Wirevolution

Mobile Unified Communications

Subscribe!

Archive for the ‘QoS’ Category

VoIP on the cellular data channel

Thursday, September 17th, 2009

In a recent letter to the FCC, AT&T said that it had no objection to VoIP applications on the iPhone that communicate over the Wi-Fi connection. It furthermore said:

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.

Posted in AT&T, DiVitas, HD Voice, QoS, VoIP, controversies, dual-mode, impairments | No Comments »

Transparency and neutrality

Wednesday, February 4th, 2009

Google and the New America Foundation have been working together for some time on White Spaces. Now they have (with PlanetLab and some academic researchers) come up with an initiative to inject some hard facts into the network neutrality debate.

The idea is that if users can easily measure their network bandwidth and quality of service, they will be able to hold their ISPs to the claims in their advertisements and “plans.” As things stand, businesses buying data links from network providers normally have a Service Level Agreement (SLA) which specifies minimum performance characteristics for their connections. For consumers, things are different. ISPs do not issue SLAs to their consumer customers. When they advertise uplink and downlink speeds, these speeds are “typical” or “maximum,” but they don’t specify a minimum speed, and they don’t offer any guarantees of latency, jitter, packet loss or even integrity of the packet contents. For example, here’s an excerpt from the Verizon Online Terms of Service:

VERIZON DOES NOT WARRANT THAT THE SERVICE OR EQUIPMENT PROVIDED BY VERIZON WILL PERFORM AT A PARTICULAR SPEED, BANDWIDTH OR DATA THROUGHPUT RATE, OR WILL BE UNINTERRUPTED, ERROR-FREE, SECURE…

Businesses pay more than consumers for their bandwidth, and providing SLAs is one of the reasons. Consumers would probably not be willing to pay more for SLAs, but they can still legitimately expect to know what they are paying for. The Measurement Lab data will be able to confirm or disprove accusations that ISPs are intentionally impairing traffic of some types.

This is a complicated issue, because one man’s traffic blocking is another man’s network management, and what a consumer might consider acceptable use (like BitTorrent) may violate an ISP’s Acceptable Use Policy (Verizon:”…it is a violation of… this AUP to… generate excessive amounts of email or other Internet traffic;”). The arguments can go round in circles until terms like “excessive” and “unlimited” are defined numerically and measurements are made. So Measurement Lab is a great step forward in the Network Neutrality debate, and should be applauded by consumers and service providers alike.

Posted in Google, Policy, QoS, network, neutrality | No Comments »

Wi-Fi certification for voice devices

Thursday, July 3rd, 2008

In news that is huge for VoWi-Fi, the Wi-Fi Alliance announced on June 30th a new certification program, “Voice-Personal.” Eight devices have already been certified under this program, including enterprise access points from Cisco and Meru, a residential access point from Broadcom, and client adapters from Intel and Redpine Signals.

Why is this huge news? Well, as the press release points out, by 2011 annual shipments of cell phones with Wi-Fi will be running at roughly 300 million units. The Wi-Fi in these phones will be used for Internet browsing, for syncing photos and music with PCs, and for cheap or free voice calls.

The certification requirements for Voice-Personal are not aggressive: only four simultaneous voice calls in the presence of data traffic, with a latency of less than 50 milliseconds and a maximum jitter of less than 50 milliseconds. These numbers will produce an acceptable call under most conditions, but a network round-trip delay of 300 ms is generally considered to approach the limit of acceptability, and with a Wi-Fi hop at each end running at the limit of these specifications there would be no room in the latency budget for any additional delays in the voice path. The packet loss requirement, 1% with no burst losses, is a very good number considering that modern voice codecs from companies like GIPS can yield excellent sound quality in the presence of much higher packet loss. This number is hard to achieve in the real world, as phones encounter microwave ovens, move through spots of poor coverage and transition between access points.

Since this certification is termed “Voice-Personal,” four active calls per access point is acceptable; a residence is unlikely to need more than that. Three of the four access points submitted for this certification are enterprise access points. They should be able to handle many more calls, and probably can. The Wi-Fi Alliance is planning a “Voice-Enterprise” certification for 2009.

There are several things that are good about this certification. First, the WFA has seen fit to highlight voice as a primary use for Wi-Fi, and has set a performance baseline. Second, this certification requires some other certifications as well, like WMM power save and WMM QoS. So far in 2008, of 99 residential access points certified only 6 support WMM power save, and of 52 enterprise access points only 13 support WMM power save. One of the biggest criticisms of Wi-Fi in handsets is that it draws too much power. WMM power save yields radical improvements in battery life - better than doubling talk time and increasing standby time by over 30%, according to numbers in the WFA promotional materials.

Posted in QoS, Specifications, UMA, VoIP, Wi-Fi, dual-mode, enterprise, impairments, network, phones | No Comments »

Reliable VoIP

Friday, September 14th, 2007

QoS metrics are important, and several companies have products that measure packet loss, jitter, latency and so on. But you can have perfect QoS, and your VoIP system can still be defective for all sorts of reasons.

I spoke with Gurmeet Lamba, VP of Engineering, at Clarus Systems at the Internet Telephony Expo this week. He said that even if a VoIP system is perfectly configured on installation, it can decay over time to the point of unusability. Routers go down and are brought up again with minor misconfigurations; moves, adds and changes accumulate bad settings and policy violations.

