Addressing Issues

SIP procedures are a standardized way to setup sessions (mostly phone calls) over the Internet.
In the world of Plain Old Telephone Service (POTS), each phone is addressed using a digit string, e.g. a phone number, internationally agreed according to ITU E.164. Users are used to dial digits to setup a phone call. SIP uses an email like format to address SIP endpoints (e.g. soft client or desktop phones).


SIP uses usually SIP URIs in the format of 'sip:This email address is being protected from spambots. You need JavaScript enabled to view it.'. As we like to make phone calls from SIP phones to POTS and vice versa, we need to find a way to map those identities.

Providers with own PSTN gateways and numbers

Those providers often hide the SIP addressing to end users. They assign E.164 phone numbers out of the PSTN numbering range to SIP subscribers. Sometimes the E.164 number is used as the userpart of the SIP ID, e.g. This email address is being protected from spambots. You need JavaScript enabled to view it.. But often the POTS phone number is kept separate from the SIP ID. The advantage is, that the POTS phone number can be changed independent from the SIP ID, or can be even optional, e.g. SIP service can be used for outgoing calls to PSTN only, no phone number for incoming calls assigned.

Traditional POTS providers who add SIP access to their portfolio often even do not allow incoming calls from other SIP providers, incoming calls must come from the PSTN and go via the providers gateway. In this case the fact that SIP access is used, gives no added benefit to the end user. The SIP phone appears as a normal phone.
The reason for this is that there is a termination fee to be paid to the operator for each call coming in on the legacy PSTN interfaces. Originators from the Internet will not pay for termination, SIP to SIP is usually free.

How does SIP addressing work technically in this case? For those POTS to SIP replacements, usually IADs (Integrated Access Devices, often a DSL modem with interface for an analog phone) are used. The SIP access appears to the end user as a normal phone. Dialing a phone number, the SIP setup message (INVITE) from the IAD to the providers server will use the dialed number in the user part of the To: SIP URI, e.g. sip:This email address is being protected from spambots. You need JavaScript enabled to view it.. The provider's server extracts the user part and uses the phone number for call routing.
For incoming calls, the destination E.164 POTS number will be translated into a SIP URI and setup to the SIP phone takes place, SIP addressing totally invisible for the users.

If incoming SIP calls (over the Internet) are allowed besides the path over PSTN, the user would be reachable free of charge for the caller.

SIP Providers without PSTN

It is quite easy to become a SIP provider. There is free open source SW available, renting a server is not expensive. The costly part is providing a PSTN gateway and PSTN numbers.

This is why there are numerous SIP VoIP providers, VoIP only, offering their services. Registration is for free, as SIP to SIP calls are. Calls to PSTN are often possible, using a prepaid account. For those calls, calls are forwarded to wholesale VoIP providers, specialized in gateway services to the PSTN. Incoming calls from PSTN are not so easily possible, because getting a permanent PSTN number is not for free.

As the focus is on SIP to SIP calls, full reachability over the Internet is given. The SIP URI is not only used for device registration and routing calls from the server to the device, but also for incoming calls from other networks / SIP providers. E.g. the SIP URI replaces the phone number.

Still, many users like to use desktop type phones with their SIP accounts, instead of PCs with soft clients installed. Desktop phones do not so easily allow to enter full SIP URIs. This is why even the providers focusing on SIP services use numbers as user part of SIP URIs. E.g. 'sip:This email address is being protected from spambots. You need JavaScript enabled to view it.'. Instead of using the full SIP URI, a caller can enter just the user part of the URI, e.g. the number, on the numeric keypad. The SIP phone will convert this into a SIP URI in the form shown above.

Those number used in the SIP URI's user part have nothing to do with publicly reachable PSTN phone numbers! They are just valid and unique inside a provider's network.
Calling from sip:This email address is being protected from spambots. You need JavaScript enabled to view it. to This email address is being protected from spambots. You need JavaScript enabled to view it. is easily possible by just typing 2202 on the keypad. But how to reach 'sip:This email address is being protected from spambots. You need JavaScript enabled to view it.'?

Of course using the full SIP URI would make any SIP to SIP call possible. But how can we reach provider-b with a keypad only phone?

Similar to area codes in the PSTN, SIP providers have introduced prefixes as breakout codes to other SIP networks. The difference is that the dialed number including prefix is not used end-to-end, but translated into a valid SIP URI of the destination network.

Let's say provider-a has introduced *234 as prefix to reach provider-b:
A subscriber of provider-a dials *2342200331.
The SIP phone converts this into 'sip:*This email address is being protected from spambots. You need JavaScript enabled to view it.'.
The server of provider-a analyses the user part and convert the SIP URI into 'sip:This email address is being protected from spambots. You need JavaScript enabled to view it.'.

The problem is that those numbers and prefixes are not well coordinated. The numbers are in no way standardized as PSTN E.164 numbers are. Provider-a might chose *234 to forward SIP calls to provider-b, provider-x might chose 0005 as prefix for the same destination.

Bridging The Gap From PSTN Numbers to VoIP: ENUM

For POTS PSTN subscribers there is no way to dial SIP URIs. Furthermore, every PSTN susbcriber is part of the globally agreed E.164 numbering plan and all dialed numbers must fit into this plan.

ENUM is a system to route a call dialed as E.164 number using SIP and SIP URIs. ENUM is integrated into the DNS. There is a special public domain reserved to map all E.164 number to SIP URIs. It's ''. This mapping does not work to SIP subscribers not having any POTS PSTN number assigned, E.164 numbers are the key to access ENUM's SIP records!

