February 27, 2005

Peerio Patent Application

Previously many have taken issue with Popular Telephony’s Peerio and related GNUP. Some have criticized PT because the software was not made available after the public announcement; some have pointed out that the software does not work. One could explain them away by saying that these are execution lapses. Probably they will fix them; if they are incapable, somebody else could be brought in. In other words they are transitory. With this background, a few days back Andy informed us that a patent application filed by PT is now publicly available and that “should go a long way to telling their story and answering some questions.” James feels that the system should work as advertised because “chipmakers [and] the various phone and gateway makers [will not risk] public association with something that turns out not to work.” This entry is an analysis of the patent application as I understand it. Previously, I had questioned the need for and acceptability of a “serverless” architecture. This patent application has not changed my mind.

(In my limited understanding of the patent filing process, submitting the application does not imply much. I could develop a patent application for a perpetual motion machine and can file it, if I am willing to spare the application fee which is something like $100. Secondly, the process is far from granting a patent. Thirdly a patent does not have to describe the full system; only the aspect that is being patented. Accordingly I still have some open questions on the workings of the system.)

Concept of virtual ring

A Peerio network is a virtual ring. All the nodes in the ring have an address. The address space can be ordered. (The patent uses IP address.) This ordering is used to determine the neighbors of a node on the ring. Periodic query traffic (which can be sent in either direction) will maintain the integrity of the ring. If the ring is broken, the ring will be reordered. To reduce the maintenance traffic, the size of the ring is controlled. A ring that exceeds this limit will be split into two rings. A node joins an existing ring by identifying two nodes that “flank” its address. There is also a mechanism to join two rings.

Open questions:

  1. The default size of a ring is 6. So there has to be multiple levels of rings. It is not clear how are they formed.
  2. How will two rings know that they can merge? Presumably these bridge nodes will coordinate the merging process.

Data Storing

Data is stored in a sub-ring. This way of replicating data in multiple nodes provides for a node failure or leaving a ring. The members of the sub-ring are determined via a hash function or a similar mechanism.

This is the first hint that there are classes of nodes. These nodes are special in the sense that they have storage facilities.

Login procedure

A user who wants to use the services of Peerio will go through a login procedure at a Peerio node. The Peerio node can validate the user using the local data or getting data from other nodes. Once a user has successfully logged on, the user information is distributed to other nodes.

Open questions:

  1. The ring needs to have user profile for all potential users. Since a ring is organically formed, it is not clear how the first node in the first ring will get the profile for all the users?

Call Establishment

When user A wants to call user B, node A will get the profile of user B from the ring. The profile will give the IP address of node B and the status of user B. Using this information, node A can determine whether to establish a session or not. In the affirmative case, node A will use one of the standard mechanisms (H.323 or SIP) to establish the call. Otherwise node A will collect “voice mail data” and store it in the ring.

One of the criteria for not establishing the call is the busy status of user B. This means, Peerio call processing logic has to replicate telephony features like call waiting, multiple call appearances, instead of using those capabilities available in H.323 or SIP. More importantly, these features have implications to user interface. How will the terminal coordinate the user interface projected by Peerio and that by H.323/SIP?

Voice Mail

As I remarked in the Call Establishment section, node A can encrypt the voice mail and store it in the “data store sub-ring”. If the user is logged on when a voice mail was stored, the node will be informed of the arrival of a new message. Otherwise, the user will be notified of stored messages during the login process.

Open questions:

  1. How will the system ensure that there is at least one “data store” terminal is available at any given time?
  2. What happens if all the terminals that held messages for user A are unavailable when that user logs in?

Call Center Application

Call request for the common user id is forwarded from one terminal to the next in “hot potato” fashion, till a free agent is located.

Sundry observations

  • Conferencing is done through multicast and not standard bridging
  • There multiple kinds of terminals - Standard, Data store, Administrator
  • A major implication of this is that one needs to engineer so that there are enough terminals of each kind. It is not as simple as buying a terminal at Walmart and connecting it to the network. People have theorized that it is targeted for SMB. An open question is will SMBs have enough technical knowledge to do this and other admin functions.
  • Posted by aswath at 01:05 PM | Comments (4)

    February 23, 2005

    Creating a Business out of a Hint from Heloise

    The other day I came across an interesting product in a SkyMall catalogue, called Power Strip Liberators (catalogue number 238359). I am sure you have faced the situation of trying to plug multiple peripherals into a power source, but couldn’t because they have a bulky transformer that takes up two or three of the outlets. The Power Strip Liberator, which is nothing more than a 13” long mini-extension cord, eliminates that problem. You plug the transformer into the Liberator's grounded outlet, and plug the Liberator's grounded plug into the power strip. The catalogue item comes with a bundle of 5 cords and costs $13. The problem is real and the solution is legitimate. Still I wonder whether there is a market for this. (I know pet rocks and Baby on board decals sold a lot.)

    Instead of buying this, I suppose one can use power cords from older, discarded machines like PCs and monitors. Alternatively, one can daisy chain power strips. Since only the anchor power strip needs to be a “good” quality, others could be as inexpensive as possible. I located some for $3 in some stores. So there is an analogous solution that is either inexpensive or free.

