I will be moderating a panel on this topic at ITExpo East 2012 in Miami at 3:00pm on Thursday, February 2nd.
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.
I will be moderating a session at ITExpo West on Monday 4th October at 2:15 pm: “The State of VoIP Peering,” will be held in room 304C.
Here’s the session description:
VoIP is a fact – it is here, and it is here to stay. That fact is undeniable. To date, the cost savings associated with VoIP have largely been enough to drive adoption. However, the true benefits of VoIP will only be realized through the continued growth of peering, which will keep calls on IP backbones rather than moving them onto the PSTN. Not only will increased peering continue to reduce costs, it will increase voice call quality – HD voice, for instance, can only be delivered on all-IP calls.
Of course, while there are benefits to peering, traditional carriers have traditionally not taken kindly to losing their PSTN traffic, for which they are able to bill by the minute. But, as the adoption of IP communications continues to increase – and of course the debate continues over when we will witness the true obsolescence of the PSTN – carriers will have little choice but to engage in peering relationships.
This session will offer an market update on the status of VoIP peering and its growth, as well as trends and technologies that will drive its growth going forward, including wideband audio and video calling.
The panelists are:
This is shaping up to be a fascinating session. Rico can tell us about the hardware technologies that are enabling IP end-to-end for phone calls, and Mark and Grant will give us a real-world assessment of the state of deployment, the motivations of the early adopters, and the likely fate of the PSTN.
ARCchart has just published a report summarizing the data from a “test your Internet speed” applet that they publish for iPhone, Blackberry and Android. The dataset is millions of readings, from every country and carrier in the world. The highlights from my point of view:
- 3G (UMTS) download speeds average about a megabit per second; 2.5G (EDGE) speeds average about 160 kbps and 2G (GPRS) speeds average about 50 kbps.
- For VoIP, latency is a critical measure. The average on 3G networks was 336 ms, with a variation between carriers and countries ranging from 200 ms to over a second. The ITU reckons latency becomes a serious problem above 170 ms. I discussed the latency issue on 3G networks in an earlier post.
- According to these tests, Blackberries are on average only half as fast for both download and upload on the same networks as iPhones and Android phones. The Blackberry situation is complicated because they claim to compress data-streams, and because all data normally goes through Blackberry servers. The ARCchart report looks into the reasons for Blackberry’s poor showing:
The BlackBerry download average across all carriers is 515 kbps versus 1,025 kbps for the iPhone and Android – a difference of half. Difference in the upload average is even greater – 62 kbps for BlackBerry compared with 155 kbps for the other devices.
Source: ARCchart, September 2009.
I have been calling myself a lot recently, because I am chairing a panel on network interconnection at Jeff Pulver’s HD Communications show this week, and I wanted to get some real-world experience. The news is surprisingly good.
I subscribed to several VoIP service providers, and Polycom was kind enough to send me one of their new VVX 1500 video phones. So with the two Polycom phones on my desk (the other, an IP 650, is the subject of my HD Voice Cookbook) I was able to make HD Voice calls to myself, between different VoIP service providers.
All the calls I made were dialed with SIP URIs rather than phone numbers. Dialing with a SIP URI forces the call to stay off the PSTN. This means that the two phones are theoretically able to negotiate their preferred codec directly with each other. For these particular phones the preferred codec is G.722, a wideband codec. The word “theoretically” is needed because calls between service providers traverse multiple devices that can impose restrictions on SIP traffic – devices like SIP Proxies and Session Border Controllers. I presumed that HD compatibility would be the exception rather than the rule, but it turns out I was wrong about that. Basically all the calls went through with the G.722 codec except when the service provider’s system was misconfigured. Even more pleasingly, I was able to complete several video calls between the X-Lite client on my PC and the Polycom VVX 1500 (though the completion was random at about a 50% rate), and when I had a friend from Polycom call me from his VVX 1500 using my SIP address, the call completed in video on the first attempt.
Effectively 100% of VoIP calls made from phones are dialed using E.164 (PSTN) phone numbers, and consequently wideband codecs are almost never used (Skype is the huge exception, but Skype calls are normally made from a PC, not a phone). The benefit of E.164 addressing is that you can call anybody with a phone. What I learned from my experiment is that with SIP addressing you can call anybody with Internet connectivity, and have a much better audio experience.
This is somewhat surprising. Many engineers consider the Internet to be too unreliable to carry business-critical phone calls, and VoIP service providers like to interconnect directly with each other using peering arrangements like the Voice Peering Fabric and Xconnect.net. There is an exhaustive series of articles about VoIP Peering at VoIP Planet.