Why You Need SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the communications protocol 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 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 in conjunction with video and instant messaging. Future improvements will enable SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Chemult.
Manufacturers of SIP hardware are quickly creating innovative hardware and software to get the upper hand in this new Internet communications technique. SIP telephones, PC client programs, SIP servers, routers, and firewalls are now obtainable from companies such as Ingate Systems and Cisco.
Before setting up a SIP plan in your organization, you should consider the appropriate configuration of your corporate firewall to accept SIP. Many of the common firewalls currently installed in business offices are not designed to allow the SIP protocol. First, SIP media streams are transferred over dynamically assigned UDP ports that are usually blocked on firewalls. Second, SIP clients within a firewall can not be reached using IP addresses since these addresses are local and unique to the LAN. Third, you need to ensure that either your T1 line carrier or Metro 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 properly support SIP to overcome these technical problems. By adding a SIP proxy and registrar for managing the firewall, it is possible to handle complicated SIP situations for secure and private communications.
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