Consider SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the transport 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 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 expanded to deal with IP telephony in conjunction with video and instant messaging. Future enhancements will enable SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Kerhonkson.
Makers of SIP components are rapidly creating innovative hardware and software to get the upper hand in this new Internet communications technique. SIP telephones, PC client software, SIP servers, routers, and firewalls are now available from companies such as Ingate Systems and Cisco.
Before installing a SIP plan in your organization, you should think about the proper configuration of your corporate firewall to accept SIP. Many of the typical firewalls currently deployed in business offices are not developed to allow the SIP protocol. First, SIP media streams are transmitted over dynamically allocated UDP ports that are usually shut on firewalls. Second, SIP clients inside a firewall can not be accessed using IP addresses since 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 support the SIP protocol from your internal network to the outside world. Your IT administrator will need to ascertain how to correctly support SIP to overcome these technical issues. By including a SIP proxy and registrar for managing the firewall, it is feasible to handle complicated SIP scenarios for reliable and private communications.
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