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Unread 02-02-2008, 02:56 PM   #1
v164
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Default VoIP situation in Japan - how to improve with ENUM and peering.

It's well known that there are large numbers of VoIP users in Japan. However the current arrangements don't enable VoIP users to fully benefit from the possibilities that VoIP enables. I am hopeful that VoXaLot / SIP Broker, e164.org, sipix.jp, and other like-minded groups can help to enable ENUM and VoIP peering for consumers in Japan.

I'll start by explaining the current SIP / VoIP situation, so it can be seen what VoXaLot / SIP Broker et al could do to facilitate improvements. I write this from a consumer's perspective, as someone living in Japan and having seen and used these services, as well as the way the services are marketed.


The situation for consumers is summed up concisely in this blog posting from Oct 18th 2005:

Japan VoIP plan not all it's cracked up to be, insiders say - The VoIP Weblog

Quote:
Japan's (and the world's) largest commercial VoIP network (in terms of subscribers) is Yahoo BB. They don't use SIP but a modified MGCP which only works with hardware manufactured specifically for Yahoo BB and distributed exclusively by Yahoo BB. It also only works with Yahoo BB internet service as they are actually an ISP. Similarly, the other VoIP services use a provider-locked SIP derivative. It looks just like SIP, but it will not recognise any equipment other than the "official" black box of the service provider.

And the worst thing is that all Japanese VoIP services are locked to your ISP and they cannot be unbundled because their protocols and equipment are designed this way. The only VoIP provider you can get service from is your ISP. If you want to choose a different one, you have to change ISPs first. And since the ATA they provide to you is built right into your ADSL modem, you can only use your VoIP service at the location where the ADSL modem has been installed. It won't work anywhere else.

Background - widespread broadband use in Japan.

Broadband services in Japan (ADSL, cable, fibre optic), providing a high speed, permanent Internet connection with unlimited downloads and uploads are plentiful and cheap.

Every major residential broadband offering comes with the option of a bundled "IP Phone" VoIP service - there are millions of "IP Phone" subscribers in Japan. The provider with the largest number of subscribers would be Softbank's "BBphone" service, with around 5 million subscribers. Another provider, "OCN", with their "dot phone" service, states that you can call "4 million" subscribers for free. All up, there are probably around ten million or more "IP Phone" subscribers in Japan. The number of PSTN phone lines in Japan is around eighty million, so that means that around one in eight PSTN phones in Japan are connected to an ATA, ready to receive calls via VoIP.

With a bundled "IP Phone" service, the provider either supplies a locked ATA for the service (eg, Softbank "YahooBB" ADSL, cable internet) or in the case an ISP accessed through NTT "Flets" (ADSL / fibre optic), supplies a list of "approved" NTT-branded ATAs (which are not locked (they can be set to use any SIP server) but are specifically designed for the "IP Phone" service (eg dial plan, etc is hard coded)), which the subscriber can either rent or purchase from NTT. Some ATA's are stand-alone devices, that would connect to a router (and use uPnP to open the necessary ports), although most are combined with a router / bridge and/or ADSL modem, and so have a public IP address. In some cases, the "IP Phone" adapter (ATA) has it's own public IP address, separate to the public IP address assigned to the subscriber's PC or router.

The ATA has one FXO and one FXS port. There's never any discussion about codecs - everyone uses G711 ulaw (the native codec of the PSTN in Japan), and in many cases, this is the only codec supported.

In some cases, an extra monthly fee (500 - 700yen) is charged for the "IP Phone" service (which in some cases covers rental of a supplied ATA). Call charges are typically 8.4yen per 3-minute block for calls to Japanese landlines. This is around the same price of a local call on the PSTN in Japan. For comparison, an economical PSTN long-distance provider (G-Call) charges 15.75yen per 3-minute block.

