Consider SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the transmission technology for person-to-person real time data over the Web. Its defining specifications come from the SIP working group of the Internet Engineering Task Force. SIP provides access to the public switched telephone network for voice data at 3 kHz bandwidth and common number dialing using Voice over IP (VoIP). It can also be extended to deal with IP telephony in conjunction with video and instant messaging. Future improvements will enable SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Hartman.
Manufacturers of SIP components are rapidly creating new hardware and software to get the upper hand in this new Internet communications scheme. SIP telephones, PC client programs, SIP servers, routers, and firewalls are now obtainable from companies such as Ingate Systems and Cisco.
Prior to setting up a SIP plan in your organization, you should think about the appropriate settings of your corporate firewall to accept SIP. Many of the typical firewalls already deployed in business offices are not designed to support the SIP protocol. First, SIP media streams are transmitted over dynamically allocated UDP ports that are normally blocked on firewalls. Second, SIP clients within a firewall can not be accessed using IP addresses since these addresses are local and unique to the LAN. Third, you need to ensure that either your T1 line broadband carrier or Ethernet fiber provider is able to handle the SIP protocol from your internal network to the outside world. Your IT manager will need to ascertain how to appropriately support SIP to overcome these technical issues. By including a SIP proxy and registrar for managing the firewall, it is feasible to deal with complex SIP scenarios for secure and private communications.
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