Fundamental of VoIP
Content
Fundamental of VoIP
Muhammad Fayzan Siddiqui Bahria university Karachi Asad Ahmed Singani Bahria university Karachi
Abstract
Voice Over internet protocol is a technique in which ordinary voice signals are converted to packets and send over IP networks. VoIP is very cheap way to transfer voice over a long distance. In 1995 Vocal Tec Company of Israel started Internet telephone software as “Internet Phone Since then in just four year it has grown very rapidly. In modern era the VOIP is being used to rout the international calls. Keywords: VOIP, Voice Over Internet Protocol, Packets, Switching, Gateways.
3. Types of Switching
There are two type of switching
3.1. Circuit Switching:
In circuit switching there is a dedicated communication path between two stations. There are three phases call establishment, maintenance and termination. In circuit switching there must have switching capacity and channel capacity to establish connection.
1. Introduction:
VoIP is sometimes referred to as Internet telephony; it is a method of digitizing voice, into packets. Headers are added to the packets and transmitting those packets over a packet switched IP network. [2]
3.2. Packet Switching:
In packer switching, single node to node link can be shared by many packets over time. Packets are accepted even when network is busy. Packets are handled in two ways virtual circuit and datagram.
“Fig1.1 Rough Conceptualization of VoIP”
2. Internet Protocol:
IP itself is a connectionless protocol that resides at Layer 3 (the network layer) It is unreliable. Other protocols, such as TCP, can sit on top of IP (Layer 4, session) and can add flow control, sequencing, and other features. Protocols split transmitted data into packets, add necessary addressing information to the packets and transmit them and assemble again data in receiving end. [3]
“Fig 1.2 Circuits vs. Packets”
Source: Peter Ingram, “Voice over Internet Protocol- An
introduction”, OfCom
4. Overview of PSTN:
The telephone infrastructure starts with a simple pair of copper wires running to the subscriber, known as a local
loop. The local loop physically connects subscriber telephone to the central office switch or exchange. The communication path between several central office switches is known as a trunk. (e.g. E1, T1) Switches are currently deployed in hierarchies. End office switches (or central office switches) interconnect through trunks to tandem switches (also referred to as Class 4 switches). Higher-layer tandem switches connect local tandem switches. Central office switches often directly connect to each other. Where the direct connections occur between central office switches depends to a great extent on call patterns. If enough traffic occurs between two central office switches, a dedicated circuit is placed between the two switches to offload those calls from the local tandem switches. Some portions of the PSTN use as many as five levels of switching hierarchy. [1]
• •
Call forwarding—Enables a subscriber to forward incoming calls to a different destination. Three-way calling—Enables conference calling. With the deployment of the SS7 network, advanced features can now be carried end to end. A few of the CLASS features are mentioned in the following list:
•
ď€ Display—Displays the calling party's directory number, or Automatic Number Identification (ANI). Call blocking—Blocks specific incoming numbers so that callers are greeted with a message saying the call is not accepted. Calling line ID blocking—Blocks the outgoing directory number from being shown on someone else's display. (This does not work when calling 800-numbers or certain other numbers.) Automatic callback—enables you to put a hold on the last number dialed if a busy signal is received, and then place the call after the line is free. Call return (*69)—Enables users to quickly reply to missed calls. [1]
•
•
• •
“Fig1.3. PSTN—Traditional VoIP” Source: Peter Ingram, “Voice over Internet Protocol- An introduction”, OfCom
7. Drawbacks to the PSTN:
Although the PSTN is effective and does a good job at what it was built to do (that is, switch voice calls), many business drivers are striving to change it to a new network, whereby voice is an application on top of a data network. This is happening for several reasons: • Data has overtaken voice as the primary traffic on many networks built for voice. • • The PSTN cannot create and deploy features quickly enough. Data/Voice/Video (D/V/V) cannot converge on the PSTN as currently built. The architecture built for voice is not flexible enough to carry data. [1]
5. PSTN Signaling:
Generally, two types of signaling methods run over various transmission media. The signaling methods are broken into the two groups: • User-to-network signaling— this is how an end user communicates with the PSTN. • Network-to-network signaling— this is generally how the switches in the PSTN intercommunicate. Network-to-network signaling also uses an out-of-band signaling method known as Signaling System 7 (SS7) (or C7 in European countries). [1]
•
6. PSTN Services and Applications:
The popular custom calling features commonly found in the PSTN today: • Call waiting—Notifies customers who already placed a call that they are receiving an incoming call.