    VoIP is in the same boat. Once the word gets out that with the advent of always on connection to the Internet, two or more people can communicate among themselves without the need for a service provider. This is the dilemma facing the industry.

    Posted by aswath at 07:30 PM | Comments (2)

    February 15, 2005

    Skype Math Anxiety

    Yesterday Stuart did a simple arithmetic on Skype usage to ask (rhetorically) whether mobile operators can keep up with Skype usage. This note is not related to that question specific, but an attempt to understand the arithmetic.

    First a recap of his observations:

    • 2M active users concurrently online
    • Skype serves 3M minutes per hour

    His assumption:
    • An user logs on for 12 hours in a day

    My assumptions (due to lack of knowledge of precise definition):
    • Active user and logged on user are the same
    • One minute call between two users contribute to 1 minute to served minutes count (I doubt it because double counting is straight forward; but this assumption is against me, so I will take it.)

    Stuart builds his analysis the following way:
    • Each user generates 1.5 minutes per hour
    • Or 36 minutes per day
    • Since the user is active only 12 hours, a user generates 72 minutes per day

    Me thinks:

    • At any time 1M calls going on and they generate 3M minutes in an hour
    • So each user generates 3 minutes per hour
    • Since the user is active only for 12 hours per day, each user generates 36 minutes per day.

    So what gives? Also note that if they are double counting the minutes (as I suspect) then each user generates only 18 minutes per day.

    There is an unobserved nugget buried in Stuart’s assumptions. If a Skype user is logged on for 12 hours per day, then 4M users log on per day out of 70.5M downloads – 5.7%. Just an observation, with nothing implied.

    Posted by aswath at 02:09 PM | Comments (1)

    February 11, 2005

    VoIP (as practiced today) is “Intelligent” After All

    Last week Tom Evslin wrote why VoIP will be known for its features. That entry is made up of three components:

    1. The list of features his current VoIP service provider (Vonage, but it is incidental) make them available to him are very useful to him.
    2. The price difference between PSTN and VoIP is inconsequential. (It is not clear whether he anticipates that these price differences will disappear in time or that the feature set will be so attractive that potential users will not even look at the price advantage.)
    3. PSTN service providers, hobbled by the Intelligent Network, will never be able to match the features facilitated by VoIP technology.

    The purpose of this note is to take issue with some of the logic and to suggest that for VoIP to truly deliver on its promise, we have to think differently.

    The following is a summary of the features he likes, followed by one way I think how PSTN may be able to offer the same feature without “shoehorning” them into their “intelligent” network.

    • He really likes the portability feature. He is able to go to his beach house or the winter place and receive calls meant for the primary residence. More importantly, there is really no setup excepting connecting the ATA into the network at the new place.
      Of course a PSTN service provider can easily imitate this feature by asking the subscriber to dial an 800 number which will note down the number of the new place and automatically provision the call forwarding feature at the switch associated with the primary residence. Since I am assuming flat rate pricing for PSTN as well, there is no need for additional charge for forwarded calls. Of course this scheme may not work if the new place is behind certain PBXs; but then again VoIP has its own difficulties in the corporate environments due to firewalls.
    • VoIP service providers offer simultaneous ringing feature. Indeed early SIP documents called this “forking” and many in the community were claiming this to be a unique capability of SIP (not VoIP) architecture.
      Excepting that Cincinnati Bell (now you can guess how long ago) offered this as a tariffed service and either they or their partners had a patent on it as well. Of course in the PBX environment this was a standard feature.
    • The third feature he benefits from is the fabled Virtual numbers. He states that he does not incur any usage charges. Of course this is true, but we know that this service is viable because of the access charge regime that many VoIP proponents oppose.
      Be that as it may. This feat is not accomplished by VoIP technology. After all the call directed to the virtual number comes to a PSTN switch (most probably a CLEC partner) which forwards it to the designated end-point. A PSTN service provider could do the same, except the current regulatory regime may very well prohibit an ILEC to strike a partnership with another service provider.
    • Finally, his VoIP service provider offers all the features available in PSTN. But more importantly it is very easy to manage these features and there is a night and day difference when comparing the user experiences between VoIP and PSTN environments.
      He fails to mention the enhanced features available to ADSI phones (granted a market failure; but we are comparing technologies here) or new services like Verizon iobi.

    He concludes by saying that even if all these features were/are/can be implemented in PSTN, it is “almost impossible to shoehorn [them] into the over-complex “intelligent” networks of the traditional carriers, it is the VoIP providers like Vonage who are making these features practically available.” This veiled reference to “The Rise of Stupid Network” is standard fare and has become this industry’s “shibboleth”. From an architectural point of view there is no difference between PSTN and VoIP. In IN-PSTN architecture, there are SSP – consisting of call control logic and switching fabric, SCP and IP (Intelligent Peripheral). VoIP architecture used by service providers consists of Proxy servers/Gatekeeper/Call Agent and media servers. The routers take on the role of switching fabric; but they are not so tightly integrated with the call control logic. But from feature invocation and execution point of view, the switching fabric is not all that critical. It is in this respect I claim that VoIP architecture is as “intelligent” as the traditional PSTN. So, if Vonage and others offer these features while traditional PSTN service providers procrastinate, the explanations lie elsewhere. In other words, invoking “intelligent” network comes across as “sibboleth”.

