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Mobile Unified Communications

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Archive for the ‘VoIP Peering’ Category

3G network performance test results: Blackberries awful!

Thursday, October 1st, 2009

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:

  1. 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.
  2. 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.
  3. 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.

VoIP Peering

Sunday, May 17th, 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.