All "IP Phone" services give you an 050 phone number. This is a Japanese domestic phone number, with the format 050 xxxx xxxx (in International format, +8150xxxxxxxx). The 050 number range is a new non-geographic number range used for "IP Phone" service DIDs. Calls to your 050 phone number arrive at your ATA via VoIP. The first four digits after "050" identify the provider of the "IP Phone" service, so it is possible to tell, by looking at the 050 number, which provider the subscriber is with, which is important for determining whether or not you can call them for free (no porting of 050 numbers (yet)).

Common to all "IP Phone" services, calls between subscribers of the same provider are free. Most providers (except Softbank's BBphone) have alliances with some other IP Phone providers ("business partners"), for free calls between their subscribers. There are about three or four major groupings of providers whose subscribers can call each other for free, a rough breakdown being:

1. Softbank "BBphone"

2. Plala "Plalaphone" + OCN "dot phone" + so-net

3. KDDI - related (including some Cable TV internet providers that resell KDDI's VoIP service)

4. others


Just about all providers have mutual connection agreements with the other providers, however in most cases these are not free. The cost of calling a non-free 050 number is (typically) 8.4yen / 3 minute block (ie, the same price as calling their PSTN number). (This means that you can end up paying timed call charges for a call that doesn't touch the PSTN).


Although calls between subscribers with the same provider are free, getting these calls for free is actually easier said than done. The reason is because in order to call the person for free, you need to know their 050 phone number (except in the case of Softbank's "BBphone" service - see below). If you have the PSTN number of the person you want to call, there is a reasonable chance that the phone you're calling is connected to an ATA that's subscribed to the same "IP Phone" service as you (or one of the free affiliates), however, because you have no way of knowing this, and no way of knowing the 050 number, it is impossible for you make the call for free.

People tend not to tell others their 050-xxxx-xxxx phone number, for reasons such as:

- It's inconvenient. The number is long, unfamiliar (you have to explain it to people), and (perhaps most significantly), the number will be gone if you leave your ISP. It's not uncommon for subscribers to not know or remember their 050 number. If asked for it, they'll typically look it up among the papers they got from their ISP, or look it up in their mobile phone.

- In many cases (particularly local / in-prefecture calls, or calls from overseas), it costs callers more to call your 050 number than your normal PSTN number.

(An exception is some people who have an IP Phone service, but no PSTN service (eg Internet connection by Cable / Fibre Optic) - the 050 number is the only number they have.)


On the other hand, there are some advantages of the 050 number:

- Calling number display at no extra charge (typically)
- Lower call cost for some long-distance callers.
- With the same ISP, you can use the 050 number if you move within Japan.


For IP Phone providers, it's in their interests for consumers to tell others their 050 numbers. In addition to making it harder for the consumer to leave (the 050 number will be lost), the IP Phone provider receives interconnection fees (quite possibly, on a timed basis) for calls from the PSTN or non-free affiliates. IP Phone providers often promote the idea of incoming calls to your 050xxxxxxxx number, such as a mention in the email newsletter about the above benefits, or awarding points for incoming calls. In one recent campaign, an IP Phone provider was giving away cash prizes to the subscribers that racked up the most number of chargeable incoming 050xxxxxxxx number minutes.



"Private ENUM" used with Softbank's BBphone

Softbank's "BBphone" service, unlike the other "IP Phone" services, includes what is effecitvely "private ENUM". If the PSTN number you've dialed corresponds to the phone line of a Softbank "Yahoo!BB" ADSL subscriber with "BBphone", and their ATA is switched on and connected, Softbank will automatically connect the call for free to the subscriber's ATA. This means that you do not need to know the person's 050 number - Softbank will, effectively, automatically lookup the 050 number and re-route the call for you.

When making an outgoing call via "BBphone", if you hear "pip pip pip" just before the ringing tone, it means that the call is going via "BBphone" (not via the PSTN), and if you hear "pip pip pip, pip pip pip" (ie, 3 pips, twice) just before the ringing tone, it means that the call is going via "BBphone", and is a free call (ie, you're calling another BBphone subscriber, either by 050 number or PSTN number), so there is a means for you to check, after you've dialed the number, if your call is going to be free. However, there appears to be no way of checking this without actually dialing the number and listening for the pips.

