In this web page, we will use the term Internet telephony to refer to the transmission of digitized voice conversations over the Internet by individual PC users. The technology associated with Internet telephony is primarily based on the use of sound cards installed in a PC a microphone connected to the sound card and appropriate software. However, later we will note that there are many flavors of Internet telephony to include phone-to-phone transported over an IP network as well as PC-to-PC and PC-to-phone. Because of the difficulty in classifying these techniques as an individual method used by a consumer or as individual techniques cumulatively used by businesses.
The basic operation of an Internet telephony system commences when a person talks into a microphone. The microphone is in turn connected to a sound card installed in the computer, which accepts an analog waveform and converts it into a digital data stream. Internet telephony software operating on the computer takes the digitized voice data stream, which normally represents a 64-Kbps PCM or a 32-Kbps ADPCM-encoded voice, and compresses the standard encoding data stream into a lower data rate based on the use of a proprietary or standardized voice-compression technique. Once this is accomplished, the software packages the digitized and compressed data stream into packets using a protocol for transmission over the Internet. Most Internet telephony products were originally developed primarily to support modem connections; however, modern products also support LAN-based operations when the LAN is connected to the Internet.
There are two primary transport protocols used for an Internet telephony session. TCP is used to transport addressing or directory information, while UDP is used for the actual transfer of voice-digitized packets. Although the actual ability to digitize voice entered through a microphone is a relatively simple process, differences in the manner in which connections are established over the Internet, voice-digitization methods, and the framing of digitized voice samples result in a high degree of incompatibility between vendor products. Before turning our attention to the operation of specific products, let‘s digress a bit and discuss the economic issues associated with Internet telephony and its basic operation.
Understanding how the public switched telephone network (PSTN) function is useful for discussions on VoIP technology. Hence, this section briefly describes how the current PSTN works. There are four major tasks the PSTN must perform to connect a call. Although there are other services besides an end-to-end voice call
(for example, conference calls and other services), they are based on the following requirements.
- Signaling Database services
- Call connect and tear-down
- Voice to digital conversion
Phone calls are inherently connection-oriented. That is, the connection to the called person must be established ahead of time before the conversation can occur. Switches, the central components in a PSTN, are responsible for creating this connection. Between the circuit switches are connections (trunk links) that carry the voice traffic. These links vary in speed from T1 and E1 to OC-192c/STM-64, with individual channels (DS-0s) in each link type representing one voice channel. Switches are also responsible for converting the analog signal (voice) to a digital format that is transported across the network.
Signaling notifies both the network and its users of important events. Examples of signaling range from the ringer activation letting you know that a call is coming, to the dialing of digits used to make a call. Network elements also use signaling to create connections through a network.
The Signaling System Seven (SS7) network is a packet-based (connectionless) network that transports the signaling traffic between the switches involved in the call. The service control points (SCPs) are the databases that execute the queries to translate phone numbers into circuit-switching details. They also make it possible for such features as 800 number support, 911 service, and caller ID. Signaling switch points (SSP) are the interfaces between the circuit switching equipment and the SS7 network. It is here where SS7 messages are translated into the connection details that the switch needs to connect a call.
Generally, the SS7 control network is out of band (not included) with the same links used to carry the actual voice channels. Specialized equipment called signal transfer points (STPs) transport the signaling messages. These STPs are analogous to IP routers in that the messages are carried in packets called the message transfer parts (MTPs).
The SS7 network is quite extensive (a large collection of networks) and is deployed throughout most of the developed world. There are many technical and historical reasons why the signaling portion of the network is broken out from the rest of the system. However, the greatest value in such a design is to enable you to add network intelligence and features without a dependency on the underlying circuit-switching infrastructure.
When someone picks up the phone receiver, the public switch is alerted and prepares for the phone number digits to be dialed. This phone switch might be a private branch exchange (PBX) in the same building as the phone or a public switch that is miles away. As the digits are dialed, the originating switch analyzes the digits to see if they are valid and if the destination phone is connected to this same switch. If the call is a local call (not outside the exchange), the switch connects the logical channels of the phones involved and the call is completed.
If the call is not local (for example, an 800 number), the originating switch directs a message to a database. Note the query might not be resolved directly by any particular database and that other provider databases might resolve the requested connection. The initial query results in the intervening switches connecting the logical channels that lead to the destination phone. The destination switch signals the destination phone by activating the ringer. The called party has the option to answer the phone and complete the connection.
When the conversation takes place, the switches at this point must be able to convert the voice (analog signal) into a digital form for transport over the network. Once the call is completed, the switches notify the rest of the network to tear down the connections. There are many more details to this transaction; however, these steps describe the basic flow of events in completing a call (Figure 1). In addition, there are a great many supervisory messages that are passed along the network, such as ringing indication, busy signal, and hang up.
VoIP components must be able perform the same features as the PSTN network.
- Database services
- Call connect and disconnect (bearer control)
- CODEC operations
Signaling in a VoIP network is just as critical as it is in the legacy phone system. The signaling in a VoIP network activates and coordinates the various components to complete a call. Although the underling nature of the signaling is the same, there are some technical and architectural differences in a VoIP network. Signaling in a VoIP network is accomplished by the exchange of IP datagram messages between the components. The format of these messages is covered by any number of standard protocols. Regardless of which protocol and product suites that are used, these message streams are critical to the function of a voice-enabled network and might need special treatment to guarantee their delivery.
Database services are a way to locate an endpoint and translate the addressing that two (usually heterogeneous) networks use. For example, the PSTN uses phone numbers to identify endpoints, while a VoIP network could use an IP address (address abstraction could be accomplished with DNS) and port numbers to identify an endpoint. A call control database contains these mappings and translations. Another important feature is the generation of transaction reports for billing purposes. You can employ additional logic to provide network security, such as to deny a specific endpoint from making overseas calls on the PSTN side. This functionality, coupled with call state control, coordinates the activities of the elements in a VoIP network.
The connection of a call is made by two endpoints opening communications sessions between each other. In the PSTN, the public (or private) switch connects logical DS-0 channels through the network to complete the calls. In a VoIP implementation, this connection is a multimedia stream (audio, video, or both) transported in real time. This connection is the bearer channel and represents the voice or video content being delivered. When communication is complete, the IP sessions are released and optionally network resources are freed.
Legacy Voice Services
Call Connect & Disconnect
Voice Protocols & Usages
Signaling System Seven
Real-Time Transport Protocol
Transport Control Protocol
Media Gateway Control Protocol
Session Initiation Protocol
Resource Reservation Protocol
VoIP Service Consideration
Internet Telephony Operations
Hardware and Software Requirements
Constraints and Compatibility Issues
Lan Connectivity Operation
VPN (Virtual Private Network)
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