Consider SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) 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 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 handle IP telephony combined with video and instant messaging. Future improvements will enable SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for companies in Joppa.
Suppliers of SIP hardware are rapidly creating new hardware and software to get the upper hand in this new Internet communications technique. SIP telephones, PC client programs, SIP servers, routers, and firewalls are now obtainable from companies such as Ingate Systems and Cisco.
Before installing a SIP plan in your organization, you should consider the appropriate settings of your corporate firewall to accept SIP. Many of the common firewalls currently installed in business offices are not designed to allow the SIP protocol. First, SIP media streams are transmitted over dynamically assigned UDP ports that are often closed 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 broadband provider or Ethernet fiber service is able to support the SIP protocol from your internal network to the outside world. Your IT administrator will need to ascertain how to appropriately support SIP to get around these technical challenges. By adding a SIP proxy and registrar for controlling the firewall, it is feasible to handle complex SIP scenarios for reliable and private communications.
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