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Group Members Nadia Malik01 Malik Fawad03

Voice Over IP Presented To:- Engr. Muhammad Iqbal

This presentation helps you to understand the Voice Over IP Telephony. Basic Theme of Our Project

What is VoIP?   “ A Voice over Internet Protocol (VoIP) is a protocol optimized for transmission of voice through the Internet or other packet switched networks ”.   VoIP is often used abstractly to refer to the actual transmission of voice (rather than the protocol implementing it).   VoIP is also known as IP Telephony, Internet telephony, Broadband telephony, Broadband Phone and Voice over Broadband. "VoIP" is sometimes pronounced voyp.

 Providing voice over a data network is challenging because  Voice & Data Traffic have totally different attributes.  Quality of Service (QoS) requirements are different for both. Voice Over Data Network

 Voice requires a continuous stream of unacknowledged data that has traffic demands every 10, 20, and 30 ms, depending on the sampling rate used by the coder/decoder (codec) device. These voice packets are not retransmitted and, therefore, the voice application is susceptible to packet loss, delay, and jitter. Voice Traffic

 Data traffic such as web traffic, on the other hand, tends to have short, high bandwidth bursts with long durations of silence. Data is less susceptible to small amounts of packet loss, delays, and jitter since data traffic typically uses the Transport Control Protocol (TCP) as the transport layer over the network layer internet protocol (IP), which allows for packets to be retransmitted if necessary. Data Traffic

General VoIP Applications   As digital cellular systems entered the market in the 90’s, it was evident that transmitting 64 Kbps raw voice over a wireless link was not practical.   The industry responded to this challenge by developing and standardizing several compression algorithms specifically designed for voice traffic.

Compression Algorithms

  Standardized compression algorithms have since been developed for use in voice applications. The most popular compression algorithms are   G.723   G.729   Uncompressed G.711 The main differences between them are packet size, voice quality, and complexity.

Transport Medium   With VoIP, the transport medium has changed since the voice traffic is carried over IP, requiring   Source   Destination   User Datagram Protocol (UDP)   Real-time Protocol (RTP) headers   In the case of G.729, for example, 58 bytes are needed to transport 20 bytes of compressed Voice.   With overhead like this, it is clear that TCP over Internet Protocol (TCP/IP) is not suited to transporting compressed voice (or very small packets). An alternative protocol is to use RTP header extensions in combination with UDP over IP.

Transport Medium   With overhead like this, it is clear that TCP over Internet Protocol (TCP/IP) is not suited to transporting compressed voice (or very small packets).   An alternative protocol is to use RTP header extensions in combination with UDP over IP.

VoIP Transport

VoIP Signaling Call Control & Network Design   CALL CONTROL Voice is delivered using VoIP protocols, with the Base Transceiver Station (BTS) and Customer Premise Equipment (CPE), or Modem, supporting the low latency and jitter QoS medium.   VoIP SIGNALLING The call signaling is transported using TCP/IP. Therefore, interoperability is between the edge voice node, the call control server, and the termination voice node. elect the call signaling that meets their needs, such as standards-based protocols like Session Initiation Protocol (SIP), Media Gateway Control Protocol (MGCP), or H.323, each of which differs on call setup and service support.

Call Flows & Services

  Once any VoIP call is set up based on the selected call control protocol, the RTP data stream is point-to-point (from source to destination). Depending on the selected capabilities and services required, the call flows can get complex. For example, in the call flow shown, Session Initiation Protocol (SIP) User A making a call to User B1 can be redirected to alternate User B2 in case B1 is busy. This scenario would cover the situation where User B1 is primary and User B2 is backup. In more complex calling class features, application servers would be included to handle services, such as voice mail, conference calling, call blocking, and others.

Standard Voice Services

Billing System   The interface from a customer’s billing system, and operations systems and support, is provided by direct interface to a SoftSwitch (a hosted switch used for VoIP routing). For billing purposes, the SoftSwitch generates Call Detail Records (CDRs) and stores them on the system. A standard billing system can retrieve these records using File Transfer Protocol (FTP) or File Transfer and Access Management (FTAM) Protocol.

Open Source Solutions (OSS)   the Open Source Solutions (OSS) can communicate with the SoftSwitch through a Simple Network Management Protocol (SNMP) interface to provide Fault, Configuration, Accounting, Performance, and Security (FCAPS). Usage records can be generated for billing on a call-by-call basis, or alternate billing plans can be realized, such as calling card or flat rate.

VoIP Features   Voice & Data Quality of Service IP packet types are mapped to one of three levels of traffic prioritization supported in the system 1. Real time/Voice 2. High Priority Data 3. Low Priority Data   Session Rotation Manage multiple users on the system simultaneously by using a session management and rotation feature

VoIP Features   Congestion Control Congestion control is based on three configurable options: Tail Drop is a means of avoiding congestion that treats all traffic equally and does not differentiate between classes of service (CoS). Random Early Detection (RED) is a congestion avoidance mechanism that takes advantage of TCP’s congestion control mechanism. Weighted RED (WRED) generally drops packets selectively based on IP precedence.

Thanks for Your Concentration.