Consider SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the transport 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 extended to handle IP telephony combined with video and instant messaging. Forthcoming enhancements will enable SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Leeds.
Makers of SIP components are rapidly crafting innovative hardware and software to take advantage of this new Internet communications method. SIP telephones, PC client programs, 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 take into account the proper configuration of your corporate firewall to support 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 typically closed on firewalls. Second, SIP clients within 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 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 correctly support SIP to get around these technical problems. By including a SIP proxy and registrar for controlling the firewall, it is possible to handle complex SIP scenarios for secure and private communications.
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