Consider SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the transport technology for person-to-person voice data 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 extended to deal with IP telephony in conjunction with video and instant messaging. Future enhancements will allow SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for companies in Globe.
Suppliers of SIP components are quickly creating innovative products and software to get the upper hand in this new Internet communications method. SIP telephones, PC client software, SIP servers, routers, and firewalls are now on the market from companies such as Ingate Systems and Cisco.
Prior to installing a SIP plan in your organization, you should take into account the proper settings of your corporate firewall to accept SIP. Many of the typical firewalls already deployed in business offices are not developed to support the SIP protocol. First, SIP media streams are transferred over dynamically allocated UDP ports that are often shut on firewalls. Second, SIP clients within a firewall can not be reached using IP addresses because these addresses are local and unique 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 manager will need to ascertain how to appropriately support SIP to get around these technical issues. By including a SIP proxy and registrar for managing the firewall, it is possible to deal with complex SIP situations for secure and private communications.
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