Why You Need SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the transmission technology for person-to-person real time data over the Web. Its technical 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 deal with 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 companies in Johnson City.
Suppliers of SIP hardware are quickly crafting new hardware and software to get the upper hand in this new Internet communications scheme. SIP telephones, PC client programs, SIP servers, routers, and firewalls are now available from companies such as Ingate Systems and Cisco.
Prior to implementing a SIP plan in your organization, you should take into account the appropriate configuration of your corporate firewall to accept SIP. Many of the typical 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 normally 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 provider 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 overcome these technical issues. By including 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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