Why would we need to map from E.164 to SIP, if the subscriber is reachable over his E.164 number anyway? The answer is, to optimize the routing. If the caller is using an E.164 number to call, the originating network does not know anything about a potential SIP reachability of the called party. If ENUM query returns a SIP address, network A can route the call to network B via SIP much cheaper.

Usually the originating network has to pay a termination fee to the terminating network. ENUM is in the interest of the calling side to route calls cheaper. If a network allows incoming SIP calls from the public Internet, no termination fees can be charged. Therefore we can not expect that existing commercial PSTN operators allow uncontrolled terminating calls using ENUM. Nevertheless, private ENUM solutions with mutual operator agreements may exist.

Proprietary ENUM Domains

ENUM is not only useful for PSTN to VoIP, but also VoIP to VoIP. If SIP subscribers dial E.164 numbers, the originating SIP network can check, if the called party is SIP reachable. Often such calls are free of charge then.

Lets say someone has a normal phone line with a well known number. Everyone calling him has to pay the usual rates. Now this person registers to a free SIP service (no PSTN number) using his DSL. By adding his phone number to ENUM and linking it with his new SIP phone, he gives callers the chance to reach him for free, using the same phone number!
This is the case if the caller is calling from SIP provider which uses a PSTN gateway for PSTN numbers, but checks ENUM before this and uses SIP free of charge, if the ENUM query returns a SIP address.

Adding an own number to the domain has been difficult in the past. Traditional PSTN operators as "owners" of E.164 numbers have no interest to give subscribers control over routing to their phone number.

Because of this, alternative ENUM domains have been introduced. They allow easy entry of phone numbers and let end users control their records.

Most popular are:

Originating networks would check first and other proprietary domains sequentially in a predefined order, until routing info is found.

This allows bypassing the traditional PSTN, using well known PSTN phone numbers, without access to the domain.

SIP Broker

Another effort to promote interconnections of SIP networks and even allow calls from PSTN to SIP-only networks is undertaken by SIP Broker.

SIP Broker is a central place to translate prefixes into SIP networks. If the own SIP provider does not support prefix dialing to the desired destination network, SIP Broker can be used. Some SIP devices, or providers, allow to add dialing rules. E.g. if the dialed number starts with '*', convert the SIP URI into Some providers use SIP Broker by default.

SIP Broker is sponsored by Voxalot, an Australian VoIP service provider. SIP Broker has also many PSTN gateways for calls from PSTN to any (numerical) SIP address which is in SIP Broker's prefix directory. Just call a local access number and make phone call to any SIP subscriber world wide!

Tired Of The National Number Monopoly? iNUM!

There are numerous small SIP service providers, with no gateway to the PSTN and no own PSTN number range. They would like to have their subscriber be reachable from the traditional PSTN as well!

E.164 PSTN numbers are traditionally tied to physical phone lines (or mobile phone subscriptions). Furthermore, traditional E.164 numbers are mostly geographical numbers, while SIP UAs can be located anyplace with IP access. Also national numbering space is sometimes tight.

Why not reserve a special number range just for VoIP? Those numbers would be just 'virtual', no phone line behind it, just linking to SIP URIs.
In some countries this has been done already, some area codes have been reserved for VoIP usage.

The ITU has assigned some country codes as "Shared Country Codes". Those are used for example for satellite services. The country code are +881..+883.

Voxbone, a Brussels based global resell VoIP operator, started an iniative to use a shared country code for global access to VoIP networks. The so called iNum service should act as gateway from PSTN to VoIP. As +49 is routed to Germany and +66 to Thailand, should +883 be the gateway to VoIP, no matter where the UA is located. Voxbone got the +883 5100 segment assigned by the ITU. Many of their customers got a sub range assigned and can receive incoming VoIP calls from callers dialing the iNum.
Voxalot, an Australia based VoIP provider, offers Voxbone's iNum to its subscribers. +883 5100 04 xxxxxx, where xxxxxx is the six digit Voxalot member ID and user part of the SIP URI, can be used to reach the Voxalot VoIP subscribers. Voxalot does not have their own PSTN gateway.


Internet bandwidth is cheap. Much cheaper than TDM transmission used by traditional PSTN operators. VoIP call control is much cheaper than traditional PSTN switches.

We see prices for communication falling already. Even more would be possible. The key is VoIP as independent 3rd party service and interworking with legacy networks. PSTN operators also "own" the numbering space. There are technical solutions like ENUM to make the transition to VoIP easy. Number portability is a first step to hand over number ownership from the operator to the subscriber. An obligation to do ENUM queries and forward calls to the SIP domain, as well as give subscribers control over their ENUM NAPTR records would be the next.

Naturally the established operators do not have much interest in this.


0 #1 dev 2016-01-27 13:44
One Contact number ( is a universal 12 digit phone number to reach VOIP users across all SIP providers. Way it works is, whenever any user dials a 12 digit number (from SIP phone dialer) SIP proxy, used by the user, checks for the registered 12 digit user id and process the calls normally. If there is no user registered corresponding to the number dialed, SIP proxy makes call to ( to get Proxy/UA for the number. SIP proxy of the original caller then forwards the request to the proxy obtained from

12 digit One Contact number is owned by the users (not SIP Service providers), so users have flexibility to change SIP Service providers without loosing their contact number.

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