AddPac AP-GS1001C
AddPac AP-GS1001C is an entry level VoIP-GSM gateway, designed to work with one GSM channel. The device allows you to connect to the VoIP-telephony, analog lines and GSM networks. At the same time, users are able to transfer traffic in any direction lines are connected to: VoIP to GSM; an analog network to GSM; from analog to VoIP network.
This device simultaneously supports the SIP, MGCP and H.323 protocols, without the need for additional software updates. A very useful function “BabyCall” which is an automatic dialing that greatly facilitates the procedure of dialing frequently used phone numbers in any company.
Among other features, there are black and white lists, which restrict the use of certain numbers, lines or on the contrary, add additional services to the appropriate users of the system.
A Real-time Transport Protocol (RTP) is a protocol standard defined by the Internet Engineering Task Force (IETF) that outlines a management system for programs using real-time transmission of data through mutlicast or unicast network services in multimedia situations. Originally designed by the IETF for videoconferencing for multiple participants, the protocol is largely used in Voice over Internet Protocol applications. Despite its name, the protocol cannot guarantee real-time delivery of data, however, the RTP does compensate for jitters and can detect when data arrives out of sequence, both issues being popular in VoIP communication. In IP telephony, RTP works with a signaling protocol, such as SIP or H.323, in order to set up connections in a network.
Netbone Digital AG
calltermination.biz offers professional and high quality VoIP call termination for VoIP service providers. Call termination can be done trough SIP or IAX as postpaid or prepaid solution. Our network is based on 336 and 330 of these server are connected to
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