Why You Need SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the transport protocol for person-to-person voice data 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 deal with 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 Cottonwood.
Makers of SIP equipment are quickly developing innovative hardware and software to take advantage of this new Internet communications scheme. SIP telephones, PC client applications, SIP servers, routers, and firewalls are now obtainable from companies such as Ingate Systems and Cisco.
Prior to setting up a SIP solution in your organization, you should consider the appropriate settings of your corporate firewall to accept SIP. Many of the common firewalls currently installed in business offices are not developed to allow the SIP protocol. First, SIP media streams are transferred over dynamically allocated UDP ports that are typically shut on firewalls. Second, SIP clients inside a firewall can not be accessed using IP addresses since these addresses are local and private to the LAN. Third, you need to ensure that either your T1 line broadband carrier or Ethernet fiber provider is able to handle the SIP protocol from your internal network to the outside world. Your IT administrator will need to evaluate how to properly support SIP to overcome these technical problems. By adding a SIP proxy and registrar for managing the firewall, it is possible to deal with complicated SIP scenarios for secure and private communications.
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