Did I say it was "one phone call"? She was here working, running her VPN while two other people were doing all the usual stuff of surfing, email and downloading. Her calls run for hours. So I think we did a fairly good test.
You would think they would provide an app on their web site to test the connection wouldn't you?
Normally I only need one at a time, so I take the unit with me and only need one extension.
On the various provider's web pages they list things like "Included channels - 3". Does that mean three extensions or simultaneous calls?
Yeah, but I'm not in NY... Oh, my use of NetTalk about is a typo, that should have been CallCentric.
NetTalk is a service sold retail that I am going to buy today if they have it in the store. At $30 a year, it beats any of the "low price" providers... if you aren't in NY.
On Mon, 19 Aug 2013 08:39:07 -0700, Jeff Liebermann wrote: Jeff, Just one quick question - How do you interface your POTS phone to a VoIP box? Do you know of a readily available phone line interface card?
If you read my previous rants on VoIP, you'll notice that I'm very reserved when I mention Skype. That's because Skype is the exception to every rule, standard, methodology, acceptable practice, and philosophy found among VoIP vendors. From my perspective, Skype is the AOL of VoIP. They provide a functional proprietary system, reasonable service, and very good pricing. Everything else is marginal or worse. I've done video broadcasts and online conferences with Skype and come away disappointed. Webex is MUCH better. If Skype ever opened their system to standard SIP phones and/or allowed Skype to SIP gateways, they would be gone rather quickly (unless Microsoft keeps them afloat).
Webex uses centralized reflector type conference servers with plenty of bandwidth for each connection. A large part of the cost is the bandwidth needed to do a proper video conference call with a large number of users. Skype uses a distributed model, where the originator acts as the central conference server. If the originating user has plenty of bandwidth, it might work. Otherwise, it will rapidly become overloaded. I once tried to hit the 25 connection limit for conference calls on an ordinary 3Mbit/sec DSL connection. I never made it past about 10 connections. See Fig 5. The Skype model has already broken down with too many users and too few supernodes: Microsoft is moving away from peer to peer and installing central servers. Until the transition is finished, Skype conferencing is still a crap shoot.
Incidentally, I couldn't find anything on the Democracy Now web pile that mentions they use Skype for podcasts.
I beg to differ somewhat. What the user hears is the voice. If the underlying technology isn't together, the voice suffers. Call a VoIP vendor and try to complain that their protocols are broken (as I've done a few times). They act dumb or claim it's someone elses headache. The VoIP vendors know about voice, not networking.
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Jeff Liebermann jeffl@cruzio.com
150 Felker St #D http://www.LearnByDestroying.com
Santa Cruz CA 95060 http://802.11junk.com
Skype: JeffLiebermann AE6KS 831-336-2558
Play the video at: I think you can handle the delay and possibly half duplex.
You probably don't recall the 1950's style long distance calls, where half duplex was the norm. Cell phones do the same thing. When the error rate or lost packets start to climb, they revert to half duplex before giving up and dropping the call. In a cell phone to cell phone call, the accumulated latency can easily add up to a full second. It's still full duplex but you still have to tell the other person when it's time to talk. I have no problem using half duplex or saying over. However, we will probably need to expose an entire generation to it before it will be considered acceptable.
Some day, computahs will be powerful enought to anticipate what we will say next. Then, we can eliminate the latency and half duplex problem by producing that the caller would be expected to say.
Exede Speedtest.net video: Note the 800 msec latency.
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Jeff Liebermann jeffl@cruzio.com
150 Felker St #D http://www.LearnByDestroying.com
Santa Cruz CA 95060 http://802.11junk.com
Skype: JeffLiebermann AE6KS 831-336-2558
Analog Telephone Adapter. Freestanding units start in the ~$30 range (of varying quality).
Some modems can act as FXS gateway/adapter -- though you are then stuck with the provider on the other side of that "modem". Ditto for routers. (e.g., I have a couple of cable modems with this capability as well as some "routers". Typically, support two "lines")
I'm more interested in the other side of the equation: the FXO adapter (let me interface a network to the PSTN). Freestanding units introduce more latency (if they aren't *directly* feeding a VoIP "switch"/exchange). I've yet to find a reference design that isn't terribly tied to a particular architecture...
Here's a better article on how Skype (and others) work in conference mode. See Fig 2:
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Jeff Liebermann jeffl@cruzio.com
150 Felker St #D http://www.LearnByDestroying.com
Santa Cruz CA 95060 http://802.11junk.com
Skype: JeffLiebermann AE6KS 831-336-2558
There are 3 ports on the common ATA adapter. Phone 1 Phone 2 Internet Run a cable from the Internet port to one of the LAN ports on your existing router. That's it for the networking part of the puzzle.