Described here:
Ip telephone invention




So to sum up, residential VoIP in Japan is a bundled "IP Phone" service from your ISP, using an ISP-provided or ISP-approved ATA.

On the plus side, this makes things extremely simple for consumers - the arrangement is seemless, and incoming and outgoing calls are made in the same way, from their normal analog phone. (It will even work with grandma's rotary dial phone.) Callers don't even need to know that they're using VoIP, as the ATA automatically routes eligible calls via VoIP (in some cases, even cancelling-out carrier override codes). Ineligible calls ( emergency calls to 119 / 110, and free calls ( 0120 xxx xxx, etc )) are automatically sent via the PSTN. Callers can manually specify the PSTN by dialing 0000 in front of the number.

On the minus side, things are set up to generate revenue for the VoIP providers, by locking subscribers in, and controlling incoming calls and charging termination fees.


It seems like such a waste, when you have a permanent, high speed Internet connection, with unlimited downloads and uploads, and an ATA sitting on your public IP address, ready and waiting to receive calls, but due to VoIP provider restrictions (blocking incoming SIP calls from non-business partner external domains), relatively few people can call you for free, but instead have to pay interconnection charges to go through your provider.



Summing up the three different types of ATA hardware used:


1. Softbank's "BBPhone" ATA, using MGCP (not SIP). Earlier types were a separate ATA with built-in bridge, that would connect to the ADSL modem. Most types these days are combined with the ADSL modem, which has router / bridge mode. (In the case of bridge mode, the ATA part has it's own separate public IP address).


2. Locked SIP ATAs (mostly Cable Internet providers reselling a white label VoIP service). Typically, the ATA includes a bridge, and so goes between the cable modem and the customer's PC / router. The ATA has it's own public IP address. The ATA uses the SIP protocol, but it's locked.


3. "Unlocked" SIP ATAs for use with ISP's accessed via NTT "Flets" ADSL / Fibre. The ISP supplies a list of "approved" NTT-branded ATAs (which are not locked (they can be set to use any SIP server) but are specifically designed for the "IP Phone" service (eg dial plan, etc is hard coded)), which the subscriber can either rent or purchase from NTT.

This is the type of ATA written about in this post:
http://forum.voxalot.com/voxalot-sup...t-dial-ip.html
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Unread 02-02-2008, 03:00 PM   #2
v164
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Default using these ATAs

Consider how VoIP users could use ENUM / peering with the three different types of ATAs -


Quote:
Originally Posted by v164 View Post
Summing up the three different types of ATA hardware used:

1. Softbank's "BBPhone" ATA, using MGCP (not SIP). Earlier types were a separate ATA with built-in bridge, that would connect to the ADSL modem. Most types these days are combined with the ADSL modem, which has router / bridge mode. (In the case of bridge mode, the ATA part has it's own separate public IP address).
As I explained here,
Eikaiwa--Discussing Japan in English: VOIP, SIP, MGCP

Quote:
Between the MGCP protocol, and their own VoIP adapters... the "BB Phone" service is locked down.

Outside of reverse engineering and emulation measures, the only option would be for the subscriber to...

Quote:
... buy an unlocked SIP-compliant analog telephone adapter (ATA), and put that between your analog telephone and your (locked) Yahoo BB modem.



Quote:
Originally Posted by v164 View Post
2. Locked SIP ATAs (mostly Cable Internet providers reselling a white label VoIP service). Typically, the ATA includes a bridge, and so goes between the cable modem and the customer's PC / router. The ATA has it's own public IP address. The ATA uses the SIP protocol, but it's locked.
These ATAs accept incoming SIP calls direct from anywhere on the Internet, so if you can determine a SIP URI that the ATA will answer to (typically 050xxxxxxxx@ipaddress), incoming ENUM can be enabled by simply mapping the PSTN and/or 050 number to that SIP URI at e164.org (and watching out for IP address changes). The ATA is typically a bridge (not a router), and hence has a separate public IP address (it may be necessary to capture packets between the ATA and modem find the ATA's IP address).

