The LanScape Centrex Proxy Server® is an enhanced and highly integrated SIP (Session Initiation Protocol) proxy server. It has been developed to include a wide range of capabilities that will simplify the deployment of SIP based VOIP telephony networks. It is simple to install, configure and deploy on any version of 32 bit Microsoft Windows operating systems. The LanScape Centrex Proxy Server® also functions seamlessly with LanScape's VOIP Media Proxy Server® software to enable you to deploy a fully functional, scalable SIP and RTP media proxying solution. If you do not understand why there is a need to also proxy RTP VOIP media streams, please read on. We will describe the reasons below.

The LanScape Centrex Proxy Server® is primarily used to control and manage SIP/RTP VOIP calls that enter and leave your specifically configured VOIP domain. The LanScape Centrex Proxy Server® can perform such tasks as user authentication, call routing, registrar capabilities and control media proxying/media stream management. These are just a few of the capabilities you have at your disposal.

In addition, the LanScape Centrex Proxy Server® allows you to deploy your VOIP network with added security due to the fact that all call traffic must pass through the proxy server and optionally be verified via MD5 challenge handshake authentication. A simple VOIP deployment can be as small as one Centrex Proxy Server® and one VOIP Media Proxy Server® running on a single machine. Larger deployments can use multiple Centrex Proxy Servers and multiple VOIP Media Proxy Servers on individual host machines all accessing the same registration database.

The list of capabilities the LanScape Centrex Proxy Server® can offer you is quite large and extensive. These capabilities will be discussed below.

Background Information:

Currently all VOIP systems attempt to connect two or more call endpoints (peers) that wish to interact with each other. This interaction involves setting up calls, negotiating media exchange parameters and then finally exchanging media streams (the voice path). When a VOIP phone call is over, the call session must be "torn down" and the exchange of voice media stream data terminated.

If you are new to deploying SIP based VOIP telephony networks, we have additional information for you that you may not know. If you are a veteran VOIP developer or a seasoned VOIP IT professional, then you already know the inherent problems facing the deployment of VOIP networks in today's IP4 network environment. Specifically, we are speaking of the hostile nature of todays network environment to VOIP and the role that network address translation (NAT) and port translation play in the deployment. From this point forward, we will use the term NAT to represent network address translation and port translation.

The session Initiation Protocol (SIP) is used to manage call setup and termination as described above. The RTP protocol is used to exchange the call's media stream (voice data). SIP is a very useful and relatively simple protocol that is effective in establishing multimedia interchanges (VOIP phone calls). However, the SIP protocol was developed without the ability to overcome network addressing problems caused by NAT. In short, NAT is the worst thing to happen to VOIP and the SIP protocol. The problems introduced by NAT completely destroy the ability of call endpoints (peers) to communicate without errors. It should be noted that the issues NAT introduces is not specific to the SIP protocol and VOIP networks. The same problems occur with any other peer to peer networking environment that encode IP address information and UDP port information in the protocol packets that are exchanged.

One way to overcome most of the problems NAT introduces is to use other protocols to assist in discovering proper IP address and port information. Notice that we have used the phrase "most of the problems". There is a reason for this. Additional protocols and "NAT discovery methods" such as STUN, TURN or ICE can be used to obtain this information. Once this real address and port information is known, it is possible to "spoof" or "alter" the information that gets placed in SIP protocol packets. However, with the addition of these new protocols and methods, comes additional complexity, additional points of failure and it still does not solve the problems facing user's that deploy symmetrical NAT routers in their network environment.

Other proprietary protocol schemes used by some companies or VOIP service provider attempt to "get around" the problems introduced by NAT. They employ proprietary signaling protocols and servers that allow peer to peer connectivity to be achieved. However, this also has been shown not to be a general solution. There are issues facing these proprietary schemes such as the failure to interoperate with other VOIP networks. There are also security risks. One such security risk is the use of protocols that tunnel their VOIP traffic using the HTTP protocol. By using a tunneling approach, network administrators no longer can treat normal HTTP web traffic differently from VOIP signaling and media traffic. This is not desirable in highly secure network environments.

To achieve a secure and general solution for all types of NAT devices that is low on the "complexity scale" requires two basic components: A SIP proxy and a media proxy. With these two network elements properly deployed, it will not matter if call endpoints are in their own private networks or in the global IP address space. It will also not matter what kind of firewall or router the call endpoint are behind. The SIP proxy assists the call endpoints in setting up and tearing down calls sessions and the media proxy allows the call endpoints to exchange media securely. Using this scenario will allow every SIP and RTP compliant application or device to communicate. It should also be noted that the "media deadlock problem" associated with deploying VOIP media proxies has been solved when you deploy your VOIP network using LanScape SIP and media proxies.

For those individuals that argue that adding SIP and media proxies to the VOIP deployment add an "additional point of failure", we say this: redundancy. If you are deploying a 24x7 VOIP service and you require 99.9999% up time, plan your load sharing and redundancy architecture appropriately. Compared to deploying additional protocols and servers to support STUN, TURN and ICE for NAT resolution, you will still be faced with additional "points of failure" and undue complexity. Also, we agree that proxying media adds an extra "hop" to the media stream's network path. However, we like the fact that all connectivity issues fade and every SIP/RTP software application and device will work as expected.

