DX-200 Nokia is a modular-based hardware and software digital switching system, is designed to be operated as a terminal exchange and terminal and transit exchange with local telephone networks (TE, TTE) and to organize transit nodes.
IP-technology approved by the communications market stronger with each passing year. Already no one doubts that the future belongs to IP networks that will keep pace with the ever-growing needs of network users, as opposed to TDM networks that already today are outdated. Is it possible to “painless” transition, we consider in this article.
SS7 – is an individual network, so voice and data are transmitted one-way, all the signals it sends completely different. Before there was a network of SS7, telephone connection going through the same channels on which there was talk of subscribers. The fact that the user information and signals transmitted at different times. Has contrast SS7 ISDN. The latest network subscribers and switches can send each other signaling channel D, and here in the SS7 this exchange of information is possible only between network components.
SS7 is architecture for performing out-band signaling in support of the creation of call billing, routing information exchange functions of the PSTN (public switched telephone network). SS7 is what functions are executed common channel signaling network, as well as what protocols tedious for its successful execution.
The main problem of IP telephony, is the loss of IP packet transmission in the network. In this publication, I would like to analyze the most common cases. Below is a diagram of a transmission path of voice under ideal conditions.
VoIP-GSM gateway is also called SIP-GSM gateway. It is a device that is responsible for the live broadcast of the telephone signal from the IP network to the cellular network, and then back again. Through this process of saving mechanisms are consistent and they are implemented through the use of GSM-gate together with office phone systems, which are based on VoIP-technology.
Ever since Skype was released in late 2003, it was used as the final adjudication of communication, because it “just works” for consumers. The reason also is its simplicity: do not need any additional modems, routers, etc. You simply install the program and use it, taking calls and messages.
The last time, in the world of web development concept of cloud telephony has become quite frequently used. It appeared a few years ago with Ribbit, and attracted a lot of attention, along with the advent of Twilio and Tropo. As long as the phone features were not easily accessible, the industry itself considered sufficient telephony boring. Now, asynchronous access to telephony API, a variety of applications for telephony, as well as their integration with Web sites pop up everywhere on the Internet. And we can say that this is really a brand new ecosystem.
Operating a VoIP system with a focus on great customer experience can be quite challenging, especially if you run a heterogeneous network with lots of different SIP clients (like various software clients, all kinds of SIP Phones and Terminal Adapters and especially IP PBXs). SIP clients are known to have all kinds of quirks and implementation errors, and if you don’t control them yourselves (e.g. with a central device provisioning tool), the additional factor of configuration errors introduced by your customers comes into play.
Protocol SIP (Session Initiation Protocol) – one of the central protocol of IP telephony, which is described in the recommendations of RFC 2543. It describes the processes of establishing and terminating multimedia sessions.