ITExpo: The Realities of Mobile Videoconferencing

I will be moderating a panel on this topic at ITExpo East 2012 in Miami at 1:00pm on Thursday, February 2nd.

The panelists will be Girish Khavasi of Dialogic, Trent Johnsen of Hookflash, Anatoli Levine of RADVISION and Al Balasco RadiSys. This is a heavy hitting collection of panelists. Come with your toughest questions – you will get useful, authoritative answers.

The pitch for the panel is:

As 4G mobile networks continue to be rolled out and new devices are adopted by end users, mobile video conferencing is becoming an increasingly important component in today’s Unified Communications ecosystem. The ability to deliver enterprise-grade video conferencing including high definition voice, video and data-sharing will be critical for those playing in this space. Mobile video solutions require vendors to consider a number of issues including interoperability with new and traditional communications platforms as well as mobile operating systems, user interfaces that maximize the experience, and the ability to interoperate with carrier networks. This session will explore the business-class mobile video platforms available in the market today as well as highlight some end-user experiences with these technologies.

ITExpo West — Building Better HD Video Conferencing & Collaboration Systems

I will be moderating a session at ITExpo West on Tuesday 5th October at 9:30 am: “Building Better HD Video Conferencing & Collaboration Systems,” will be held in room 306A.

Here’s the session description:

Visual communications are becoming more and more commonplace. As networks improve to support video more effectively, the moment is right for broad market adoption of video conferencing and collaboration systems.

Delivering high quality video streams requires expertise in both networks and audio/video codec technology. Often, however, audio quality gets ignored, despite it being more important to efficient communication than the video component. Intelligibility is the key metric here, where wideband audio and voice quality enhancement algorithms can greatly improve the quality of experience.

This session will cover both audio and video aspects of today’s conferencing systems, and the various criteria that are used to evaluate them, including round-trip delay, lip-sync, smooth motion, bit-rate required, visual artifacts and network traversal – and of course pure audio quality. The emphasis will be on sharing best practices for building and deploying high-definition conferencing systems.

The panelists are:

  • James Awad, Marketing Product Manager, Octasic
  • Amir Zmora, VP Products and Marketing, RADVISION
  • Andy Singleton, Product Manager, MASERGY

These panelists cover the complete technology stack from chips (Octasic), to equipment (Radvison) to network services (Masergy), so please bring your questions about any technical aspect of video conferencing systems.

HD Voice – state of deployment

At the HD Voice Summit in Las Vegas last week, Alan Percy of AudioCodes gave a presentation of the state of deployment of HD Voice, citing three levels of deployment: announced interest, trials and service deployment.

Percy’s take was that in the “Crossing the Chasm” technology adoption lifecycle, HD Voice is right at the chasm.

Here is his list, augmented with input from Jan Linden of GIPS,Tom Lemaire of FT/Orange, Doug Mohney of HD Voice News and Dave Erickson of Wyde Voice:

Category Company Stage
PC VoIP Skype >500 m downloads
QQ (China) >500 m downloads
Gizmo5 (now Google)
Wireline telco France Telecom 500K HD users
British Telecom Trials
FT/Orange Spain Deployed 1Q09
FT/Orange Poland Deploys 1Q10
Mobile Orange (Moldova) Production
Orange (UK) Deploys 3Q10
Orange (Belgium) Deploys 2010
CLEC VoIP Alteva Production
SimpleSignal Production
Ooma 25K HD users
8×8 >70K HD users
OnSIP Production Trials
US MSOs CableVision/Lightpath Limited Trials
Conferencing ZipDX Production
ClearOne Production
Citrix Production Production
Global Crossing Limited Trials

The main codecs in each of these deployments are: Skype:SILK; QQ, Citrix, Freeconferencecall:iSAC; mobile:AMR-WB; all others: G.722.

Alan pointed out the conspicuous lack of involvement of the cable companies (MSOs), even though Cable Labs has done a good job of creating HD specifications for them.

Wideband audio conferencing bridge

Skype lets you do audio conferencing with wideband codecs, and a service called Vapps High Definition Conferencing does the same thing for non-Skype VoIP calls.

Now other VoIP providers can offer wideband conferencing too. A company called Wyde Voice sells an all-IP conferencing platform that natively uses wideband codecs. The Wyde platform uses the iSAC codec from GIPS, so anybody calling in from a soft phone like the Gismo5 client, or the Google, AOL or Yahoo VoIP clients can enjoy a conference in wideband. If one of the participants in the call is using a narrow-band codec, the Wyde device up-samples the signal to wideband quality for mixing.

I have always been an enthusiastic proponent of wideband audio – it is one of the major potential advantages of VoIP over circuit switched telephony. Circuit switched calls are encoded with G.711, which yields 12 bits of effective dynamic range and a maximum frequency of about 3.5KHz. Human speech has harmonics even above 10KHz, which is why it is hard to tell the difference between an “F” and an “S” over the phone. The G.711 codec places an absolute limit on the sound quality of a regular phone call. A VoIP phone call can use a wideband codec, with whatever dynamic range and frequency range you want. There are several of them, commonly with a sample size of 16 bits and a sampling rate of 16KHz which captures a maximum audio frequency of 8KHz. When you have a good enough connection Skype uses a wideband codec by default, which is why it can sound better than “toll quality” (if you aren’t limited by your loudspeaker and microphone.)

Unfortunately, for the non-Skype world there’s a chicken and egg problem – almost no phones support wideband codecs, so the carriers aren’t motivated to support them either. Worse, any VoIP call that traverses the PSTN at any point is converted to G.711, losing the wideband frequencies. Worse yet, to cut costs most carrier implementations of VoIP use a bandwidth-saving codec that intrinsically delivers inferior sound quality to G.711; for example, last I heard Vonage was using G.729A.

As VoIP matures, and more and more calls are IP end-to-end through VoIP peering and ENUM arrangements (what Gizmo5 calls “back-door dialing”) wideband codecs will become more pervasive and our conversations will become clearer. The Wyde announcement is a step towards that world.