Consider SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the transmission protocol for person-to-person voice traffic over the Web. Its technical 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 enhancements will enable SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for companies in Crump.
Producers of SIP hardware are rapidly creating new products and software to get the upper hand in this new Internet communications method. SIP telephones, PC client programs, SIP servers, routers, and firewalls are now on the market from companies such as Ingate Systems and Cisco.
Prior to setting up a SIP plan in your business, you should take into account the appropriate configuration of your corporate firewall to accept SIP. Many of the common firewalls currently deployed in business offices are not developed 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 because these addresses are local and private to the LAN. Third, you need to ensure that either your T1 line provider or Ethernet fiber carrier is able to handle the SIP protocol from your internal network to the outside world. Your IT administrator will need to evaluate how to properly support SIP to get around these technical difficulties. By adding a SIP proxy and registrar for managing the firewall, it is possible to deal with complicated SIP situations for secure and private communications.
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