Phone Commander. Version 1.0
Download free version (1Mb)
Phone Commander is a free portable softphone that you can use to make and receive VoIP phone calls from your PC, iPhone or Android based smartphone.
Phone Commander like a telephone to let you make calls through your computer. Call anyone via the internet who also has a softphone installed and if you sign up with a VoIP gateway service company you call regular telephone numbers as well.
The advantage of using Phone Commander is that you can leverage low cost or free VoIP calls and you can connect to the company SIP PBX and work remotely.
Easy to use and powerful Windows program was designed for people who want to improve dynamic interactions with contacts and to manage contact information. It supports multiple SIP accounts and save calls history.
See all contact information for incoming calls. For example, you can see Firstname, Lastname, Birhdate, Company name and other information before call answering. Keep detailed records and notes in your call log about each call and contact.
You can use Phone Commander as a customer database, or personal contact address/phone book, working with groups for managing contact info of individuals and organizations with relationships. Phone numbers, emails, web pages, faxes, pagers, addresses, customer notes - you can save all this data in an organized format.
You can dial phone numbers in one click via Internet. Program has a simple intuitive interface and quick and easy contact lookup. Import and export features are also available.
Attractive and easy-to-use organizer & PIM will keep track of your contacts, addresses, distribution lists, manage your schedule, remind about appointments, and keep your daily notes in order, store your contacts and call by one click. The slick user interface makes it a snap to find addresses and phone numbers, enter reminders.
With its intuitive & familiar interface, users can seamlessly transition from a traditional hard phone environment into the world of Voice over IP. Also by making the navigation simple and user friendly, Phone Commander provides users with easy access to address book management.
The program detects DTMF user input, sends DTMF (Inbound, SIP INFO, RFC2833), supports G.721 A-law/Mu-law, GSM.610, Speex. You can select audio In/Out Devices.
What Is SIP? - Characteristics
What Is SIP Characteristics Introduction About SIP Overview Basics
> ABOUT SIP > WHAT IS SIP > CHARACTERISTICSCharacteristics
Characteristics
SIP is described as a control protocol for creating, modifying and terminating sessions with one or more participants. These sessions include Internet multimedia conferences, Internet (or any IP Network) telephone calls and multimedia distribution. Members in a session can communicate via multicast or via a mesh of unicast relations, or via a combination of these. SIP supports session descriptions that allow participants to agree on a set of compatible media types. It also supports user mobility by proxying and redirecting requests to the user's current location. SIP is not tied to any particular conference control protocol. In essence, SIP has to provide or enable the following functions:
Name translation and user location
Ensuring that the call reaches the called party wherever they are located. Carrying out any mapping of descriptive information to location information. Ensuring that details of the nature of the call (Session) are supported.
Feature negotiation
This allows the group involved in a call (this may be a multi-party call) to agree on the features supported - recognizing that not all the parties can support the same level of features. For example, video may or may not be supported.
Call participant management
During a call a participant can bring other users onto the call or cancel connections to other users. In addition, users could be transferred or placed on hold.
Call feature changes
A user should be able to change the call characteristics during the course of the call. For example, a call may have been set up as 'voice-only', but in the course of the call, the users may need to enable a video function. A third party joining a call may require different features to be enabled in order to participate in the call.
SIP fulfils these functions and re-uses other web elements to make it flexible and scalable.
Rather than defining a new type of addressing system, SIP addresses users by an email-like address. Each user is identified through a hierarchical URL that is built around elements such as a user's phone number or host name (for example, sip:user@company.com). This means that it is just as easy to redirect someone to another phone as it is to redirect someone to a webpage.
SIP uses MIME, the de facto standard for describing content on the Internet, to convey information about the protocol used to describe the session. As a result, SIP messages can contain Java applets, images, audio files, authorization tokens or billing data.
SIP borrows from the email model, using the Domain Name System to deliver requests to the server that can appropriately handle them. This simplifies the integration of voice and email. Servers along the call path can easily create and forward email messages, and vice versa, enabling various combined services.
SIP provides its own reliability mechanism and is therefore independent of the packet layer and only requires an unreliable datagram service. SIP is typically used over UDP or TCP.
SIP provides the necessary protocol mechanisms so that end systems and proxy servers can provide services: