1.

Define the VoIP concept.

VoIP (voice over IP) is an IP telephony term for a set of facilities used to manage the delivery of voice information over the Internet. Voice over Internet Protocol (Voice over IP, VoIP) is one of a family of internet technologies, communication protocols, and transmission technologies for delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. Other terms frequently encountered and often used synonymously with VoIP are IP telephony, Internet telephony, voice over broadband (VoBB), broadband telephony, and broadband phone. 2. Analyze VoIP vs. classical telephony.

VoIP involves sending voice information in digital form in discrete packets rather than by using the traditional circuit-committed protocols of the public switched telephone network. A major advantage of VoIP and Internet telephony is that it avoids the tolls charged by ordinary telephone service. 3. Compare Packet Switching and Circuit Switching

The old telephone system (PSTN) uses circuit switching to transmit voice data whereas VoIP uses packetswitching to do so. In circuit-switching, this path is decided upon before the data transmission starts. The system decides on which route to follow, based on a resource-optimizing algorithm, and transmission goes according to the path. For the whole length of the communication session between the two communicating bodies, the route is dedicated and exclusive, and released only when the session terminates. In packet-switching, the packets are sent towards the destination irrespective of each other. Each packet has to find its own route to the destination. There is no predetermined path; the decision as to which node to hop to in the next step is taken only when a node is reached. Each packet finds its way using the information it carries, such as the source and destination IP addresses. 4. VoIP elements, network structure and interfaces

Basic components of an IP telephone system include IP telephones, the Internet backbone, and signaling servers. 5. Shortly describe H.323 protocol

H.323 is a protocol standard for multimedia communications. H.323 was designed to support real-time transfer of audio and video data over packet networks like IP. The standard involves several different protocols covering specific aspects of Internet telephony. 6. Shortly describe SIP

The Session Initiation Protocol (SIP) is one of the most important VoIP signaling protocols operating in the application layer in the five-layer TCP/IP model. SIP can perform both unicast and multicast sessions and supports user mobility. SIP handles signals and identifies user location, call setup, call termination, and busy signals. SIP can use multicast to support conference calls and uses the Session Description Protocol (SDP) to negotiate parameters.

7.

H.323 architecture and specific elements

The H.323 system defines several network elements that work together in order to deliver rich multimedia communication capabilities. Those elements are Terminals, Multipoint Control Units (MCUs), Gateways, Gatekeepers, and Border Elements. Collectively, terminals, multipoint control units and gateways are often referred to as endpoints. While not all elements are required, at least two terminals are required in order to enable communication between two people. In most H.323 deployments, a gatekeeper is employed in order to, among other things, facilitate address resolution. 8. SIP architecture and specific entities

SIPs basic architecture is client / server in nature. The main entities in SIP are the User Agent, the SIP Proxy Server, the SIP Redirect Server and the Registrar. 9. Provide H.323 vs. SIP compare H.323 Architecture
H.323 covers almost every service, such as capability exchange, conference control, basic signaling,QoS, registration, service discovery, and so on. Terminal/Gateway Gatekeeper

SIP
SIP is modular because it covers basic call signaling, user location, and registration. Other features are in other separate orthogonal protocols UA Servers SI SDP No No No No SIP inherently supports wide area addressing. When multiple servers are involved in setting up a call, SIP uses a loop detection algorithm similar to the one used in BGP, which can be done in a stateless manner, thus avoiding scalability issues. Call control can be implemented in a call stateless manner. SIP supports n to n scaling between UAs and servers. SIP takes less CPU cycles to generate signaling messages; therefore a server could theoretically handle more transactions. SIP supports caller and callee authentication via HTTP mechanisms. Cryptographically secure authentication and encryption is supported hop-by-hop via SSL/TSL, but SIP could use any transport-layer or HTTP-like security mechanism, such as SSH or S-HTTP. SIP supports any IANA-registered codec (as a legacy feature) or other codec whose name is mutually agreed upon. Payload types can be specified statically or dynamically Reliable or unreliable , e.g., TCP or UDP. Most SIP entities use an unreliable transport for signaling.

