Dr. Henning Schulzrinne

Friday, August 1st will see Dr. Henning Schulzrinne appearing as our featured guest for the 500th(!) edition of the VoIP Users Conference. While presently the CTO of the FCC, Dr. Schulzrinne has along history as a pioneer in the field of real-time communications over IP networks. He is responsible for the creation of such protocols as RTP and RTSP, and co-creator of SIP. He is a co-chair of the Internet Technical Committee of the IEEE Communications Society. His name appears on numerous RFCs.

In 2013 Dr. Schulzrinne was inducted into the Internet Hall Of Fame. The following clip is his acceptance speech from that ceremony. I offer it in the hope that it will help inspire some good questions.

The technical part of this weeks exercise will involve the use of Jitsi Video Bridge (http://jitsi.org) to host the call instead of our usual G+ Hangout. Thus we’ll be taking a pure, open source, WebRTC approach to things. Given a limited number of seats on the Jitsi video bridge they are by invitation only.

To allow more people to watch the call the Jitsi Video Bridge feed will be live streamed to our YouTube channel*. http://youtu.be/-pfXBE2POxo

Jitsi Video Bridge has a newly implemented ability to dial out to a SIP URI. This is how we’ll be interconnecting with ZipDX. Since both JVB and ZipDX support  the Opus codec that will be a good quality connection.

Interactive, if audio-only, participation in the call will be possible by connecting to ZipDX via SIP URI (200901@conf.zipdx.com)

The audio will also be streamed live via Mixlr. http://mixlr.com/voipusers/

A big thank you goes out to Emil Ivov and the Jitsi team for setting this up!

*The mechanics of exactly how we’re feeding YouTube from JVB I will likely describe in a future post.

LifeSize On WebRTC

LifeSize_Icon_SeriesLast week LifeSize had a webinar on the topic of WebRTC. I took an hour to listen to what they had to say and pose a couple of questions. Their target audience appeared to be people who might have heard some of the hype about WebRTC, but were not otherwise familiar with this new phenomenon. Suffice it to say that the material covered was introductory.

The webinar started with a pre-recorded video of Casey King, LifeSize CTO and Simon Dudley, who is described as LifeSize video evangelist. Their pre-recorded conversation was followed by an audio-only live segment where they answered questions arising from the audience, which was reported to be over 1000 people.

If you care to view the event after the fact you’ll find a recording of the webinar here.

During the live event I posed a couple of questions in the text chat. I asked if they had any plans to support the Opus audio codec and VP8 video codec. These are core aspects of WebRTC, although the debate about whether VP8 or H.264 should be “mandatory” rages on.

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IETF 87 Berlin: WebRTC Opus Experiment

Next week the IETF will be holding a conference in Berlin. Part of that conference is a Technical Plenary Session about the Opus audio codec  scheduled for Monday, July 29th 5:40-7:40pm CET.

The IETF usually streams much of their conferences so people who are not free to travel may still participate. Quite often there’s an audio stream, sometimes there’s video and a web share of any slides. They usually stream from several of the meeting rooms, as multiple sessions are typical going on in parallel.

This coming Monday’s Technical Plenary is also going to be the basis of an experiment. The session is going to be streamed via WebRTC. That means that anyone with a WebRTC capable browser will be able to monitor the session. It further implies that the session on Opus will in fact be streamed using Opus…which seems only fitting.

The session will be recorded for those who cannot participate live. Since 5:40pm is Berlin is 10:40am in Houston I’m hopeful that I may be able to list in using Chrome.