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Virgin American In-Flight Internet Review, From In-Flight

Posted by timothy on from the latency-desires dept.

wintersynth writes "I've posted a review of Virgin America's in-flight internet provided by Gogo. Here's the scoop: Avg. .90 megabits/sec DL, .283 megabits/sec UL, ping: 130.6 msecs, $12.95 for the duration of the flight. Verdict: AWESOME. In fact, I'm posting this from 36,000 feet right now. Skype did not work for voice, even though I'm pretty sure those stats are over the minimums. Any ideas from the slashdotters on what might be going on?"

6 of 198 comments (clear)

  1. Skype by eldavojohn · · Score: 5, Informative

    You could be experiencing a difference of bandwidth versus latency. Although the two are related, you could be suffering high latency with Skype's servers. You might try pinging those servers compared to pinging www.google.com. If you are experiencing high latency, Skype uses UDP rather than TCP (like normal web traffic). If I remember correctly, UDP packets are many small packets which may perform badly over connections of very high latency. Your bandwidth readings on a TCP sight might look just large enough to use Skype but since it's a UDP service it could be unusable.

    Another possibility is that Gogo is demoting UDP traffic in some sort of QoS scheme to ensure that things like e-mail and regular HTTP traffic aren't slow or interrupted because 4 people are using Skype.

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    1. Re:Skype by Shakrai · · Score: 5, Informative

      Skype uses UDP rather than TCP (like normal web traffic). If I remember correctly, UDP packets are many small packets which may perform badly over connections of very high latency.

      UDP shouldn't have anything to do with latency, nor is it limited to "many small packets". UDP is just a transport protocol that lacks the error checking/data integrity and ordering mechanism of TCP. If such features are important to you then you need to use TCP or build them into your application that uses UDP.

      The advantage of UDP comes in time critical applications where it's probably better to lose a few packets (i.e: have a second or two of dead air during a phone call) than delay the transmission (conversation stops while it waits for the lost packets to be retransmitted). Latency really doesn't have anything to do with it, although latency is bad for VoIP for other reasons.

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    2. Re:Skype by Spazmania · · Score: 4, Informative

      I fired up Skype and dialed out. Massive failure. For some reason the sound is horrendously choppy and thin sounding. It was completely unusable.

      You're experiencing high "jitter." Jitter is the change in delay from packet to packet. If odd numbered packets take 100 ms and even numbered packets take 150 ms then you have 50ms of jitter.

      Certain protocols like VoIP and NTP require connections with low jitter in order to perform acceptably.

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    3. Re:Skype by amoeba1911 · · Score: 3, Informative

      Just in case anyone wants more info:
      http://www.networksorcery.com/enp/protocol/udp.htm
      http://www.networksorcery.com/enp/protocol/tcp.htm

  2. Southwest airlines testing now by Eharley · · Score: 5, Informative

    Southwest is testing Wi-Fi on four of its planes now. I was on one on a flight from Las Vegas to Baltimore. They sent me an email the day before telling me that the plane would have wi-fi and that it would be free during this test period.

    The speed was fantastic, but I didn't benchmark it. However, I was able to do a video iChat with my wife at home. Didn't try to do any audio, just video.

    The big drawback about Southwest is that their planes have no power outlets. Not sure if they're going to add them. But they're aware of the issue.

  3. Jitter Buffer by pathological+liar · · Score: 3, Informative

    Asterisk 1.4+ has a jitter buffer for at least IAX and SIP which helps to work around jitter in most cases. Given that they know what they're doing, I assume Skype does too.

    Jitter is (relatively) okay, it's packet loss that VoIP is particularly sensitive to. Packet loss at levels that will only mildly inconvenience most other traffic will screw up VoIP quite badly... there's no mention of packet loss in the article that I see, but I suspect that's what's causing the poor quality.