Components Protocols Message Waiting Indication Name Identification Call Offer Call Intrusion Large Number of Domains

RAS/Q.931 H.245 Yes Yes Yes Yes The initial intent of H.323 was for the support of LANs, so it was not inherently designed for wide area addressing. The concept of a zone was added to accommodate wide area addressing. Procedures are defined for user location across zones for email names. H.323 call control can be implemented in a stateless manner. A gateway can use messages defined in H.225 to assist the gatekeeper in performing load balancing across gateways. Defines security mechanisms and negotiation facilities via H.235 , can also use SSL for transport-layer security.

Large Number of Calls

Security

Codecs

H.323 supports any codec, standardized or proprietary, not just ITU-T codecs.

Transport Protocol

Reliable or unreliable , e.g., TCP or UDP. Most H.323 entities use a reliable transport for signaling.

Addressing Video and Data Conferencing

Flexible addressing mechanisms, including URLs and E.164 numbers. H.323 fully supports video and data conferencing. Procedures are in place to provide control for the conference as well as lip synchronization of audio and video streams.

SIP only understands URL-style addresses. SIP has limited support for video and no support for data conferencing protocols like T.120. SIP has no protocol to control the conference and there is no mechanism within SIP for lip synchronization.

10. Describe SGCP Simple Gateway Control Protocol (SGCP) is a communications protocol used within a Voice over Internet Protocol (VoIP) system. It has been superseded by MGCP, an implementation of the Media Gateway Control Protocol architecture. The Simple Gateway Control Protocol was published in 1998 by Christian Huitema and Mauricio Arango,[1] as part of the development of the "Call Agent Architecture" at Telcordia. In this architecture a central server, the "Call Agent", controls "media gateways" and receives telephony signaling requests through a "signalling gateway". Later implementation of the architecture refer to the "Call Agent" as a "Softswitch". SGCP was intended to be compatible with the Session Initiation Protocol (SIP), enabling the Call Agent to relay calls between a Voice over IP network using SIP and a traditional telephone network. The SGCP commands are encoded with a syntax somewhat comparable to the SIP or HTTP headers. They carry a payload describing the voice over IP media stream. This payload is encoded using the same "session description protocol" (SDP) as SIP. 11. Describe MGCP MGCP is an implementation of the Media Gateway Control Protocol architecture[1] for controlling media gateways on Internet Protocol (IP) networks and the public switched telephone network (PSTN). The general base architecture and programming interface is described in RFC 2805 and the current specific MGCP definition is RFC 3435 (obsoleted RFC 2705). It is a successor to the Simple Gateway Control Protocol (SGCP). MGCP is a signalling and call control protocol used within Voice over IP (VoIP) systems that typically interoperate with the public switched telephone network (PSTN). As such it implements a PSTN-over-IP model with the power of the network residing in a call control center (softswitch, similar to the central office of the PSTN) and the endpoints being "low-intelligence" devices, mostly simply executing control commands. The protocol represents a decomposition of other VoIP models, such as H.323, in which the media gateways (e.g., H.323's gatekeeper) have higher levels of signalling intelligence. MGCP uses the Session Description Protocol (SDP) for specifying and negotiating the media streams to be transmitted in a call session and the Real-time Transport Protocol (RTP) for framing of the media streams 12. Describe MEGACO Megaco (officially H.248) is an implementation of the Media Gateway Control Protocol architecture[1] for controlling media gateways in Internet Protocol (IP) networks and the public switched telephone network (PSTN). The general base architecture for the Media Gateway Control Protocol and programming interface was originally described in RFC 2805 and the current specific Megaco definition is ITU-T Recommendation H.248.1.