LanScape's Centrex Proxy Server®:

The graphic image below shows the user interface of the LanScape Centrex Proxy Server®.

There are three main areas to the user interface. The first area displays current critical settings for the proxy such as IP address, WAN IP address, the server port, etc. The second area of the user interface displays proxy activity and is depicted above as the "Activity log area". In this area, you will see all proxy activity as it happens real time. Through the configuration dialogs, a user can enable or disable certain logging information types. The third area of the user interface that is of interest is the "Runtime Statistics" bar. The information that is displayed there gives you an indication relative to the current registration and call rates in addition to max number of registrations and calls that have been processed since the proxy server was started.

List of Capabilities

The following categories will assist you in understanding the capabilities of the LanScape Centrex Proxy Server®:

Network Configurations:

The following network capabilities are available to you: Specify your VOIP domain name. Specify the server's IP address and port. Configure your outermost router/gateway IP address. Configure the maximum allowed size of received SIP messages. Enable or disable denial of service attack protection. Note that the Centrex Proxy Server® uses symmetrical signaling to assist in overcoming NAT related issues. Symmetrical signaling is just a fancy way of saying the Centrex Proxy Server® sends and receives SIP protocol packets using the same configured UDP port.

Call Processing Timeouts:

Allows you to configure call processing timeouts for call operations such as call rollover, call start response and final call answer.

Local User Directory:

The LanScape Centrex Proxy Server® allows you to configure static registered user names or extensions so that they always appear in the registry database. In this way, you can configure call routing based on these static names. If a call is received by the proxy, it will attempt to apply a user defined call routing plan and route the call. Also SIP devices that can receive registry notifications can display local directory information that is the result of configuring the local directory.

Call Routing:

The LanScape Centrex Proxy Server® allows you to configure call routing lists for any number of registered users. When a phone call is received by the proxy, it determines the destination of the call and looks up the call routing plan for the user name (or phone extension). If a call routing plan is not defined, the call is sent on to the call destination if the call destination is registered. If a route plan is found, the proxy will ring all call destinations sequentially in the route list until the call is answered or the maximum call processing timeout is reached. Call routing definitions can also use regular expression syntax so you can generically forward any type of call to a specific destination like a PSTN gateway.

Global iNet® Account Support:

If you have key individuals or employees in your organization, you can purchase optional Global iNet® user accounts and allow them to obtain a directory listing in the Global iNet® phone system. The proxy logs in any configured Global iNet® user automatically when it starts. This way, other Global iNet® user can search the online directory server database via your company name, your last name or whatever and call you. Global iNet® is supported by this proxy server and all LanScape soft phones.

Full control over one or more LanScape VOIP Media Proxy Servers:

To obtain a complete VOIP connectivity solution, the LanScape Centrex Proxy Server® can control one or more LanScape VOIP Media Proxy servers. The Centrex Proxy Server® communicates with the media proxies and assigns media streams to one of the available media proxies via a load sharing algorithm. All active call media streams will be divided equally among all available media proxies. LanScape VOIP Media Proxy solutions also overcome the common "media proxy deadlock issue" most media proxies are faced with. Communications with subordinate media proxies are authenticated using an MD5 challenge algorithm for added security.

Full registrar database support:

Full registrar database support using LanScape's integrated (very fast) registrar database capability or an external registrar database using any database type you require (Microsoft Access, SQL, Open source MySql, etc). The only requirement for external registrar database is to have proper ODBC drivers installed on the host machine. Sharing a registrar database among multiple Centrex Proxy Servers requires that an external registrar database be used.

Integrated WAN IP address detection and discovery:

To be useful in the most arcane circumstances, the LanScape Centrex Proxy Server® can be configured to determine its WAN IP address using: DNS and a configured domain name, by specifying a static WAN IP address or by dynamically monitoring your WAN IP address using HTTP requests to a web server or router configuration page.

Custom DLL Plug-in Interface:

The LanScape Centrex Proxy Server® can use a user supplied plug-in DLL that it accesses at runtime. With this simple plug-in interface, you have the ability to alter any incoming or outgoing SIP message the proxy receives or transmits. The example plug-in that is supplied with the proxy server gives you a starting point. It include source code and performs realtime SIP protocol logging to the Sip Log Server application. The Sip Log Server application is part of your software distribution.

SIP Proxy event log control:

Enable or disable the types of events the Centrex Proxy Server® reports on its user interface.

SIP protocol logging:

Enable or disable SIP protocol logging to a log file of your choice.

Complete challenge handshake authentication using the MD5 algorithm:

For the most secure VOIP networks, the LanScape Centrex Proxy Server® supports encrypted authentication using the MD5 algorithm. You have the ability to configure when authentication should be applied to specific SIP message types. You can control authentication of the following SIP messages: REGISTER, INVITE, BYE, SUBSCRIBE and NOTIFY. Using a single authentication "realm" you can assign one or more login names and password pairs. In this way, you can authenticate a whole group of users using the same login and password or authenticate each individual VOIP user by assigning them their own user name and password.