Communications Service Overview

Article
06/30/2006

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.

Feature Summary

The following table shows operating system design information for the Communications Service.

Element	Information
Dependencies	TCP\IP Winsock 2.2 Audio Hardware support Cryptography Services Schannel XML Core Services XML Query Languages
Hardware considerations	Input Minimum: 16 KHz, mono, 16 bit Ideal: 44.1 KHz, stereo, 16 bit Output Minimum: 22.050 KHz, mono, 16 bit Ideal: 44.1 KHz, stereo, 16 bit

Dependencies

TCP\IP

Winsock 2.2

Audio Hardware support

Cryptography Services

Schannel

XML Core Services

XML Query Languages

Hardware considerations

Input

Minimum: 16 KHz, mono, 16 bit
Ideal: 44.1 KHz, stereo, 16 bit

Output

Minimum: 22.050 KHz, mono, 16 bit
Ideal: 44.1 KHz, stereo, 16 bit

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.

Item	Module	Component
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
Media Manager	voipmedia_ref	None
NAT Interface	dpnhupnp	None
PINT Stack	rtcclient	None
RTP protocol and Codecs	dxmrtp	None
SIP Stack	rtcclient	None
VoIP Manager	voipmanager	None