Megaco defines the protocol for media gateway controllers to control media gateways for the support of multimedia streams across computer networks. It is typically used for providing Voice over Internet Protocol (VoIP) services (voice and fax) between IP networks and the PSTN, or entirely within IP networks. 13. VoIP packing

14. Overheads contribution in VoIP services

15. VoIP limitations The Internet can throw a variety of glitches at a VoIP signal that can often be serious enough to make communication impossible. These problems include: Packet Loss: sometimes when networks are very busy bundles of data (packets) arent successfully received. Packet Error: some of the data within one or more packets are degraded. Phase Jitter: intermittent shortening or lengthening of the signal disrupts data. Packet Reordering: packets dont arrive in the proper order.

Over the Internet, speech is divided into about 10 data packets per second. Very little interference quickly renders speech unintelligible. The Internet is a largely unpredictable place. Many things can bog it down and slow down traffic. Such routine delays can also make VOIP unusable. 16. QoS rules impacting on VoIP services A VoIP connection has several QoS factors: Packet loss is accepted to a certain extent. Packet delay is normally unacceptable. - Jitter, as the variation in packet arrival time, is not acceptable after a certain limit. 17. Define the main QoS goals and parameters There are many data communication traffic parameters that may be considered QoS parameters. The list below includes some of the more common QoS parameters. Latency Jitter Throughput Reliability Startup delay Termination delay 18. QoS latency, problem and solutions

The latency is also known as delay. It is not a specific problem of the connectionless networks and therefore a VoIP problem. It is a general problem of telecommunication networks. For example, the latency in satellite connections is very high because the distances are big. Latency is technically the amount of time it takes a packet to travel from source to destination. Together, latency and bandwidth define the speed and capacity of a network. There is not a simple solution for this problem. It often depends on the devices that route the packets, in other words, in the same network. One solution could be to reserve a bandwidth from the origin to the final end or to mark the packets with TOS values. This way, routers can know that the traffic is in real time. Nowadays this is not very useful because we can control the network itself. 19. QoS, resource reservation (RSVP) The Resource Reservation Protocol (RSVP) is a Transport Layer protocol designed to reserve resources across a network for an integrated services Internet. RSVP operates over an IPv4 or IPv6 Internet Layer and provides receiver-initiated setup of resource reservations for multicast or unicast data flows with scaling and robustness. It does not transport application data but is similar to a control protocol, like ICMP, IGMP. RSVP is described in RFC 2205. RSVP can be used by either hosts or routers to request or deliver specific levels of quality of service (QoS) for application data streams or flows. RSVP defines how applications place reservations and how they can relinquish the reserved resources once the need for them has ended. RSVP operation will generally result in resources being reserved in each node along a path. RSVP is not itself a routing protocol and was designed to inter-operate with current and future routing protocols.

20. Scalable QoS guarantees (DiffServ) Differentiated Services or DiffServ is a computer networking architecture that specifies a simple, scalable and coarse-grained mechanism for classifying, managing network traffic and providing Quality of Service (QoS) on modern IP networks. DiffServ can, for example, be used to provide low-latency to critical network traffic such as voice or streaming media while providing simple best-effort service to non-critical services such as web traffic or file transfers. DiffServ uses the 6-bit Differentiated Services Code Point (DSCP) field in the IP header for packet classification purposes. DSCP replaces the outdated Type of Service field. 21. Describe MPLS Multiprotocol Label Switching (MPLS) is a mechanism in high-performance telecommunications networks which directs and carries data from one network node to the next with the help of labels. MPLS makes it easy to create "virtual links" between distant nodes. It can encapsulate packets of various network protocols. MPLS is a highly scalable, protocol agnostic, data-carrying mechanism. In an MPLS network, data packets are assigned labels. Packet-forwarding decisions are made solely on the contents of this label, without the need to examine the packet itself. This allows one to create end-to-end circuits across any type of transport medium, using any protocol. The primary benefit is to eliminate dependence on a particular Data Link Layer technology, such as ATM, frame relay, SONET or Ethernet, and eliminate the need for

multiple Layer 2 networks to satisfy different types of traffic. MPLS belongs to the family of packetswitched networks. 22. VoIP codecs Codec
G.711 G.722 G.723.1 G.726 G.729 GSM iLBC Speex

