SIP CONTD.

I forgot to mention in the last post, the cost of outgoing number is ($36 /Year).

Quotes from Wikipedia :

SIP reuses most of the header fields, encoding rules and status codes of HTTP, providing a readable text-based format. SIP clients typically use TCP or UDP on port numbers 5060 and/or 5061 to connect to SIP servers and other SIP endpoints.


Network Elements:

A SIP user agent (UA) is a logical network end-point used to create or receive SIP messages and thereby manage a SIP session. A SIP UA can perform the role of a User Agent Client (UAC), which sends SIP requests, and the User Agent Server (UAS), which receives the requests and returns a SIP response. These roles of UAC and UAS only last for the duration of a SIP transaction.

A SIP phone is a SIP user agent that provides the traditional call functions of a telephone, such as dial, answer, reject, hold/unhold, and call transfer. SIP phones may be implemented by dedicated hardware controlled by the phone application directly or through an embedded operating system (hardware SIP phone) or as a softphone, a software application that is installed on a personal computer or a mobile device, e.g., a personal digital assistant (PDA) or cell phone with IP connectivity. As vendors increasingly implement SIP as a standard telephony platform, often driven by 4G efforts, the distinction between hardware-based and software-based SIP phones is being blurred and SIP elements are implemented in the basic firmware functions of many IP-capable devices. Examples are devices from Nokia and Research in Motion.

Each resource of a SIP network, such as a User Agent or a voicemail box, is identified by a Uniform Resource Identifier (URI), based on the general standard syntax^[9] also used in Web services and e-mail. A typical SIP URI is of the form: sip:username:password@host:port. The URI scheme used for SIP is sip:. If secure transmission is required, the scheme sips: is used and SIP messages must be transported over Transport Layer Security (TLS).

Last paragraph confirms that I will need a SIP device at the user end. Maybe the ATA was the UAC here. Now, the problem is how do I route the numbers to existing PSTN.

Lets start all over again.