Why You Need SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the transport technology for person-to-person voice traffic over the Internet. 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 such as video conference calls, application sharing, home monitoring, and interactive gaming for companies in Ware Shoals.
Suppliers of SIP components are rapidly creating 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 obtainable from companies such as Ingate Systems and Cisco.
Before installing a SIP solution in your company, you should take into account the appropriate configuration of your corporate firewall to support SIP. Many of the common firewalls currently installed in business offices are not designed to support the SIP protocol. First, SIP media streams are transferred over dynamically allocated UDP ports that are typically blocked on firewalls. Second, SIP clients within a firewall can not be reached using IP addresses because these addresses are local and private to the LAN. Third, you need to ensure that either your T1 line provider or Ethernet fiber carrier is able to handle the SIP protocol from your internal network to the outside world. Your IT manager will need to ascertain how to properly support SIP to get around these technical challenges. By including a SIP proxy and registrar for managing the firewall, it is feasible to deal with complex SIP scenarios for reliable and confidential communications.
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