Communications Service Overview
Communications Service support enables you to build applications that perform text messaging, Voice over IP (VoIP), IP telephony bridging from IP to Public Switched Telephone Network (PSTN) telephony, and receiving/sending presence information. Microsoft® Windows® CE .NET provides Real-time Communications (RTC) application programming interfaces (APIs), which are built on the Session Initiation Protocol (SIP), which is an Internet Engineering Task Force (IETF) standard. SIP allows a device to call, or receive calls from, any SIP client. A SIP client can be a Windows CE device, Windows XP machine, or a third-party SIP user agent. The Windows Messenger for Microsoft Windows CE .NET is a client application that supports the Communications Service as a backend.
Windows CE .NET 4.2 and later include the following new additions:
- RTC Client API version 1.2
- VoIP Application Interface Layer (VAIL)
- VoIP reference Telephony User Interface (TUI)
The Communications Service performs the following functions:
- Setting-up VoIP connections among Windows CE devices, .NET devices, Microsoft Windows XP systems, and third-party SIP user agents, or PSTN phones.
- Initiates text-messaging sessions, which can occur simultaneously with voice connections.
- Receiving presence information to see if your contacts are online; enabling you to change your own presence state and send notifications; and monitoring who is watching the user's online presence.
The following table shows operating system design information for the Communications Service.
Audio Hardware support
XML Core Services
XML Query Languages
For fully functional audio capabilities, one or more of the following solutions must be implemented:
A PCI bus reserved for an appropriate audio card.
If the audio is onboard, it must have an AC97 codec along with a codec controller that provides hardware-assisted audio data transfer to and from the codec. The controller must have enough bandwidth to service the audio and other devices in the system.
The codec controller must also have an interrupt mechanism that can notify the driver when a data transfer is complete. Additionally, the controller should contain sufficient buffering capabilities in audio data to allow the interrupt service thread to start another data transfer without causing audio breakup.
To support VoIP, the capture device and driver should be able to handle wave-in capture buffers as small as 10 milliseconds and return them to the application when they are filled with data.
Note Peripheral hardware like the microphone and earpiece should ideally have automatic gain control and built-in echo cancellation.
Modules and Components
The following table shows the components and modules that implement the Communications Service.
|Call Log Database||voipstore||None|
|Caller Info Database||voipstore||None|
|Communications Service Client API (RTCClient)||rtcclient||None|
|File Directory Client||voipprov_file||None|
|LDAP Directory Client||voipprov_ldap||None|
|Manual Directory Client||voipprov_manual||None|
|RTP protocol and Codecs||dxmrtp||None|
Application Development Topics
Last updated on Friday, April 09, 2004
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