Consider SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the transmission protocol for person-to-person real time data 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 extended to deal with IP telephony in conjunction with video and instant messaging. Forthcoming improvements will enable SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Angel Fire.
Makers of SIP components are quickly crafting new hardware and software to take advantage of this new Internet communications method. SIP telephones, PC client applications, SIP servers, routers, and firewalls are now available from companies such as Ingate Systems and Cisco.
Prior to setting up a SIP plan in your company, you should take into account the appropriate settings of your corporate firewall to support SIP. Many of the common firewalls already 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 within a firewall can not be reached 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 correctly support SIP to overcome these technical problems. By including a SIP proxy and registrar for managing the firewall, it is possible to handle complex SIP situations for secure and private communications.
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