New SIP/RTP VOIP Server software now available - Solves all NAT Traversal Problems.

For: SIP/RTP/VOIP related developers and deployers.

LanScape

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has recently released the latest version of the LanScape Centrex Proxy Server (SIP Proxy) and the VOIP Media Proxy (RTP Proxy) server.

If you have to deploy your own VOIP domain, using our server software will make your task an easy one. By using our scalable SIP and RTP media proxy servers, you will be able to overcome all NAT related issues for your VOIP domain in addition to having future scalability as your VOIP network grows. No longer do you need expensive session border controllers to deploy a properly working SIP/RTP VOIP network. You can achieve all of this functionality using any version of Microsoft Windows.

The VOIP servers come in three product grades: Personal, Professional and Enterprise. We also offer trial versions of the server software so you are able to perform a proper evaluation before you make a purchase. For trial software availability, please see our "Voip Test Drive" web page located at:

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Here is a brief feature list for the LanScape Centrex Proxy Server (SIP Proxy):

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Designed for the Microsoft Windows=AE family of operating systems - Versions 9x/Me/NT/2000/XP/2003

Available in three product configurations: Personal, Professional and Enterprise.

Realtime event logging to a log file and to the GUI.

Full control over the network configuration (IP address and SIP port, domain name, Max SIP message length, Router/Gateway IP address, etc).

Denial of Service (DOS) attack protection built in.

Configurable call processing timouts.

Fully integrated registrar capability. Use the internal proprietary registrar database support for simplicity and ease of use. Alternatively, use an external registrar database (Microsoft Access, SQL, MySql, etc). If an external registrar database is used, multiple proxies can share the database for large VOIP deployments. Registrar presence events using SUBCRIBE/NOTIFY are supported.

Configure static registrar entries. Acts as an "always available" local directory.

Perform complex sequential call routing tasks. Route calls to any location anywhere and in any order. Call routing supports regular expression syntax if desired. Route calls to other SIP endpoints without having the endpoints register. Perfect for routing calls to PSTN gateways, Asterisk PBX or any other VOIP element.

Supports Global iNet=AE user accounts. LanScape's Global iNet=AE user accounts will allow you to publish your VOIP call endpoints to our global VOIP directory service. Anyone in the world can search the Global iNet=AE system for your business or personal contact information and call you via your VOIP domain.

Deploy with one or more LanScape VOIP Media Proxy=99 servers to obtain a complete session and media handling solution not previously available. Media proxying is very important in overcoming peer to peer media issues introduced by NAT network elements. When using media proxying, call media is load shared with all available media proxies. Highly recommended. LanScape media proxy solutions also solve the dreaded "media deadlock" problems other products face.

Fully automatic and configurable WAN IP address monitoring and detection when deploying the Centrex Proxy Server=99 inside of your private network. Includes support to allow you to publish the WAN IP address, private IP address, proxy server SIP port and domain name to any web server using HTTP POST. A perfect solution for those who use dynamic DHCP services to access their VOIP domains.

Supports a custom plug-in DLL. With this "user developed" custom plug-in DLL and API, you have the ability to alter all received and transmitted SIP protocol data. The custom plug-in DLL API will also allow you to be notified of special call related events (call start, call terminate, etc). There are many uses for this capability such as: SIP message filtering, SIP message logging, getting badly behaving SIP devices to interoperate properly. The list is endless. A complete example custom plug-in DLL with full C++ source code is included. Requires Microsoft Visual C++ 6.x or higher. The example plug-in that ships with the server allows you to echo all SIP traffic to an external SIP message log server (also included).

Full event logging: Enable/disable logging error tone for errors. Log proxy errors, warnings, informational messages, registration activity, and all call related events. Events can be logged to the server's GUI and to a log file. Log all SIP message activity to a log file.

Full challenge handshake authentication using the MD5 algorithm. Extremely secure since no password information is ever sent over the network. Configure the authentication domain and nonce expire time. Define a single user name and password for your VOIP domain or multiple user names and passwords for each user. Enable/disable authentication as it applies to the following SIP message types: REGISTER, INVITE, BYE, SUBSCRIBE and NOTIFY.

Here is a brief feature list for the LanScape VOIP Media Proxy (RTP Proxy):

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Designed for the Microsoft Windows=AE family of operating systems - Versions 9x/Me/NT/2000/XP/2003

Available in three product configurations: Personal, Professional and Enterprise.

Can handle any RTP media type.

Realtime event logging to a log file and to the GUI.

Full control over media session time outs, media session evaluation interval, orphaned media sessions and SIP proxy command time outs.

Full control over the network configuration (IP address and Media ports used). Configurable to communicate with one or more SIP proxies (LanScape Centrex Proxy=99 Severs)

Full call and media event logging. Logging to the GUI and to a log file supported.

Can handle thousands of active media sessions per host machine. Actual maximum number of active media streams is only limited by license restrictions and underlying host hardware.

Full challenge handshake authentication using the MD5 algorithm. All communications between the LanScape VOIP Media=99 Proxy and LanScape Centrex Proxy=99 Severs are authenticated. Helpful to ensure no malicious hacker attacks.

Specifically designed to overcome the well known "media deadlock" problem other media proxy solutions face. With the LanScape VOIP Media=99 Proxy, your call media will flow uninterrupted within your domain and when traversing the domain boundary.

A simple intuitive easy to use interface.

It will not matter if you are a seasoned VOIP/SIP/RTP deployment specialist or a rookie. The VOIP server software we offer really simplifies the tasks associated with managing and deploying your own personal, professional or enterprise VOIP network.. To check out what our other LanScape VOIP products can do for your application development, please go to our web site at:

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Thanks again and happy VOIP-ing,

LanScape Sales Staff

Keywords: SIP, RTP, SIP PROXY, RTP PROXY, VOIP PROXY, MEDIA PROXY, NAT TRAVERSAL, SESSION BORDER CONTROLLER, SER, SIP EXPRESS ROUTER, VOIP, TELEPHONY, H323, SOFTPHONE, TRIAL SOFTWARE, DOWNLOAD, FREE SERVER.

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