    My intention here is not to be argumentative. When Tom speaks others listen. Hence I am concerned that we as a community may focus our attention on wrong things. It is true that certain features are possible only with IP Communication; PSTN can never hope to provide them. The most important of these is that IP Communication facilitates any-to-any communication. Here I do not mean P2P – a hijacked term that requires many nodes– and “serverless” – which requires many in the community. Let me coin a new term – “autonomous”, wherein two or more communicate directly with each other without the intervention of any special service providers or special nodes but use well established IP services provided by commodity providers. Secondly, with IP Communications one can use any application(s) and any granularity of network resource; PSTN at most can offer services that are a multiple of a specific base capability. Finally, the current level of technology and its price points, allows us to offer sophisticated user interface. None of the current set of VoIP service providers allow for any of them. Indeed, they behave like the hated “Bellheads” and charge differently for different applications. So let us focus on these features now instead of waiting for the mythical “tipping point”.

    Posted by aswath at 11:44 PM | Comments (0)

    February 06, 2005

    Bellster fwdOUT Revisited

    In my last post, I had raised the possibility that one application of fwdOUT may not be permissible in US as well, not just Singapore and India. That application is patching a VoIP call on to PSTN. At that time I did not have definitive reference and indicated that since pulver.com has legal counsel, we could assume that there shouldn’t be a problem. Still, I posted this question to one of Pulver’s entries in his blog. Seemingly as a response to this, Pulver posted a link to a story in a Long Island paper that quotes Isenberg saying that to his knowledge there are no legal barriers to the use of BellsterfwdOUT. Now Daniel Ryan weighs in by quoting a tariff from Qwest to suggest that such patching may not be allowed. The LI paper quotes Isenberg as saying "The law of unintended consequences looms large." Indeed.

    In another entry, Pulver gives a hint of another perfectly acceptable use: interconnecting large enterprises (he asks us to “think Universities”). Now many enterprises have interconnected their PBXs distributed over multiple locations using private lines. VoIP eases such interconnections even when the traffic is not sufficient to warrant deploying private lines. SIPPhone had introduced this capability sometime during August/September of last year and for the same purpose. Most of the concerns raised in the blogosphere do not apply to this application: enterprises can easily afford “1 day, 1K, 1 person” to set the system up; since they are interconnecting private networks, no concern about legal matters. But then for this application there is no need for credit management that fwdOUT provides. For that matter there is no need for intermediation from SIPPhone or fwdOUT.

    Posted by aswath at 08:39 AM | Comments (0)

    February 01, 2005

    Bellster misunderstood, but piling on continues

    Recently Pulver announced a new service called Bellster. I am sure by now you “know” what it is and might have even read many friendly articles. But there have been occasional blog entries cautioning on the potential issues with this service.

    For example Technology Futurist writes about the potential high cost of entry and the security issues. On second thoughts, he also questions the economic viability in many telecom markets. Martin says that this is not a new idea, but his main concern is also potential abuse and legal implications to the donor. Om, after personally trying it out is decidedly under whelmed. He also refers James Seng to point out in may violate the Terms and Conditions of the donor’s local service. I basically said the same thing in a comment to another of Om’s entries. It is not just Singapore and India where it may not be permissible to “donate” local access. I suspect that this is the case even in US. I asked for T&C from Verizon; but they are giving me run around. Since Pulver has sufficient legal resources, it is possible that this may not be an issue. But I remember at one time the local telephone book expressly prohibiting stringing two houses and sharing one access line. But that section is not there in the recent phone books. So these are all potential show stoppers of this service.

    Pulver’s response to all these points is an intriguing one sentence reply: “Sometimes it is better being misunderstood.” Is he suggesting that there is something else to this service than meets the eye? If all these people who are active in the field miss it, are his customers clued in on? To be fair, Pulver has been talking about this kind of application from day one. Indeed, he has said that he was attracted to VoIP in the first place because of his ham radio background and the satisfaction he got when he used to do phone relay for others. The Internet Phone Patch sold by one of his enterprises is a precursor. Bellster has many security mechanisms built in to address the many issues raised by others. For example, the donor can filter based on who is calling, the destination number, time of the day, duration of call and so on. I suspect that he is using an Asterisk to handle all these logic. Potential misuse is minimized because Bellster can identify the caller. The only issue I am not able to resolve is related to T&C. So what is the real objective of Bellster? Pulver has talked many times about social networking and has been actively pushing services like Linked In. So I think he is going to integrate FWD, Pulver Communicator, Bellster and some social network into one big ecosystem (Finally I managed to use this buzzword.) By the way Martin anticipates this in his article as well. Notwithstanding Andy’s efforts to preempt guesswork on our part, we continue..

    Posted by aswath at 12:15 PM | Comments (0)

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