VoIP systems digitize and transmit analog voice signals as a stream of packets over a digital data network. IP networks allow each packet to independently find the most efficient path to the intended destination, thereby using the best network resources at any given instant.
Packets associated with a single source may take many different paths to the destination when traversing the network. At the destination, the packets are re-assembled and converted back into the original voice signal.
VoIP technology insures proper reconstruction of voice signals, compensating for echoes made audible due to the end-to-end delay, for jitter and for dropped packets.
- Can be a standard LAN.
- A network of leased facilities
Gateway VoIP technology rides over a pure IP network but individual calls can be made or received using standard analog, digital and IP phones. VoIP gateways, usually a dedicated server, serve as a bridge between the PSTN and the IP network.
- Call is placed over the local PSTN network to the nearest gateway server.
- Which digitizes the analog voice signal, compresses it into IP packets and moves it onto the Internet for transport to a gateway at the receiving end.
- Computer-to-telephone calls.
- Telephone-to-computer calls.
- Corporate PBX(Private Branch Exchange) can be configured so that all international direct dialed calls are transparently routed to the nearest gateway.
- Tremendous cost savings relative to the PSTN - Remote offices and users can bypass long-distance carriers and their per-minute usage rates and run their voice traffic over the Internet for a flat monthly Internet-access fee.
- Integrated infrastructure - Small businesses are able to deploy one network for voice and data-communications, further reducing costs.
- Scalability - These systems are modular and can be scaled according to the needs of users.
- Database services.
- Calls connect and disconnect (bearer control).
- CODEC operations.
- The signaling in a VoIP network activates and coordinates the various components to complete a call.
- Signaling in a VoIP network is accomplished by the exchange oOP datagram messages between the components. The format of these messages is covered by any number of standard protocols
- Database services are a way to locate an endpoint and translate the addressing that two networks use.
- VoIP network could use an IP address (address abstraction could be accomplished with DNS) and port numbers to identify an endpoint.
- Another important feature is the generation of transaction reports for billing purposes.
- Additional logic to provide network security, such as to deny a specific endpoint from making overseas calls on the PSTN side
Call Connect and Disconnect
- 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-O 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.
- Voice communication is analog, while data networking is digital. The process of converting analog waveforms to digital information is done with a coder-decoder.
- CODECs compress the data stream, and provide echo cancellation. Compression of the represented waveform can afford you bandwidth saving.
- Media gateways.
- Media gateway / signaling controllers.
- IP network.
Media gateways are responsible for call origination, call detection, analog-to-digita1 conversion of voice, and creation of voice packets (CODEC functions). In addition, media gateways have optional features, such as voice (analog and/or digital) compression, echo cancellation, silence suppression, and statistics gathering.
- Media gateway controllers house the signaling and control services
- The media gateway controller has the responsibility for some or all of the call signaling coordination, phone number translations, host lookup, resource management, and signaling gateway services to the PSTN.
VoIP network behaves like a one logical switch. However, this logical switch is a distributed system, rather than that of a single switch entity; the IP backbone provides the connectivity among the distributed elements. Depending on the VoIP protocols used, this system as a whole is sometimes referred to as a softswitch architecture.
Signaling System Seven:
SS 7 is a widely used suite of telephony protocols expressly designed to establish and terminate phone calls. The SS7 signaling protocol is implemented as a packet-switched network.
The ITU recommendation H.323 is a packet-based multimedia communication system that is a set of specifications. These specifications define various signaling functions, as well as media formats related to pocketsize audio and video services.
- H.323 networks consist of (media) gateways and gatekeepers.
- Gateways serve as both H.323 termination endpoint and interface with non-H.323 networks, such as the PSTN.
- Gatekeepers function as a central unit for call admission controL bandwidth management, and call signaling.
- A gatekeeper and all its managed gateways form an H.323 zone.
- Real-time Transport Protocol.
RFC 1889 and RFC 1890 cover the R TP, which provides end-to-end delivery services for data with real-time characteristics, such as interactive audio and video. Services include payload type identification, sequence numbering, time stamping, and delivery monitoring.
Session Initiation Protocol
The Session Initiation Protocol (SIP, RFC 2543) is part of IETF's multimedia data and control protocol framework. SIP is a powerful client-server signaling protocol used in VoIP networks. SIP handles the setup and tear down of multimedia sessions between speakers; these sessions can include multimedia conferences, telephone calls, and multimedia distribution.
- SIP is a text-based signaling protocol transported over either TCP or UDP, and is designed to be lightweight. It inherited some design philosophy and architecture from the Hypertext Transfer Protocol (HTTP) and Simple Mail Transfer Protocol (SMTP) to ensure its simplicity, efficiency and extensibility.
- SIP supports user mobility by proxying and redirecting requests to the user's current location. Users can inform the server of their current location (IP address or URL) by sending a registration message to a registrar. This function is powerful and often needed for a highly mobile voice user base.
- Signaling Transport (SigTran).
- Media Gateway Control Protocol.
- Packet switched network are connectionless compared to Circuit switched which are connection oriented.
- QoS mechanism is needed for VoIP calls.
- RSVP (Resource reservation protocol) is needed to reserve bandwidth and router/switch buffer space for high priority IP Packet carrying voice traffic.
- End user demand is expected to grow rapidly over the next 5 years and according to recent research.
- VoIP will be deployed by 70% of fortune 1000 companies.
- Many developers and manufacturers now offer PC telephony hardware and software but also gateway servers are emerging to act as an interface between the Internet and the PSTN making VoIP calls almost an obvious choice.
- Multiple means of communications will be integrated into a VoIP system. These include e-mail, fax, voice mail, video conferencing, SMS, instant messenger leading to the concept of UM.
- It will include iPBX ( IP based PBX) which will allow both standard telephone and multimedia PC's to connect to either PSTN or the Internet providing seamless migration path of VoIP.
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