Just plug a POTS (plain old telephone service) phone into a Phone port. Or you can unplug your current POTS provider at the NID (network interface device), and replace the connection with an RJ11 cable from the Phone port. Do not connect an ATA to the house wiring if you still have a connection to your previous POTS service provider.
To save space, some ATA's come with L1 and L2 on the same RJ14 connector. If you're planning two lines or two instruments, you may need a line splitter/adapter to connect both phones.
If you get an ATA adapter that has a built in router, then life gets a bit more complicated. The Internet cable goes to your cable or DSL modem, and the Ethernet cable would go to your router or ethernet switch. There are lots of ways to interconnect the networking part of the puzzle, so I won't try to list all the other possible combinations. Things also get even more complex if you're using PoE (power over ethernet).
Hint: Plan your wiring for future expansion:
You ask for an "interface card" which is in never humble opinion, a bad idea. ATA's are generally a separate box, not a PC card. Cards do exist, but they are designed to interface to ISDN, T1, fiber, or other line protocols beyond POTS. I don't think that's what you mean. If you are simply trying to use a computer as a VoIP phone, there are various softphone programs that will do that. However, none of them provide a POTS to sound card mic/earphone jack interface. You can either build one or try to find a "sound card to telephone line adapter". Also search for "phone patch". Not quite but close: Could you please describe what you are trying to accomplish and what you have to work with?
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Jeff Liebermann jeffl@cruzio.com
150 Felker St #D http://www.LearnByDestroying.com
Santa Cruz CA 95060 http://802.11junk.com
Skype: JeffLiebermann AE6KS 831-336-2558
I've read all of Bamfords NSA books. He knows his stuff. "Body of Secrets" was probably the best of the bunch, especially if you never read the accounts of the USS Liberty.
"Democracy Now" podcasts are just regular downloads. But sometimes the guests on the show are fed via Skype. They will mention this on the lower third just so you know why the video sucks.
The show is also distributed live to cable systems via satellite, so all the flubs are present. This way when someone on cable sees screwed up video, they don't call the cable company to complain because they know it is skype.
I don't think that is 800ms. I think we're talking Microsoft style CES demos, i.e. faked!
I never met a happy satellite internet user, well other than they are happy to get anything at all. Once anything else is available, the satellite internet is the first thing to go. But that means more free dishes for me to play with. If you go on Craigslist, people spend a long time trying to sell old satellite internet gear before just dumping it.
That's why I posted a video instead of a screen dump. It looks quite real to me.
I just sent email to my customer with Exede satellite internet service asking him to run the same speed test. We'll see what he gets.
Incidentally, here's my seriously depressing DSL speeds.
I know several that are quite happy. However, I agree. If there's nothing else available, satellite is a good last resort. If you drive from Boulder Creek up Bear Creek Road. The end of the Comcast cable is at the mess of solar panels running the water pumping station (Greenview Dr). Beyond that, all I see on the rooftops are satellite internet dishes. Many houses have two (for TV and internet). I've moved some HughesNet satellite internet users on Ku band, to Exede (Wild Blue) which offers much faster service on Ka band. I've also installed two 5.6 GHz links across the canyon in the area to extend Comcast internet service to areas without cable.
True. I had some early Hughesnet and DirecPC hardware that I couldn't sell. I tried to give it away on Santa Cruz Freecycle, and nobody wanted it. It went to the recyclers.
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Jeff Liebermann jeffl@cruzio.com
150 Felker St #D http://www.LearnByDestroying.com
Santa Cruz CA 95060 http://802.11junk.com
Skype: JeffLiebermann AE6KS 831-336-2558
The one way delay is about 240 to 280 msec depending on where you're located on the planet. The closer to equator, the shorter the distance, and the closer to the 240 msec time. That's from the ground, to the bird, and back to the ground, one way. The ping delay (latency), for a round trip, will be twice that value, plus anything added by terrestrial network delays (typically 15 to 60 msec). So,
638 and 800 msec are quite reasonable values.
If you want a more accurate value: ViaSat-1 is at -115W longitude. The Colorado Springs ground station is at about 39.9N, 104.8W. That yields a one way distance of: 2 * 37500 = 75,000 meters for a latency of: 75*10^3 / 3*10^5 km/sec = 250 msec
So, 638 and 800 msec are fairly reasonable values.
--
Jeff Liebermann jeffl@cruzio.com
150 Felker St #D http://www.LearnByDestroying.com
Santa Cruz CA 95060 http://802.11junk.com
Skype: JeffLiebermann AE6KS 831-336-2558
Presumably free calls to endpoints on SIP services. you'd have to ask them if that includes SIP enpoints that they are not currently aware of or only SIP endpoints directly connected or peered with them.
Most SIP providers allow free calls amongst their peered endpoints.
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