Incoming ENUM can be set up conveniently, with no changes to hardare or equipment settings. For outgoing ENUM calls, I suppose you could have a transparent proxy intercept the outgoing SIP INVITE packet and re-route if there's an ENUM match, but short of doing that, the purchase of an unlocked ATA would be in order.


(It should also be noted that, in most if not all of the above cases (1) and (2), the ATA is owned by the ISP).


Quote:
Originally Posted by v164 View Post
3. "Unlocked" SIP ATAs for use with ISP's accessed via NTT "Flets" ADSL / Fibre. The ISP supplies a list of "approved" NTT-branded ATAs (which are not locked (they can be set to use any SIP server) but are specifically designed for the "IP Phone" service (eg dial plan, etc is hard coded)), which the subscriber can either rent or purchase from NTT.

This is the type of ATA written about in this post:
http://forum.voxalot.com/voxalot-sup...t-dial-ip.html
These ATAs offer the greatest promise for facilitating ENUM and VoIP peering, because they are SIP "RFC 3261" compliant, can be set to use any SIP provider (NTT keeps on pushing the fact that with their "Flets" service you have a choice of providers), and the ATA is not provided by the VoIP provider, but is procured separately (from NTT or elsewhere (Internet auction, etc)), so the subscriber is free to configure that ATA however they wish.

These are the ATAs of which I wrote,
Quote:
Originally Posted by v164
The SVIII modem is representative of the most widely used unlocked SIP ATA in Japan - those provided for use with the bundled "IP Phone" services provided by Japanese ISPs using NTT-West / NTT-East "Flets" ADSL connection. I intend to write in more detail about this at a later date, because supporting these SIP devices would be, at this point in time, the most effective way to drive adoption of ENUM and VoIP peering among Japanese VoIP users.
http://forum.voxalot.com/sip-broker-...necessary.html


All that's needed to use ENUM and peering, incoming and outgoing, in this situation, is a suitable SIP server for these ATAs to use. A SIP server like VoXaLot would be suitable, except for three issues that would need to be resolved:


(1)
Enable sending a "380 Alternative Service" message back to the ATA for calls that should go via the PSTN, as described here:
http://forum.voxalot.com/voxalot-sup...html#post10348


(2)
Filter out the "=on" part of the ";lr" parameter, to prevent the problem described here:
http://forum.voxalot.com/voxalot-sup...dial-ip-2.html


(3)
For incoming / outgoing calls that don't match ENUM: In some cases, the ISP blocks SIP messages to their server from outside their network, so INVITE and REGISTER messages from VoXaLot on behalf of the subscriber would be blocked. This problem would have to be resolved, perhaps by:
- sending a "redirect" message back to the ATA, to get the to ATA send the INVITE or REGISTER message to the ISP direct (ala sipbroker.com)
- establishing a SIP proxy for VoXaLot within the ISP's address space
- finding alternative providers for outgoing (and/or incoming) PSTN calls
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Unread 02-02-2008, 03:05 PM   #3
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Default ENUM +813xxxxxxxx -> tel:050xxxxxxxx

Quote:
Originally Posted by v164 View Post
Although calls between subscribers with the same provider are free, getting these calls for free is actually easier said than done. The reason is because in order to call the person for free, you need to know their 050 phone number (except in the case of Softbank's "BBphone" service - see below). If you have the PSTN number of the person you want to call, there is a reasonable chance that the phone you're calling is connected to an ATA that's subscribed to the same "IP Phone" service as you (or one of the free affiliates), however, because you have no way of knowing this, and no way of knowing the 050 number, it is impossible for you make the call for free.
Perhaps the "tel:" URI scheme could be used in ENUM for this?

In a similar way to NGN lookups for +6113xxxx numbers to their PSTN number equivalents, e164.org could map Japanese PSTN numbers (+813xxxxxxxx) to the equivalent 050 number.

This would bring about some of the benefits of ENUM, without any additional hardware needed.
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