The panelists are Brian Donaghy of Appcore, LLC, Jan Lindén of Google, Hugh Goldstein of Voxbone and Danielle Morrill of Twilio.

The pitch for the panel is:

The FCC has proposed a date of 2018 to sunset the Public Service Telephone Network (PSTN) and move the nation to an all IP network for voice services. This session will explore the emerging trends in the Telco Cloud with case studies. Learn how traditional telephone companies are adapting to compete, and new opportunities for service providers, including leveraging cloud computing and Infrastructure as a Service (IaaS) systems that are being deployed with scalable commodity hardware to deliver voice and video services including IVR, IVVR, conferencing plus Video on Demand and local CDNs.

In related news, a group of industry experts is collaborating on a plan for this transition. The draft can be found here. I volunteered as the editor for one of the chapters, so the current outline roughs out some of my opinions on this topic. This is a collaborative project, so please contact me if you can help to write it.

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.

Voice over WLAN has been deployed in enterprise applications for years, but has yet to reach mainstream adoption (beyond vertical markets). With technologies like mobile UC, 802.11n, fixed-mobile convergence and VoIP for smartphones raising awareness/demand, there are a number of vendors poised to address market needs by introducing new and innovative devices. This session will look at what industries have already adopted VoWLAN and why – and what benefits they have achieved, as well as the technology trends that make VoWLAN possible.

The panelists are:

• Russell Knister, Sr. Director, Business Development & Product Marketing, Motorola Solutions
• Ben Guderian, VP Applications and Ecosystem, Polycom
• Carlos Torales, Cisco Systems, Inc.

All three of these companies have a venerable history in enterprise Wi-Fi phones; the two original pioneers of enterprise Voice over Wireless LAN were Symbol and Spectralink, which Motorola and Polycom acquired respectively in 2006 and 2007. Cisco announced a Wi-Fi handset (the 7920) to complement their Cisco CallManager in 2003. But the category has obstinately remained a niche for almost a decade.

It has been clear from the outset that cell phones would get Wi-Fi, and it would be redundant to have dedicated Wi-Fi phones. And of course, now that has come to pass. The advent of the iPhone with Wi-Fi in 2007 subdued the objections of the wireless carriers to Wi-Fi and knocked the phone OEMs off the fence. By 2010 you couldn’t really call a phone without Wi-Fi a smartphone, and feature phones aren’t far behind.

So this session will be very interesting, answering questions about why enterprise voice over Wi-Fi has been so confined, and why that will no longer be the case.

This year, three new products attempted to address this issue in remarkably similar ways – clearly an idea whose time has come. The products are Apple’s FaceTime, Cisco’s IME and a startup product called Tango.

In all three of these products, you make a call to a regular phone number, which triggers a video session over the Internet. You only need the phone number – the Internet addressing is handled automatically. The two problems the automatic addressing has to handle are finding a candidate address, then verifying that it is the right one. Here’s how each of those three new products does the job:

1. FaceTime. When you first start FaceTime, it sends an SMS (text message) to an Apple server. The SMS contains sufficient information for the Apple server to reliably associate your phone number with the XMPP (push services) client running on your iPhone. With this authentication performed, anybody else who has your phone number in their address book on their iPhone or Mac can place a videophone call to you via FaceTime.

2. Cisco IME (Inter-Company Media Engine). The protocol used by IME to securely associate your phone number with your IP address is ViPR (Verification Involving PSTN Reachability), an open protocol specified in several IETF drafts co-authored by Jonathan Rosenberg who is now at Skype. ViPR can be embodied in a network box like IME, or in an endpoint like a phone of PC.
Here’s how it works: you make a phone call in the usual way. After you hang up, ViPR looks up the phone number you called to see if it is also ViPR-enabled. If it is, ViPR performs a secure mutual verification, by using proof-of-knowledge of the previous PSTN call as a shared secret. The next time you dial that phone number, ViPR makes the call through the Internet rather than through the phone network, so you can do wideband audio and video with no per-minute charge. A major difference between ViPR and FaceTime or Tango is that ViPR does not have a central registration server. The directory that ViPR looks up phone numbers in is stored in a distributed hash table (DHT). This is basically a distributed database with the contents stored across the network. Each ViPR participant contributes a little bit of storage to the network. The DHT itself defines an algorithm – called Chord – which describes how each node connects to other nodes, and how to look up information.

