This is a very interesting question to be answered before we start with further discussions being the fundamental question of this whole web site.

 Human mode of communication is the best compared to all the living creatures in the world. First we found new languages to communicate, and then we started thriving for communication over distances. The invention of telephony (Plain old telephone system, POTS) was such a break through for communicating over geographical boundaries. For more than 100 years it was the unshakeable technology. But as in every other field of technology, better ones always make the old one extinct.
VoIP is one such technology, which has born to challenge POTS in terms of technology as well as business. Business ??? You might ask me why business. Yes, even though POTS solves the problem of long distance communication, there is a cost associated with it. You cannot talk to your loved ones living across the globe with out counting the pennies. With VoIP it is free. You might now ask, “VoIP is free”; no I still pay bills to my VoIP provider like vonage, dialpad etc. etc., yes you are right and wrong. You are right because you end up paying only when you make a call to another POTS line from your VoIP line. In this case your call is carried over Internet as close as possible to the destination and then it is converted to POTS. That’s the reason why your phone bills are so less when it comes from VoIP provider. You are wrong because it is absolutely free to call from a VoIP line to another VoIP termination. The only cost associated is your internet bandwidth. For example look at Skype, Gizmo etc.,
In a perfect world where every corner is connected with internet then the cost for long distance communication is near zero. But it will take a lot more time to reach this situation as evolution takes time. Till then there is business to convert VoIP calls to POTS calls by carrying them over IP to as close to destination.

     Your VoIP service provider assigned you a DID number when you bought the service from them and left you in dark making you confused with what it is??? Let’s explore what DID is here….
     DID is an acronym for Direct Inward Dialing. It is a technology which came from traditional PBX world in to VoIP world.
     In traditional telephony PBX, DID is a concept invented to reuse limited number of physical POTS/PSTN connection with a wider range of phone numbers. Let’s take an example, assume an organization ABC with 10 employees and every employees number need to be advertised as unique to the external world. Now the company can approach their PSTN service provider to provide 10 DIDs, which are usually in a sequence. They would be allotted DIDs which look like XXXX00 to XXXX10, assuming the phone number for the particular location is 6 digits (This differs from country to countries). How does this work??? In this case the PSTN Central Office, as soon as it sees a call to a number starting with XXXX diverts the call to the ABC’s PBX. Now the PBX senses an incoming call and requests the central office to give the remaining 2 digits, in a process called ‘winking’. After collecting the last two digits, the PBX can take a decision of forwarding it to the right desk. Using this technology more than a phone number could be diverted using a single line, also the external caller need not know that this is an extension number and he thinks that it is a direct line to his friend’s desk. This is also very useful for call centers with huge number of employees and with a comparatively lesser number of telephone connections.
     Now coming to the VoIP world of DIDs. Usually VoIP service providers give a DID number when you take a VoIP connection, in order make life easier for your friend or business associate who is a PSTN user to dial up your voice over IP connection, thinking that he is dialing a PSTN line. Otherwise it will be very difficult for a PSTN user to dial you, as in the VoIP world you may be identified as yourname@something.com (How can a PSTN user dial this). The VoIP service providers buy these DIDs from their CLECs or some tier 1 companies like Level 3. There are also a number of websites on the internet where DID numbers are traded. When some one calls to this number, the traditional PSTN forwards the call to the VoIP service provider and he manages to transfer the call to you, since he knows that you are assigned this DID number from his database.
     This is also very desirable in some business scenarios where you live in California and have different DID numbers that are local UK or Singapore numbers and let your VoIP provider manage diverting calls to these numbers to your single VoIP connection to California. This helps reaching your customers at UK or Singapore easily since they assume that they are making a local call in their country (and will be charged for a local call). This is also very useful for call center scenarios, where you can buy a bulk of DIDs of different countries and distribute to your clients there and get all the calls diverted to your call center.

I am not sure how many of you know/remember phil zimmerman. He is the same old guy who invented PGP ( Pretty Good Privacy), which we use in every day of our communications knowingly or unknowingly. He was then arrested by the US government and was trialed for 5 years.
He is back now and with a good news for VoIP users. He has come up with a software suite called zphone, which is encryption for your voice data on voip calls. Right now this works only between PC to PC. In future if this becomes another break through like PGP and all network element providers like Linksys, etc., start implementing them on different network elements like switches, ATAs etc., then this can also reach your analog telephone.
You can see more details about zphone at
http://www.philzimmermann.com/EN/zfone/index-start.html

I was able to understand the setup after a few emails between the poster of the question.

The setup is something like, a user has SIP express router as his registrar and has another Xlite softphone registering to the same registrar. Every thing works fine when calls are placed from the Xlite softphone to the SNOM station through registrar. But when the softphone directly places call to SNOM phone with the user name and the IP, SNOM returns a SIP error message ‘403 Use proxy’.

Solution:

SNOM phones have a default property where it could accept SIP INVITE messages only from its registrar. This is supposed as a security measure, but which is not found in other SIP phones, which led to the confusion.

This value can be tweaked by setting “Filter Packets from Registrar” to OFF, this is detailed in the following SNOM wiki,

http://wiki.snom.com/Web_Interface/Advanced/Advanced_Network#Filter_Packets

This solved all the problem. But it is advisable to give a thought on, is turning this setting OFF is a real necessity.

Question from BOB:

In the same way that an ISP can record every website you visit (if you don’t take steps to prevent them), under current U.S. laws, can VOIP providers record conversations (not just phone #s) with OUT some warrant or court order?

In other words, can the VOIP provider legally record your voice conversation as a standard business practice, if they choose to do so?

I asked this of AT&T’s customer service, but they could give me nothing in writing that addresses this one way or another.

Voipguru answer:

First about the technical possibilities to record a VOIP call,

It is very easy to learn about some ones conversation over VOIP if the RTP streams are not encrypted. You can check this yourself easily by capturing the raw RTP packets using Wireshark and then using the menu item “Statistics->Voip Calls” and select the call you are interested and press the ‘player’ button. Voila there you could hear the whole conversation. Believe me many VOIP implementers were shocked to learn this, when they saw this first time.

Next coming to the law,

Law with respect to this is also confusing as with any other law. But my understanding is that if the government decides to tap the conversation of any VOIP user then they will go about with it with out any notification. All VOIP providers are mandated to keep a provision to tap the voice when required by the intelligence agencies. They need to provide this facility even if they give a secure VOIP communication solution to their customers.

Here is a short extract from wikipedia on the lawful interception with VOIP

“During the 1990s, to help law enforcement and the FBI more effectively carry out wiretap operations, especially in view of the emerging digital voice and wireless networks at the time, the US Congress passed CALEA in 1994 [1]. This act provides broad guidelines to network operators on how to assist the LEAs in setting up interceptions and the types of data to be delivered. CALEA does not, as many believe, provide specific implementation directives on interception. More recently, the US Federal Communications Commission (FCC) mandated that CALEA be extended to include interception of publicly-available broadband networks and Voice over IP services that are interconnected to the Public Switched Telephone Network (PSTN).”

Here is the link for CALEA (Communications Assistance for Law Enforcement Act) website if you want to learn more about it,

www.askcalea.com

http://en.wikipedia.org/wiki/Communications_Assistance_for_Law_Enforcement_Act