Application VOIP

Today’s Advanced Intelligent Network (AIN) provides many useful features common to callers today. These features include Caller ID, Voice Mail, Call Waiting, Pre-and-post paid calling cards, 911, Call Blocking, and Auto Call-back to name a few. These VoIP Application and features represent years of development and investment by vendors and service providers, and are delivered via a proven circuit switched infrastructure. Advanced VoIP Application: The migration to VoIP networks is being driven by a number of factors; the key concept is that of advanced services. These services will be able to provide all the existing AIN features and add new ones based on IP services. A key aspect of the new VoIP infrastructure is that there is no need to build circuit switched connections between the devices, which reduces the cost of providing the services, and simplifies deployment. All systems, except the media gateway itself, require only IP interfaces for VoIP Application. The IP interfaces of these devices can provide the telephony signaling as well as the media interfaces. This provides for simpler distributed signaling and processing capability reduces the cost of components, and speeds up application development and deployment. The new IP based application or VoIP Application can be delivered in a variety of methods depending upon their complexity.
For simple VoIP Application the Media Gateway Controller (MGC) can provide the application intelligence in a very distributed fashion. The MGC provides call control to the user via the Media Gateway with a client/server protocol called MGCP. The MGC controls all routing and call control to the devices within its’ MGCP domain. These functions very similarly to a Class-5 End Office switch and provide the same features one would expect on a standard POTS line. However it also has the ability to play tone and announcements to a caller, as well as gather digits from the caller. This provides capabilities similar to an SCP or Announcement Server for simple VoIP Application. These features are provided by the capabilities defined in MGCP and H.248. VoIP Application Interface Layer: Windows CE .NET VoIP Application Layer Interface (VAIL), at the core of which is the VoIP Manager, enables original equipment manufacturers (OEMs) to quickly build and distribute customized VoIP devices. Similar to the sample Telephony User Interface (TUI) application, OEMs and network operators can use VAIL or replace it with a different solution. VAIL consists of seven key components:
VoIP Manager: The VoIP Manager interface is the engine of the Windows CE .NET VoIP stack. It manages all aspects of a phone VoIP Application , including:

The Media Manager: The VoIP Manager accesses the pluggable Media Manager interface to set up and respond to all media events. The Media Manager controls the sending and receiving of voice data over the RTP.
Real-Time Communication (RTC) Layer: The RTC layer enables call hold, call transfer, forwarding functionality, and interoperability with the RTC Server. To tailor the VAIL to specific VoIP solutions, OEMs can disable the RTP stack built into the RTC layer and instead implement a custom solution that resolves jitter buffering, acoustic echo cancellation, and other call-related features. The VoIP software provides a telephony API set that is independent of the RTP implementation. RTC handles the signaling (SIP) part of the session while the Media Manager sends and decodes session descriptions using the Session Description Protocol (SDP).Provisioning: To register a phone with a SIP server, you must provide its URI, SIP Server, authentication type, password, and other identifying information. The process of determining this information is known as provisioning. The VAIL contains a dedicated interface called the VoIP Directory Client that simplifies the provisioning process and allows remote or automatic provisioning. The VoIP Directory Client interface accesses the installed Directory Client objects to retrieve the information needed to register the phone with the SIP server.
Databases: VAIL provides two interfaces for persistent local database storage:
1. Call log. A database that stores all call logs. These records indicate the type, start time, end time, and duration of the call, as well as name and URI of the remote party.

2. Caller info. A database that stores caller-specific information such as friendly name, speed dial index, specialized ring tone, blocked status, and forwarding URI.

(Signaling) Protocol Layer: VAIL interfaces with the protocol layer. Frequently used protocols for VoIP devices are SIP, RTP and the interface between a public switched telephone network (PSTN) and an IP network known as the PSTN/Internet (PINT). XML and SIP enable the development of powerful and sophisticated interfaces, applications, and Web services that can be customized to meet individual and vertical industry requirements. To add signaling protocols such as H.323, Media Gateway Control Protocol (MGCP), or Megaco to the protocol layer, use Telephony API (TAPI) or a custom interface. See the image below: Plug-in interface options for signaling protocols.Audio Codices Interface: Various standardized audio codices such as G.711, G.722, and SIREN are included in the VOIP solution of Windows CE .NET 4.2. The Audio Compression Manager (ACM) allows an application to convert data between different formats. In addition, ACM enables device manufacturers and network operators to add additional audio codices as needed. See the image below: ACM plug-in interface.