8. VoIP Technical Description:
Figure 1.5 shows a simplified block diagram of VoIP operation from an analog signal deriving from a standard telephone, which is digitized and transmitted over the Internet via a conversion device. Then, at the
distant end, it is converted back to analog telephony using a similar device suitable for input to a standard telephone. The gateway is placed between the voice codec and the digital data transport circuit. An identical device will also be found at the far end of the link. This equipment carries out the signaling role on a telephone call among other functions. Moving from left to right in Figure 1.5, we have the spurty analog signal deriving from a standard telephone set. The signal is then converted to a digital counterpart using one of seven or so codecs [coder-decoder(s)] that the VoIP system designer has to select from. The binary output of the codec is then applied to a conversion device (i.e., a “packetizer”) that loads these binary 1s and 0s into an IP payload of from 20 to 40 octets in length. The output of the converter consists of IP packets1 that are transmitted on the web or other data circuit for delivery to the distant end. At the far end the IP packets or frames are input to a converter (i.e., depacketizer) that strips off the IP header, stores the payload, and then releases it in a constant bit stream to a codec (i.e., a D–A converter). Of course this codec must be compatible with its nearend counterpart. The codec converts the digital bit stream back to an analog signal that is then input to a standard telephone subset. [2]
“Fig1.5. PC-to-PC VoIP” Source: Peter Ingram, “Voice over Internet Protocol- An
introduction”, OfCom
9.2. Phone-to-Phone:
If the traditional telephone is use it will connect to an Analogue Telephone Adapter (ATA) or use an IP phone. The called party may be another VoIP user or, via a gateway, a traditional PSTN customer. [6]
“Fig1.6. Phone-to-Phone VoIP”
Source: Peter Ingram, “Voice over Internet Protocol- An
introduction”, OfCom
10. VoIP in PSTN:
Many traditional PSTN calls are carried as VoIP in part, for efficiency reasons they may travel with other IP traffic. These calls are different from other VoIP calls. It is invisible to end customer. The private IP network is used instead of internet cloud to control quality. [6]
“Fig1.4. Conceptualization of VoIP”
9. VoIP Calls:
Types of VoIP calls
9.1. PC-to-PC:
Software install on the machine (PC) allows voice “calls” from one PC to another. It enables voice to convert into IP packets at PC. [6]
“Fig1.7. VoIP in the PSTN”
Source: Peter Ingram, “Voice over Internet Protocol- An
introduction”, OfCom
11. Function of Gateway:
Gateway
Internet
PSTN
“Fig1.8. Concept of Gateway” Source: Roger L. Freemen, “Fundamentals Telecommunication, IEEE press,
of
Gateway is a server; it may also be called a media gateway. It is the bridge of Internet and PSTN to connect. For PSTN the Gateway has the interface of telephone net. (e.g. E1 (2048 kb/s)), and for Internet side, Gateway has network interface (E.g. Ethernet interface). Through the cooperation of hardware and software, realize between" Circuit switch" and "Packet switch" voice coding form change mutually. Media gateways are part of the physical transport layer. They are controlled by a call control function housed in a media gateway controller. It supports several types of access networks including media such as copper, fiber, radio, and CATV cable. [2]
MGCP is an implementation of the Media Gateway Control Protocols architecture for controlling Media Gateways on Internet protocols networks and the public switched telephone network. MGCP is a signaling and call control protocol used within VoIP systems that typically interoperate with the public switched telephone network. Media Gateway Control Protocol is used between elements of a decomposed multimedia gateway which consists of a Call Agent, which contains the call control "intelligence", and a media gateway which contains the media functions, e.g., conversion from TDM voice to Voice over IP. [4]
12.2. Multipurpose Internet Mail Extensions:
MIME is a specification for enhancing the capabilities of standard Internet electronic mail. It offers a simple standardized way to represent and encode a wide variety of media types for transmission via Internet mail. The MIME standard contains the following types of messages: • Text messages in US-ASCII • Character sets other than US-ASCII. • Multi-media: Image, Audio, and Video messages. • Multiple objects in a single message. • Multi-font messages. • Messages of unlimited length. • Binary files. [4]
12. VoIP Protocols:
Protocols are set of rules determining the format and transmission of data, so protocols will determine that what will be the format of the data and how will be data transmitted. [4] The most common VoIP protocols used are: 1. Megaco H.248 2. MGCP 3. Media Gateway Control Protocol 4. MIME 5. RVP over IP 6. Remote Voice Protocol Over IP Specification 7. SAPv2 8. Session Announcement Protocol 9. SDP 10. Session Description Protocol 11. SGCP 12. Simple Gateway Control Protocol 13. SIP 14. Session Initiation Protocol 15. Skinny 16. Skinny Client Control Protocol (SCCP) 17. Gateway Control Protocol
12.3. Remote VoIP:
