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VoIP System
Gatekeeper: A gatekeeper is useful for handling VoIP call connections includes managing terminals, gateways and MCU's (multipoint control units). A VoIP gatekeeper also provides address translation, bandwidth control, access control A VoIP gatekeeper can improve Security and Quality of Service (QoS) VoIP Gateway: A VoIP gateway is also required to handle external calls. A VoIP gateway functions as a converter that converting VoIP calls to/from the traditional PSTN lines VoIP Clients: Other required VoIP hardware includes a VoIP client terminal, a VoIP device could be an IP Phone, or a multimedia PC or a VoIP-enabled workstation runs VoIP software.
PSTN vs. VoIP PSTN Voice networks use circuit switching. Dedicated path between calling and Called party. Bandwidth is reserved in advance. Each line is 64kbps. Cost is based on distance and time. Features such as call waiting, Caller ID and so on are usually available at an extra cost VoIP VoIP uses packet switching. No dedicated path between sender and receiver. It acquires and releases bandwidth, as it is needed. Cost is not dependent on time and distance. Features such as call waiting, Caller ID and so on are usually included free with service
PSTN VoIP Can be upgraded or expanded with new equipment. Long distance is usually per minute or bundled minute subscription. Hardwired landline phones (those without an adapter) usually remain active during power outage. Upgrades usually requires onl y bandwidth and software up grades. Long distance is often included in regular monthly p rice. Lose power, lose phone service without power backup in place.
Basic Principles of VoIP Audio Codec, Video Codec Data Transport (RTP, RTCP) Addressing Signaling (SIP, H.323)
Audio Codecs Are used to convert analog signal into digital data. The most common codecs for VoIP are Codec Bandwidth/kbps G.711 64 G.722 48/56/64 G.723.1 5.3/6.3 Stands for coder-decoder: Since voice contains lot of data, it is compressed by coders without compromising the reliability and quality of voice signal.
Translation of Analog Signal to Digital Signal
Video Codec Video Codec: common examples include H.261 (for 64kbps and above), H.263 (for 64kbps and below), and MPEG 4. The encoded information is then encapsulated within an IP packet and these packets are then transported across the network to their destination.
Data Transport (RTP,RTCP) RTP It stands for Real time Transport Protocol. Application layer protocol for transmitting real time data (audio, video,...) Includes sequence numbering, time stamping, delivery monitoring.
RTCP It stands for Real time Transport Control Protocol. While RTP carries the media streams (e.g., audio and video), RTCP is used to monitor transmission statistics and quality of service (QoS) and aids synchronization of multiple streams. Main functions: support for multi-point communication
VoIP SIGNALING PROTOCOLS Signaling in VoIP is needed for -To establish a point to point connection and to keep it open for the duration of the call. -Agreeing on coding /decoding procedures. Types of Signaling Protocols: H.323 SIP
H.323
H.323 H.323 is a set of protocols for sending voice, video and data over IP network to provide real-time multimedia communications. H.323 is reliable and easy to maintain technology 1. Terminals 2. Gateways 3. Gatekeepers 4. Multipoint control units (MCUs) An H.323 zone is a collection of all terminals, gateways, and MCUs managed by a single gatekeeper. A zone includes at least one terminal and may include gateways or MCUs. A zone has only one gatekeeper. A zone may be independent from network topology.
There are four basic entities in a default H.323 network : Terminal: H.323 terminal also called H.323 client is the end-user device. It could be IP telephone or a multimedia PC with another H.323 client. That provides real-time two-way media communication. H.323 Gateways (GW): A Gateway (GW) is an optional component that provides inter-net work translation between terminals. Gatekeepers (GK): A Gatekeeper (GK) is an optional component provides address translations and access control services. Multipoint Control Units (MCU):A Multipoint Control Unit (MCU) functions as a bridge or switch that enables three or more terminals and gateways in a multipoint conference.
H.323 Characteristics Allows transmission of video and data while a phone call is in progress Incorporates protocols for security. Uses a special hardware Multipoint Control Unit for conferencing calls. Defines servers for address resolution, authentication, accounting, features, etc.
SIP SIP stands for Session Initiation Protocol. Developed by IETF since 1999. SIP is the core protocol for initiating, managing and terminating communication sessions (i.e audio & video call) over the Internet. These sessions may be text, voice, video or a combination of these. SIP sessions involve one or more participants and can use unicast or multicast communication. Sessions include Internet Multimedia conferences or Internet Telephone calls.
SIP SIP is a signaling control protocol which is similar to http. it s designed to initial and terminate VoIP sessions with one or more participants. It is less weight and more flexible than H.323 that also can be used for multimedia sessions such as audio, video and data.
SIP SIP has two components: User Agents and SIP servers. User agents are peers in a SIP. User agents could be either an agent client or an agent server. A user agent client initiates by sending a SIP request. A user agent server can accept, terminate or redirect the request as responses to this SIP request. There are three types of SIP servers include SIP proxy servers, SIP registrar servers, and SIP redirect servers. A SIP server functions as a server that handles these requests, e.g. requests transferring, security, authentication, and call routing. e.g. Microsoft MSN Messenger, Apple ichat.
SIP Characteristics Operates at the application layer. Encompasses all aspects of signaling, e.g. location of called party, ringing a phone, accepting a call, and terminating a call. Provides services such as call forwarding. Relies on multicast for conference calls. Allows two sides to negotiate capabilities and choose the media and parameters to be used.
SIP Methods Six basic message types, known as methods:
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