Consider SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the transport technology for person-to-person real time traffic over the Internet. 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 expanded to handle IP telephony in conjunction with video and instant messaging. Future enhancements will allow SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Sugar Grove.
Makers of SIP hardware are rapidly crafting new products and software to take advantage of this new Internet communications technique. 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 organization, you should think about the appropriate settings of your corporate firewall to accept SIP. Many of the common firewalls already deployed in business offices are not designed to support the SIP protocol. First, SIP media streams are transmitted over dynamically allocated UDP ports that are normally blocked 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 administrator will need to evaluate how to appropriately support SIP to get around these technical challenges. By adding a SIP proxy and registrar for managing the firewall, it is feasible to handle complex SIP situations for secure and private communications.
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