Remote Voice Protocol (RVP) is MCK (McKesson Corporation) Communications' protocol for transporting digital telephony sessions over packet or circuit based data networks. The protocol is used primarily in Mack’s Extender product family, which extends PBX services over Wide Area Networks (WANs). RVP/IP uses TCP to transport signaling and control data, and UDP to transport voice data. Control and signaling packets carried over TCP are encapsulated using the following format; a header followed by signaling or control messages: [4]
12.1. Media Gateway Control Protocol:
“Fig1.9. RVP over IP packet structure”
12.4. Session Initiation Protocol (SIP):
Session Initiation Protocol (SIP) is an application layer control simple signaling protocol for VoIP implementations using the Redirect Mode. SIP is a textual client-server base protocol and provides the necessary protocol mechanisms so that the end user systems and proxy servers can provide different services. Call forwarding in several scenarios: no answer, busy, unconditional, address manipulations (as 700, 800, and 900- type calls). Called and calling number identification Personal mobility, caller and called authentication, invitations to multicast conference, basic Automatic Call Distribution (ACD). SIP supports five distinct features of establishing and terminating multimedia communications: User location, User capabilities, User availability, and Call setup, Call handling. [4]
•
Benefits of Voice over IP (VoIP), including cost savings, single infrastructure savings, and new applications Using a packet telephony call center versus a circuit-switched call center Service provider applications prepaid calling card
• •
•
Service provider enhanced services (such as Internet call waiting and click to talk) [2]
14. VoIP in Pakistan:
Voice over IP is nothing new in Pakistan from - VOIP has played an important role for people to stay close to those who are far away from them. In the current times when life is becoming more mobile, keeping oneself attached to PC, is very cumbersome and the increasing power crises in the country makes it more difficult to use it. Here comes VOIP on mobile, supported by SKYPE and Fring. FRING is free software which supports to use SKYPE with ease, just register once and you can talk away any where you are - all you need is a supporting cell phone and data cable connection. Fring is versatile, it’s not limited only to high end phones, from Nokia 1680 to Nokia N97, from Symbian to UIQ to Windows Mobile and iPhone, and it works on every kind of cell phones effortlessly. There are a lot of advantages of VOIP, as international phone calls from Pakistan is almost free but here is a need to pay charges for calling to Pakistan. With Mobile VOIP, a businessman can have his worldwide associates call him over on his SKYPE id whether he is in office or lounging at home, a family can talk away via speaker phone with their loved ones abroad even if the lights are out, those concerned with security can sleep easy, it’s rumored that only NASA has the technology to listen over SKYPE and that too will take much of their precious super computer time to hack into the conversation. [5]
12.5. T.38
The T.38 IP-based fax service maps the T.30 fax protocol onto an IP network. Both fax and voiced data are managed through a single gateway. T.38 uses 2 protocols, one for UDP packets and one for TCP packets. Data is encoded using ASN.1 (Abstract Syntax Notation One) to ensure a standard technique It allows users to transfer facsimile documents between 2 standard fax terminals over the Internet or other network using IP protocols. [4]
12.6 The Real-time Transport:
The Real-time Transport (RTP) Protocol provides endto-end network transport functions suitable for applications transmitting real-time data such as audio, video or simulation data, over multicast (the delivery of information to a group of destinations simultaneously) or unicast (the sending of information packets to a single network destination.) network services. RTP does not address resource reservation and does not guarantee quality-of-service for real-time services. The data transport is augmented by a control protocol “Real Time Transport Control Protocol” to allow monitoring of the data delivery. [4]
15. REFRENCES
[1] Jonathan Davidson, James peter “Voice over IP Fundamentals”, Cisco press release, USA, 2000 [2] Roger L. Freemen, “Fundamentals of Telecommunication, IEEE press, Canada, 2005 [3] www.wikepedia.com [4] www.protocols.com [5] www.propakistan.com [6] Peter Ingram, “Voice over Internet Protocol- An introduction”, OfCom, 18th January 2005
13. Voice over IP Benefits and Applications:
Muhammad Fayzan Siddiqui Bahria university Karachi Asad Ahmed Singani Bahria university Karachi
Abstract
Voice Over internet protocol is a technique in which ordinary voice signals are converted to packets and send over IP networks. VoIP is very cheap way to transfer voice over a long distance. In 1995 Vocal Tec Company of Israel started Internet telephone software as “Internet Phone Since then in just four year it has grown very rapidly. In modern era the VOIP is being used to rout the international calls. Keywords: VOIP, Voice Over Internet Protocol, Packets, Switching, Gateways.