3. Tango, like FaceTime, has its own registration servers. The authentication on these works slightly differently. When you register with Tango, it looks in the address book on your iPhone for other registered Tango users, and displays them in your Tango address book. So if you already know somebody’s phone number, and that person is a registered Tango user, Tango lets you call them in video over the Internet.

By “tampering” I mean any intentional changes to the media payload of a packet, and I am OK with the negative connotations of the term since I favor the “dumb pipes” view of the Internet. On phone calls the vast bulk of such tampering is transcoding: changing the media format from one codec to another. Transcoding always reduces the fidelity of the sound, even when transcoding to a “better” codec.

Networks vary greatly in the QoS they deliver. One of the major benefits of going with VoIP service provided by your ISP (Internet Service Provider) is that your ISP has complete control over QoS. But there is a growing number of ITSPs (Internet Telephony Service Providers) that contend that the open Internet provides adequate QoS for business-grade telephone service. Skype, for example.

But it’s nice to be sure. So I have added a “QoS Metrics” category in the list to the right of this post. You can use the tools there to check your connection. I particularly like the one from Voxygen, which frames the test results in terms of the number of simultaneous voice sessions that your WAN connection can comfortably handle. Here’s an example of a test of ten channels:

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.

Rethink mentions iCall and Skype as beneficiaries. Another notable one is Fring. Google Voice is not yet in this category, since it uses the cellular voice channel rather than the data channel, so it is not strictly speaking VoIP; the same applies to Skype for the iPhone.

According to Boaz Zilberman, Chief Architect at Fring, the Fring iPhone client needed no changes to implement VoIP on the 3G data channel. It was simply a matter of reprogramming the Fring servers to not block it. Apple also required a change to Fring’s customer license agreements, requiring the customer to use this feature only if permitted by his service provider. AT&T now allows it, but non-US carriers may have different policies.

Boaz also mentioned some interesting points about VoIP on the 3G data channel compared with EDGE/GPRS and Wi-Fi. He said that Fring only uses the codecs built in to handsets to avoid the battery drain of software codecs. He said that his preferred codec is AMR-NB; he feels the bandwidth constraints and packet loss inherent in wireless communications negate the audio quality benefits of wideband codecs. 3G data calls often sound better than Wi-Fi calls – the increased latency (100 ms additional round-trip according to Boaz) is balanced by reduced packet loss. 20% of Fring’s calls run on GPRS/EDGE, where the latency is even greater than on 3G; total round trip latency on a GPRS VoIP call is 400-500ms according to Boaz.

As for handsets, Boaz says that Symbian phones are best suited for VoIP, the Nokia N97 being the current champion. Windows Mobile has poor audio path support in its APIs. The iPhone’s greatest advantage is its user interface, it’s disadvantages are lack of background execution and lack of camera APIs. Android is fragmented: each Android device requires different programming to implement VoIP.

This will make international calls much cheaper for people who are willing to put up with the latency issues of the data channel.

The cellular companies want you to buy cellular service then pay extra to get signal coverage. Gizmodo has a coolly reasoned analysis.

AT&T Wireless is doing the standard telco thing here, conflating pricing for different services. It is sweetening the monthly charge option for femtocells by offering unlimited calling. A more honest pricing scheme would be to provide femtocells free to anybody who has coverage problem, and to offer the femtocell/unlimited calling option as a separate product. Come to think of it, this is probably how AT&T really plans for it to work: if a customer calls to cancel service because of poor coverage, I expect AT&T will offer a free femtocell as a retention incentive.

It is ironic that this issue is coming up at the same time as the wireless carriers are up in arms about the FCC’s new network neutrality initiative. Now that smartphones all have Wi-Fi, if the handsets were truly open we could use our home Wi-Fi signal to get data and voice services from alternative providers when we were at home. No need for femtocells. (T-Mobile@Home is a closed-network version of this.)

Presumably something like this is on the roadmap for Google Voice, which is one of the scenarios that causes the MNOs to fight network neutrality tooth and nail.

Consistent with this approach, we plan to take a fresh look at possibly authorizing VoIP capabilities on the iPhone for use on AT&T’s 3G network.

