Why You Need SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the transmission technology for person-to-person voice traffic over the Web. 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 extended to handle IP telephony combined with video and instant messaging. Future enhancements will enable SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Springville.
Makers of SIP equipment are quickly developing innovative products and software to get the upper hand in this new Internet communications scheme. 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 organization, you should consider the proper configuration of your corporate firewall to accept SIP. Many of the typical firewalls already deployed in business offices are not designed to support the SIP protocol. First, SIP media streams are transferred over dynamically allocated UDP ports that are often blocked on firewalls. Second, SIP clients inside 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 provider is able to support the SIP protocol from your internal network to the outside world. Your IT manager will need to evaluate how to appropriately support SIP to overcome these technical problems. By adding a SIP proxy and registrar for managing the firewall, it is possible to deal with complex SIP situations for reliable and private communications.
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