3. Types of Switching
There are two type of switching
3.1. Circuit Switching:
In circuit switching there is a dedicated communication path between two stations. There are three phases call establishment, maintenance and termination. In circuit switching there must have switching capacity and channel capacity to establish connection.
1. Introduction:
VoIP is sometimes referred to as Internet telephony; it is a method of digitizing voice, into packets. Headers are added to the packets and transmitting those packets over a packet switched IP network. [2]
3.2. Packet Switching:
In packer switching, single node to node link can be shared by many packets over time. Packets are accepted even when network is busy. Packets are handled in two ways virtual circuit and datagram.
“Fig1.1 Rough Conceptualization of VoIP”
2. Internet Protocol:
IP itself is a connectionless protocol that resides at Layer 3 (the network layer) It is unreliable. Other protocols, such as TCP, can sit on top of IP (Layer 4, session) and can add flow control, sequencing, and other features. Protocols split transmitted data into packets, add necessary addressing information to the packets and transmit them and assemble again data in receiving end. [3]
“Fig 1.2 Circuits vs. Packets”
Source: Peter Ingram, “Voice over Internet Protocol- An
introduction”, OfCom
4. Overview of PSTN:
The telephone infrastructure starts with a simple pair of copper wires running to the subscriber, known as a local
loop. The local loop physically connects subscriber telephone to the central office switch or exchange. The communication path between several central office switches is known as a trunk. (e.g. E1, T1) Switches are currently deployed in hierarchies. End office switches (or central office switches) interconnect through trunks to tandem switches (also referred to as Class 4 switches). Higher-layer tandem switches connect local tandem switches. Central office switches often directly connect to each other. Where the direct connections occur between central office switches depends to a great extent on call patterns. If enough traffic occurs between two central office switches, a dedicated circuit is placed between the two switches to offload those calls from the local tandem switches. Some portions of the PSTN use as many as five levels of switching hierarchy. [1]
• •
Call forwarding—Enables a subscriber to forward incoming calls to a different destination. Three-way calling—Enables conference calling. With the deployment of the SS7 network, advanced features can now be carried end to end. A few of the CLASS features are mentioned in the following list:
•
ď€ Display—Displays the calling party's directory number, or Automatic Number Identification (ANI). Call blocking—Blocks specific incoming numbers so that callers are greeted with a message saying the call is not accepted. Calling line ID blocking—Blocks the outgoing directory number from being shown on someone else's display. (This does not work when calling 800-numbers or certain other numbers.) Automatic callback—enables you to put a hold on the last number dialed if a busy signal is received, and then place the call after the line is free. Call return (*69)—Enables users to quickly reply to missed calls. [1]
•
•
• •
“Fig1.3. PSTN—Traditional VoIP” Source: Peter Ingram, “Voice over Internet Protocol- An introduction”, OfCom
7. Drawbacks to the PSTN:
Although the PSTN is effective and does a good job at what it was built to do (that is, switch voice calls), many business drivers are striving to change it to a new network, whereby voice is an application on top of a data network. This is happening for several reasons: • Data has overtaken voice as the primary traffic on many networks built for voice. • • The PSTN cannot create and deploy features quickly enough. Data/Voice/Video (D/V/V) cannot converge on the PSTN as currently built. The architecture built for voice is not flexible enough to carry data. [1]
5. PSTN Signaling:
Generally, two types of signaling methods run over various transmission media. The signaling methods are broken into the two groups: • User-to-network signaling— this is how an end user communicates with the PSTN. • Network-to-network signaling— this is generally how the switches in the PSTN intercommunicate. Network-to-network signaling also uses an out-of-band signaling method known as Signaling System 7 (SS7) (or C7 in European countries). [1]
•
6. PSTN Services and Applications:
The popular custom calling features commonly found in the PSTN today: • Call waiting—Notifies customers who already placed a call that they are receiving an incoming call.
