Consider SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the transmission technology for person-to-person voice traffic over the Web. Its defining specifications come from the SIP working group of the Internet Engineering Task Force. SIP gives 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 expanded to deal with IP telephony combined with video and instant messaging. Future improvements will enable SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for companies in Martin.
Producers of SIP components are rapidly crafting new hardware and software to take advantage of this new Internet communications scheme. SIP telephones, PC client applications, SIP servers, routers, and firewalls are now available from companies such as Ingate Systems and Cisco.
Before installing a SIP solution in your company, you should think about the proper settings of your corporate firewall to support SIP. Many of the common firewalls already installed in business offices are not developed to support the SIP protocol. First, SIP media streams are transferred over dynamically assigned UDP ports that are typically shut 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 carrier or Ethernet fiber service is able to handle the SIP protocol from your internal network to the outside world. Your IT manager will need to evaluate how to appropriately support SIP to get around these technical difficulties. By including a SIP proxy and registrar for controlling the firewall, it is feasible to deal with complicated SIP scenarios for secure and private communications.
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