SIP Commander. Version 1.0
Download free version (1Mb)
SIP 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.
SIP 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 SIP 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 SIP 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, SIP 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.
SIP and MGCP/Megaco Comparison
SIP and MGCP Megaco comparison
> ABOUT SIP > SIP H323 AND MGCP > SIP AND MGCP MEGACOSIP and MGCP Megaco
SIP and MGCP/Megaco
MGCP/Megaco and SIP are not peers; they can and will coexist in converged networks. There are, however, a number of issues surrounding implementations that will influence future directions and capabilities.
As discussed in the Architecture section, MGCP/Megaco does not constitute a complete system: a session initiation protocol is required between gateway controllers. SIP is eminently suitable and is a requisite where there is more than one softswitch.
A more contentious area of discussion, is the use of MGCP/Megaco to control end-points. A media gateway could be an IP phone but, due to the service limitations this imposes, this is likely to be unpopular. MGCP/Megaco would only be able to support basic IN-type services in a dumb black phone.
For advanced services (i.e. anything more sophisticated than IN services), SIP is required to reside both in the endpoints and above the signalling network, acting as the service intelligence. The issue that then arises is where should services reside?
Softswitch vendors would prefer the service intelligence to reside in the IP Central Office, tied in to the softswitch architecture. This perspective holds firm in the short-term where emphasis on convergence means that the interconnect point between a circuit-switched environment and an IP network will be a major focus. In this scenario,
SIP application servers reside with the softswitches in the IP Central Office with MGCP/Megaco controlling multiple media gateways across the network, delivering services to all endpoints. As the legacy circuit-switched network diminishes in importance, and focus shifts squarely onto the IP infrastructure, then this model will become increasingly imbalanced and irrelevant. The softswitch function will need to evolve away from the interconnect point.
In a pure IP environment, service creation would be distributed across the network. This is the model that has produced the startling innovations that we have seen on the Internet: anyone with a few dollars and a good idea has the opportunity to give it a try. The Application Service Provider model can be extended to offer voice-type services. ASPs, ISPs, or even the end-users themselves can create their own SIP services; after all, SIP's similarity to other Internet protocols makes it a familiar programming language to web developers. A SIP-centric implementation would use MGCP/Megaco only for internally controlling an IP telephony gateway. SIP application servers would distribute services throughout the network via SIP proxy servers.
SIP MGCP/Megaco Peer-to-peer signalling protocol Can be used as a control protocol for delivering services across the network A session initiation protocol required between separate softswitches* Used for internally controlling an IP telephony gateway Client-server architecture Master-slave architecture "Pure" IP solution An interim solution for co-existing
networks - "PSTN over IP" Horizontal architecture that re-uses Internet elements Mirrors the signalling and control architecture of IN Intelligent clients Assumes dumb end-points Abstracts the signalling layer from the network Pre-supposes the existence of hardware "New world" approach Ð simple open and horizontal "Old world" Ð centralised, controlled and vertical * SIP and MGCP/Megaco are complementary in certain ways and mutually exclusive in others. |