Bandwidth/kbps
64 48/56/64 5.3/6.3 16/24/32/40 8 13 15 2.15 / 44

Comments
Delivers precise speech transmission. Very low processor requirements. Needs at least 128 kbps for two-way. Adapts to varying compressions and bandwidth is conserved with network congestion. High compression with high quality audio. Can use with dial-up. Lot of processor power. An improved version of G.721 and G.723 (different from G.723.1) Excellent bandwidth utilization. Error tolerant. License required. High compression ratio. Free and available in many hardware and software platforms. Same encoding is used in GSM cellphones Robust to packet loss. Free Minimizes bandwidth usage by using variable bit rate.

23. Real time data transport principle (RTP/RTCP) The Real-time Transport Protocol (RTP) defines a standardized packet format for delivering audio and video over IP networks. RTP is used extensively in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications and web-based push-to-talk features. RTP is used in conjunction with the RTP Control Protocol (RTCP). 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. When both protocols are used in conjunction, RTP is originated and received on even port numbers and the associated RTCP communication uses the next higher odd port number. It is one of the technical foundations of Voice over IP and in this context is often used in conjunction with a signaling protocol which assists in setting up connections across the network. RTP was developed by the Audio-Video Transport Working Group of the Internet Engineering Task Force (IETF) and first published in 1996 as RFC 1889, superseded by RFC 3550 in 2003. 24. RTP packet structure The structure of a RTP packet is shown below.

25. Explain Sender Report / Receiver Report in RTCP A Sender Report message consists of the header, the sender information block, a variable number of receiver report blocks, and potentially a profile-specific extensions. Receiver Reports are structured in the same way as Sender Reports. Of course, they include no sender information block, and the packet type code is 201. 26. Jitter Estimation in real-time traffic

27. Content Distribution Networks A content delivery network or content distribution network (CDN) is a system of computers containing copies of data placed at various nodes of a network. When properly designed and implemented, a CDN can improve access to the data it caches by increasing access bandwidth and redundancy and reducing access latency. Data content types often cached in CDNs include web objects, downloadable objects (media files, software, documents), applications, realtime media streams, and database queries. 28. Stream Control Transmission Protocol, SCTP (futures, packet structure) In computer networking, the Stream Control Transmission Protocol (SCTP) is a Transport Layer protocol, serving in a similar role to the popular protocols Transmission Control Protocol (TCP) and User Datagram Protocol (UDP). It provides some of the same service features of both: it is message-oriented like UDP and ensures reliable, in-sequence transport of messages with congestion control like TCP. The protocol was defined by the IETF Signaling Transport (SIGTRAN) working group in 2000, and is maintained by the IETF Transport Area (TSVWG) working group. RFC 4960 defines the protocol. RFC 3286 provides an introduction. In the absence of native SCTP support in operating systems it is possible to tunnel SCTP over UDP,[1] as well as mapping TCP API calls to SCTP ones.[2] SCTP packets have a simpler basic structure than TCP packets. Each consists of two basic sections: - The common header, which occupies the first 12 bytes and is highlighted in blue, and - The data chunks, which occupy the remaining portion of the packet. The first chunk is highlighted in green, and the last of N chunks (Chunk N) is highlighted in red. 29. SCTP chunks management SCTP is designed to transport PSTN signaling messages over IP networks, but is capable of broader applications. SCTP is a reliable transport protocol operating on top of a connectionless packet network such as IP. The design of SCTP includes appropriate congestion avoidance behavior and resistance to flooding and masquerade attacks.