Consider SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the transmission technology for person-to-person real time data over the Internet. Its technical 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 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 businesses in Jasper.
Producers of SIP hardware are quickly crafting innovative hardware 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 setting up a SIP solution in your company, you should take into account the appropriate settings of your corporate firewall to support SIP. Many of the common firewalls currently deployed in business offices are not designed to support the SIP protocol. First, SIP media streams are transferred over dynamically assigned UDP ports that are often closed on firewalls. Second, SIP clients inside a firewall can not be accessed using IP addresses because 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 support the SIP protocol from your internal network to the outside world. Your IT administrator will need to evaluate how to correctly support SIP to overcome these technical difficulties. By adding a SIP proxy and registrar for controlling the firewall, it is feasible to handle complicated SIP scenarios for reliable and confidential communications.
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