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  IPsmarx VoIP Softswitch Solution

  Features

Built-in Billing

The IPsmarx VoIP Softswitch Solution uses an internal RADIUS server for AAA (Accounting, Authorization, and Authentication) procedures.  This internal AAA RADIUS module guarantees the highest Softswitch performance and outstanding billing functionality.

SIP Proxy - Session Border Controller

IPsmarx has a SIP proxy server which handles all of the signaling for SIP devices.  Based on the proxy settings the server will choose what carrier the SIP device will use.  It will also hide the IP address of the sender from the IP address of the receiver providing extra security to the sender. IPsmarx Session Border Controller act as a complete proxy for both Media and Signaling Packets and includes the following features:

• SIP and H323 Compatible 
• Support protocol conversion (SIP to H323 and H323 to SIP)
• Support all available codecs: G729,G723, G711 a, G711 u , H264, H264 (Pass-thru Mode)
• Support all DTMF Mode (In-band, Out-band, RFC2833)
• User Defined Caller-ID
• SIP Port modification

SIP Registrar Server

This server is responsible for authentication and registration of SIP clients.  The clients come with SIP username and SIP password and will be authorized by the server to make a call if they have funds available in their account. This SIP server keeps track of Local / Public IP-Addresses therefore there is no need for a STUN server.

SIP AAA

SIP AAA (Authentication, Authorization and Accounting) is an essential part of the IPsmarx Softswitch which keeps track of each SIP client and their services (i.e. Voicemail, Video Over IP, …), account management (i.e. surcharges that may apply), and to keep records of the cost of the calls made.

Multiple Level of Access

This feature allows multiple users to be logged in at the same time with different levels of access.  This enables providers to set user level access to the system so a customer agent can only see their CDR (Call Detail Record) or a supervisor can create customer agents but cannot change rates or dialing plans.  All of these permissions can be customizable to suit the needs of the agent.

Low-Balance Automatic Email Notification

For each Pre-paid wholesale or pre-paid SIP customer a trigger limit can be created to inform the client when their balance goes below a certain amount.  This alert is sent out via email automatically and the message can be customized.

Unlimited Number of SIP Accounts

There is no limitation for the number of SIP clients that can be created in the system.  This allows all IPsmarx customers to grow exponentially as they add more customers to their Softswitch.

Unlimited Carrier Support - Termination

There can be unlimited number of end-point carriers that are defined in the system.  Customers can have multiple carriers and take advantage of the best prices/quality from each of their providers. Customers can also use Fail-Over feature.

Flexible Dialing Plan (Hunt-Stop, Fail-Over Features)

This flexible dialing plan allows fail-over carrier redundancy, so if a call fails to connect, the system will try again with another carrier.  Additionally, the Hunt-Stop Feature will stop the call from trying additional carriers once a defined carrier has been tried.  For example, if a customer has 3 carriers, but one carrier’s buying rates are more expensive than the selling price for certain destinations, the IPsmarx system can be configured to send calls to the first 2 carriers and then stop to avoid losing funds.

NAT Client Support

IPsmarx Softswitch handles both media and signaling packets.  This means that all customers, including clients behind the NAT, will be able to send calls through the Softswitch even if there are any changes in the local/public IP addresses of the media and signaling packets.

Unlimited Rate Table

Full Feature Rate Table

System owners can establish different packages with customizable rates.  For example, with a 30 by 6 resolution, the caller will be charged for the first 30 seconds regardless of the talk time.  In this scenario, after the first 30 seconds, the caller will be charged in 6 second increments.

Flexible CDR (Call Detail Record) Report

is an enhanced reporting tool which enables providers to easily manage their business.  The CDR report allows users to view call details, cost, profit and duration of each call.

Flexible ASR (Average Success Rate) Report

The ASR report is very useful when evaluating the quality of termination providers.  The report measures call duration based on carriers, destination, and more.  The report then produces a percentage of the calls being terminated by the system which can help providers determine how effectively end-point carriers are terminating their calls.

Flexible ACD (Average Call Duration) Report

The ACD report measures call duration based on carriers or destinations.  This produces a report indicating how long customers are on the phone for then produces a time value of how well the end-point carriers are performing in terminating the calls.

User Breakdown Report

Produces reports based on individual users’ status, usage, balance and other call details.

Call Monitor (On-line/Live Call Status)

Allows providers to view live calls going through the platform.  IPsmarx’ monitoring tool displays who is making the call and to what destination, the codec being used and any other information relevant to the call.

Security Logs

Produces a Java applet that shows the logs for all unauthorized attempts on the platform.  Security features of the Softswitch will reject any unauthorized login attempt; however a log will be available for system administrators to check the IP address of the unauthorized source.

SIP & H323 Carrier Support

Supports both SIP and H323 protocols for end-point carriers.  The Softswitch will automatically assign the correct port for each protocol.  This can also be changed manually to any port requested by the end point carrier.

Instant H323 & SIP Protocol Conversion

The VoIP Softswitch Solution views the incoming protocol being used and looks at the outgoing protocol being requested and then changes the protocols required without any disruption of call quality.  This feature allows the system to link calls between VoIP providers using incompatible protocols.