So why would anybody want to do VoIP on the cellular data channel, when there is a cellular voice channel already? Wouldn’t voice on the data channel cost more? And since the voice channel is optimized for voice and the data channel isn’t, wouldn’t voice on the data channel sound even worse than cellular voice already does?

Let’s look at the “why bother?” question first. There are actually at least four reasons you might want to do voice on the cellular data channel:

1. To save money. If your voice plan has some expensive types of call (for example international calls) you may want to use VoIP on the data channel for toll by-pass. The alternative to this is to use the voice channel to call a local access number for an international toll by-pass service (like RebTel.)
2. To get better sound quality: the cellular voice codecs are very low bandwidth and sound horrible. You can choose which codec to run over the data network and even go wideband. At IT Expo West a couple of weeks ago David Frankel of ZipDX demoed a wideband voice call on his laptop going through a Sprint Wireless Data Card. The audio quality was excellent.
3. To get additional service features: companies like DiVitas offer roaming between the cellular and Wi-Fi networks that makes your cell phone act as an extension behind your corporate PBX. All these solutions currently use the cellular voice channel when out of Wi-Fi range, but if they were to go to the data channel they could offer wideband codecs and other differentiating features.
4. For cases where there is no voice channel. In the example of David Frankel’s demo, the wireless data card doesn’t offer a voice channel, so VoIP on the data channel is the only option for a voice connection.

Moving on to the issue of cost, an iPhone unlimited data plan is $30 per month. “Unlimited” is AT&T’s euphemism for “limited to 5GB per month,” but translated to voice that’s a lot of minutes: even with IP packet overhead the bit-rate of compressed HD voice is going to be around 50K bits per second, which works out to about 13,000 minutes in 5GB. So using it for voice is unlikely to increase your bill. On the other hand, many voice plans are already effectively unlimited, what with rollover minutes, friend and family minutes, night and weekend minutes and whatnot, and you can’t get a phone without a voice plan. So for normal (non-international) use voice on the data channel is not going to reduce your bill, but it is unlikely to increase it, either.

Finally we come to the issue of whether voice sounds better on the voice channel or the data channel. The answer is, it depends on several factors, primarily the codec and the network QoS. With VoIP you can radically improve the sound quality of a call by using a wideband codec, but do impairments on the data channel nullify this benefit?

Technically, the answer is yes. The cellular data channel is not engineered for low latency. Variable delays are introduced by network routing decisions and by router queuing decisions. Latencies in the hundreds of milliseconds are not unusual. This will change with the advent of LTE, where the latencies will be of the order of 10 milliseconds. The available bandwidth is also highly variable, in contrast to the fixed bandwidth allocation of the voice channel. It can sometimes drop below what is needed for voice with even an aggressive variable rate codec.

In practice VoIP on the cellular data channel can sometimes sound much better than regular cellular voice. I mentioned above David Frankel’s demo at IT Expo West. I performed a similar experiment this morning with Michael Graves, with similarly good results. I was on a Polycom desk phone, Michael used Eyebeam on a laptop, and the codec was G.722. The latency on this call was appreciable – I estimated it at around 1 second round trip. There was also some packet loss – not bad for me, but it caused a sub-par experience for Michael. Earlier this week at Jeff Pulver’s HD Connect conference in New York, researchers from Qualcomm demoed a handset running on the Verizon network using EVRC-WB, transcoding to G.722 on Polycom and Gigaset phones in their lab in San Diego. The sound quality was excellent, but the latency was very high – I estimated it at around two seconds round trip.

The ITU addresses latency (delay) in Recommendation G.114. Delay is a problem because normal conversation depends on turn taking. Most people insert pauses of up to about 400 ms as they talk. If nobody else speaks during a pause, they continue. This means that if the one-way delay on a phone conversation is greater than 200 ms, the talker doesn’t hear an interruption within the 400 ms break, and starts talking again, causing frustrating collisions.
The ITU E-Model for call quality identifies a threshold at about 170 ms one-way at which latency becomes a problem. The E-Model also tells us that increasing latency amplifies other impairments – notably echo, which can be severe at low latencies without being a problem, but at high latencies even relatively quiet echo can severely disrupt a talker.

Some people may be able to handle long latencies better than others. Michael observed that he can get used to high latency echo after a few minutes of conversation.