8. VoIP Technical Description:
Figure 1.5 shows a simplified block diagram of VoIP operation from an analog signal deriving from a standard telephone, which is digitized and transmitted over the Internet via a conversion device. Then, at the
distant end, it is converted back to analog telephony using a similar device suitable for input to a standard telephone. The gateway is placed between the voice codec and the digital data transport circuit. An identical device will also be found at the far end of the link. This equipment carries out the signaling role on a telephone call among other functions. Moving from left to right in Figure 1.5, we have the spurty analog signal deriving from a standard telephone set. The signal is then converted to a digital counterpart using one of seven or so codecs [coder-decoder(s)] that the VoIP system designer has to select from. The binary output of the codec is then applied to a conversion device (i.e., a “packetizer”) that loads these binary 1s and 0s into an IP payload of from 20 to 40 octets in length. The output of the converter consists of IP packets1 that are transmitted on the web or other data circuit for delivery to the distant end. At the far end the IP packets or frames are input to a converter (i.e., depacketizer) that strips off the IP header, stores the payload, and then releases it in a constant bit stream to a codec (i.e., a D–A converter). Of course this codec must be compatible with its nearend counterpart. The codec converts the digital bit stream back to an analog signal that is then input to a standard telephone subset. [2]
“Fig1.5. PC-to-PC VoIP” Source: Peter Ingram, “Voice over Internet Protocol- An
introduction”, OfCom
9.2. Phone-to-Phone:
If the traditional telephone is use it will connect to an Analogue Telephone Adapter (ATA) or use an IP phone. The called party may be another VoIP user or, via a gateway, a traditional PSTN customer. [6]
“Fig1.6. Phone-to-Phone VoIP”
Source: Peter Ingram, “Voice over Internet Protocol- An
introduction”, OfCom
10. VoIP in PSTN:
Many traditional PSTN calls are carried as VoIP in part, for efficiency reasons they may travel with other IP traffic. These calls are different from other VoIP calls. It is invisible to end customer. The private IP network is used instead of internet cloud to control quality. [6]
“Fig1.4. Conceptualization of VoIP”
9. VoIP Calls:
Types of VoIP calls
9.1. PC-to-PC:
Software install on the machine (PC) allows voice “calls” from one PC to another. It enables voice to convert into IP packets at PC. [6]
“Fig1.7. VoIP in the PSTN”
Source: Peter Ingram, “Voice over Internet Protocol- An
introduction”, OfCom
11. Function of Gateway:
Gateway
Internet
PSTN
“Fig1.8. Concept of Gateway” Source: Roger L. Freemen, “Fundamentals Telecommunication, IEEE press,
of
Gateway is a server; it may also be called a media gateway. It is the bridge of Internet and PSTN to connect. For PSTN the Gateway has the interface of telephone net. (e.g. E1 (2048 kb/s)), and for Internet side, Gateway has network interface (E.g. Ethernet interface). Through the cooperation of hardware and software, realize between" Circuit switch" and "Packet switch" voice coding form change mutually. Media gateways are part of the physical transport layer. They are controlled by a call control function housed in a media gateway controller. It supports several types of access networks including media such as copper, fiber, radio, and CATV cable. [2]
MGCP is an implementation of the Media Gateway Control Protocols architecture for controlling Media Gateways on Internet protocols networks and the public switched telephone network. MGCP is a signaling and call control protocol used within VoIP systems that typically interoperate with the public switched telephone network. Media Gateway Control Protocol is used between elements of a decomposed multimedia gateway which consists of a Call Agent, which contains the call control "intelligence", and a media gateway which contains the media functions, e.g., conversion from TDM voice to Voice over IP. [4]
12.2. Multipurpose Internet Mail Extensions:
MIME is a specification for enhancing the capabilities of standard Internet electronic mail. It offers a simple standardized way to represent and encode a wide variety of media types for transmission via Internet mail. The MIME standard contains the following types of messages: • Text messages in US-ASCII • Character sets other than US-ASCII. • Multi-media: Image, Audio, and Video messages. • Multiple objects in a single message. • Multi-font messages. • Messages of unlimited length. • Binary files. [4]
12. VoIP Protocols:
Protocols are set of rules determining the format and transmission of data, so protocols will determine that what will be the format of the data and how will be data transmitted. [4] The most common VoIP protocols used are: 1. Megaco H.248 2. MGCP 3. Media Gateway Control Protocol 4. MIME 5. RVP over IP 6. Remote Voice Protocol Over IP Specification 7. SAPv2 8. Session Announcement Protocol 9. SDP 10. Session Description Protocol 11. SGCP 12. Simple Gateway Control Protocol 13. SIP 14. Session Initiation Protocol 15. Skinny 16. Skinny Client Control Protocol (SCCP) 17. Gateway Control Protocol