VoIP systems are rarely configured perfectly even on installation. For example, IP phones have built-in switches so you can plug your PC into your desk phone. Those ports are unlocked by default. But some phones are installed in public areas like lobbies. It’s easy for installers to forget to lock those ports, so anybody sitting in the lobby can plug their laptop into the LAN. There are numerous common errors of this kind. Clarus has an interesting product that actively and passively tests for them; it monitors policy compliance and triggers alarms on policy violations.

Clarus uses CTI to do active testing of your VoIP system, looking for badly configured devices and network bottlenecks. Currently it works only on Cisco voice networks, but Clarus plans to support other manufacturers.

Clarus started out focusing on automated testing of latency, jitter and packet loss for IP phone systems. It went on to add help desk support with remote control of handsets, and the ability to roll back phone settings to known good configurations.

The next step was to add “Business Information,” certifying deployment configurations, and helping to manage ongoing operations with change management and vulnerability reports. Clarus’ most recent announcement added passive monitoring based on a policy-based rules engine.

Clarus claims to have tested over 350 thousand endpoints to date. It has partners that offer network monitoring services.

Posted in QoS, manageability, phones, security | No Comments »

WSJ on FMC

Thursday, May 3rd, 2007

Today’s Wall Street Journal has a good article about T-Mobile’s UMA trial in Seattle. It says that T-Mobile may be rolling it out nationally as early as next month, despite some trial particpants’ complaints about handoff and battery life issues. T-Mobile will be offering a home router to help with QoS and battery life. I presume that for the battery life this is just WMM Power Save (802.11e APSD) since that is what the phones in the trial (Samsung T709 and Nokia 6136) support. For QoS side I expect these APs will support WMM (802.11e EDCF), but they could also support some proprietary QoS on the WAN access link, the way that the AT&T CallVantage routers do, which would be interesting.

There is some background on the trial here.

The article goes on to put the trial into the context of other FMC deployments, from BT Fusion, Telecom Italia and Orange. The article quotes a Verizon Wireless spokesman saying that they aren’t convinced that Wi-Fi can deliver high enough voice quality to carry Verizon branded calls. This is amusing bearing in mind the usual quality of a cellular call in a residence.

The article also quotes Frank Hanzlik, the head of the Wi-Fi Alliance as saying that business FMC may have more potential than consumer. I agree.

Posted in FMC, Nokia, QoS, Samsung, T-Mobile, VoIP, Wi-Fi, cellular, dual-mode, enterprise, impairments | No Comments »

Dual-mode phones are the key to better-sounding calls

Monday, April 2nd, 2007

Potentially VoIP calls can sound radically better than what we are used to even on landline phones. So why don’t they? It may be lack of will. Some say the success of the mobile phone industry proves that people don’t care about sound quality on their calls. I don’t think this is a valid inference. All it proves is that people value mobility higher than sound quality.

The telephonic journey from mouth to ear, often thousands of miles in tens of milliseconds, traverses a chain of many weak links, each compounding the impairment of the sound. First, the phone. Whether it’s a headset, a desk phone or a PC, the microphone and speakers have to be capable of transmitting the full frequency spectrum of the human voice without loss, distortion or echo. Second the digital encoding of the call; it has to be done with a wideband codec. Third, the codec has to be end-to-end, so no hops through the circuit switched phone network. Finally the network must convey the media packets swiftly and reliably, since delayed packets are effectively lost, and lost packets reduce sound quality.

Discussions of VoIP QoS normally dwell mainly on the last of these factors, but the others are at least as important. The exciting thing about dual-mode cell phones is that they provide a means to cut through them. Because they must handle polyphonic ring tones and iPod-type capabilities, the speakers on most cell phones can easily carry the full frequency range of the human voice. Cell phone microphones can also pick up the required range, and DSP techniques can mitigate the physical acoustic design challenges of the cell phone form factor. Smart phone processors have the oomph to run modern wideband codecs. This leaves the issue of staying on the IP network from end-to-end. The great thing about dual-mode phones is that they can connect directly to the Internet in the two places where most people spend most of their time: at work and at home.

So if you and the person you are talking to are both in a Wi-Fi enabled location, and you both have a dual mode cell phone, your calls should not only be free, but the sound should be way better than toll quality.

Check out the V2oIP website for an industry initiative on this topic.

Posted in QoS, VoIP, Wi-Fi, phones, wideband | No Comments »

Wi-Fi Interference Experiments

Monday, March 12th, 2007

Interesting new series of white papers on Wi-Fi interference from Craig Mathias of the Farpoint Group. He set up a couple of clients and attempted various activities (file transfer, VoIP, video streaming) in the presence of interference from various sources (microwave oven, cordless phone, DECT phone, another AP, a Bluetooth headset) and characterized the impairments. His conclusions were that some interference sources can completely shut down some uses (almost all of them shut down video), but that interference can be managed and does not present a long term stopper to Wi-Fi.

Missing from the tests was 802.11n. This should make a huge difference, for several reasons. First, its MIMO operation is intrinsically more resistant to interference, second 11n operates both in the 2.4 GHz frequency range (like 11b/g) and in the 5 GHz frequency range (like 11a) . The 5 GHz waveband is immune from microwave oven interference, and most of the cordless phone interference. Its disadvantage of shorter range is mitigated by the multi-path amplification effect of MIMO.

Posted in QoS, Wi-Fi, impairments, interference | No Comments »

Linley Guide to Connectivity Chips for Cellular Handsets
The Internet Traffic Report monitors the flow of data around the world. It then displays a value between zero and 100. Higher values indicate faster and more reliable connections.