12.3. Remote VoIP:
Remote Voice Protocol (RVP) is MCK (McKesson Corporation) Communications' protocol for transporting digital telephony sessions over packet or circuit based data networks. The protocol is used primarily in Mack’s Extender product family, which extends PBX services over Wide Area Networks (WANs). RVP/IP uses TCP to transport signaling and control data, and UDP to transport voice data. Control and signaling packets carried over TCP are encapsulated using the following format; a header followed by signaling or control messages: [4]
12.1. Media Gateway Control Protocol:
“Fig1.9. RVP over IP packet structure”
12.4. Session Initiation Protocol (SIP):
Session Initiation Protocol (SIP) is an application layer control simple signaling protocol for VoIP implementations using the Redirect Mode. SIP is a textual client-server base protocol and provides the necessary protocol mechanisms so that the end user systems and proxy servers can provide different services. Call forwarding in several scenarios: no answer, busy, unconditional, address manipulations (as 700, 800, and 900- type calls). Called and calling number identification Personal mobility, caller and called authentication, invitations to multicast conference, basic Automatic Call Distribution (ACD). SIP supports five distinct features of establishing and terminating multimedia communications: User location, User capabilities, User availability, and Call setup, Call handling. [4]
•
Benefits of Voice over IP (VoIP), including cost savings, single infrastructure savings, and new applications Using a packet telephony call center versus a circuit-switched call center Service provider applications prepaid calling card
• •
•
Service provider enhanced services (such as Internet call waiting and click to talk) [2]
14. VoIP in Pakistan:
Voice over IP is nothing new in Pakistan from - VOIP has played an important role for people to stay close to those who are far away from them. In the current times when life is becoming more mobile, keeping oneself attached to PC, is very cumbersome and the increasing power crises in the country makes it more difficult to use it. Here comes VOIP on mobile, supported by SKYPE and Fring. FRING is free software which supports to use SKYPE with ease, just register once and you can talk away any where you are - all you need is a supporting cell phone and data cable connection. Fring is versatile, it’s not limited only to high end phones, from Nokia 1680 to Nokia N97, from Symbian to UIQ to Windows Mobile and iPhone, and it works on every kind of cell phones effortlessly. There are a lot of advantages of VOIP, as international phone calls from Pakistan is almost free but here is a need to pay charges for calling to Pakistan. With Mobile VOIP, a businessman can have his worldwide associates call him over on his SKYPE id whether he is in office or lounging at home, a family can talk away via speaker phone with their loved ones abroad even if the lights are out, those concerned with security can sleep easy, it’s rumored that only NASA has the technology to listen over SKYPE and that too will take much of their precious super computer time to hack into the conversation. [5]
12.5. T.38
The T.38 IP-based fax service maps the T.30 fax protocol onto an IP network. Both fax and voiced data are managed through a single gateway. T.38 uses 2 protocols, one for UDP packets and one for TCP packets. Data is encoded using ASN.1 (Abstract Syntax Notation One) to ensure a standard technique It allows users to transfer facsimile documents between 2 standard fax terminals over the Internet or other network using IP protocols. [4]
12.6 The Real-time Transport:
The Real-time Transport (RTP) Protocol provides endto-end network transport functions suitable for applications transmitting real-time data such as audio, video or simulation data, over multicast (the delivery of information to a group of destinations simultaneously) or unicast (the sending of information packets to a single network destination.) network services. RTP does not address resource reservation and does not guarantee quality-of-service for real-time services. The data transport is augmented by a control protocol “Real Time Transport Control Protocol” to allow monitoring of the data delivery. [4]
15. REFRENCES
[1] Jonathan Davidson, James peter “Voice over IP Fundamentals”, Cisco press release, USA, 2000 [2] Roger L. Freemen, “Fundamentals of Telecommunication, IEEE press, Canada, 2005 [3] www.wikepedia.com [4] www.protocols.com [5] www.propakistan.com [6] Peter Ingram, “Voice over Internet Protocol- An introduction”, OfCom, 18th January 2005
13. Voice over IP Benefits and Applications:
