VOICE OVER PACKETS IN A PACKET NETWORK

15.3 DRAWBACKS AND CHALLENGES FOR TRANSMITTING VOICE ON DATA  Impact of errored frames packets  Variation of packet arrival time, jitter buffering  Silence suppression  Prioritizi

Voice-Over Packets in a Packet Network

Lecturer: M.S: Bùi Thư Cao

Student:

1 Nguyễn Hoàng Tuấn Phương 07711271

VOICE-OVER PACKETS IN A PACKET NETWORK

15.1 AN OVERVIEW OF THE CONCEPT

Digitize the voice and break up the resulting serial bit stream into packets of some length A header is added to each packet, and the packet can now be sent over the data packet network This type of operation is most commonly called voice over IP (VoIP)

15.2 DATA TRANSMISSION VERSUS CONVENTIONAL DIGITAL TELEPHONY

Conventional voice telephony is transported in a full duplex mode on PSTN circuits optimized for voice By the duplex mode we mean that there are actually two circuits, one for “send” and one for “receive” to support a normal telephone conversation between two parties

full-When we say digital in this context, we mean that all circuits would carry 8-bit “words” (timeslots), where each “word” represents an 8-bit voltage sample ostensibly of an analog voice conversation in a PCM format

A voice circuit is established when a subscriber desires to converse by telephone with some other telephone

subscriber The circuit between the two is set up by a signaling routine

The distant subscriber has a telephone address represented by a distinct telephone number consisting of 7 to 12 digits The digit sequence of the number sets up a circuit route and connectivity for conversation

The circuit is maintained in place for the duration of the conversation, and it is terminated and taken down when one

or the other party hangs up (“goes on hook”)

The address sequence of dialed digits is sent just once, at the initiation of the connectivity This whole process of setting

up a circuit, holding the connectivity in place, and then taking down the circuit is called signaling

Signaling on data circuits is approached quite differently

But there were many drawbacks

15.3 DRAWBACKS AND CHALLENGES FOR TRANSMITTING VOICE ON DATA

 Impact of errored frames (packets)

 Variation of packet arrival time, jitter buffering

 Silence suppression

 Prioritizing VoIP traffic over regular Internet and data services

 Distortion

 Sufficient bit rate capacity on interconnecting transmission media

 Voice coding algorithm standardization

 Optimized standard packet payload size

15.4 VOIP, INTRODUCTORY TECHNICAL DESCRIPTION

15.4.1 VoIP Gateway

Gateways are defined in different ways by different people

A network device that converts voice and fax calls, in real time, between the public switched telephone network (PSTN) and an IP network The primary functions of a VoIP gateway include voice and fax compression/decompression,

packetization, call routing, and control signaling Additional features may include interfaces to external controllers, such

as Gatekeepers or Softswitches, billing systems, and network management system

Figure 3 A media gateway, from one perspective API = applications program interface

Click icon to add picture

15.4.2 An IP Packet as Used for VoIP

Assume for argument that we use either a G.711 or G.726 packet (Table 15.1) IP packet

The packet consists of a header and a payload

In the case of G.711 (standard PSTN PCM), there may be a transmission rate of 100 packets per second with 80 bytes

in the payload of each packet

The total raw bytes (octets) per channel come out as follows: 40 bytes for layers 3 and 4 overhead (IP), plus 8 bytes for layer 2 (link layer) overhead So we add 48 to 80 or160 bytes (from the previous paragraph) and we get 128 or 208 bytes for a raw packet

15.4.3 The Delay Tradeoff

ITU-T Rec G.114 [10] recommends the total delay (one-way) in a voice

15.4.4 Lost Packet Rate

A second concern of the VoIP designer is lost packet rate There are several ways a packet can be “lost.” For example In the case of G.711, this would be the time equivalent of 16 or 26 msec (duration of a packet including its header) Another reason for a packet to be lost may be excessive error rate on a packet whereby it is deleted When the lost (discarded) packet rate begins to exceed 10%, quality of voice starts to deteriorate

G.723 or G.729, it is desirable to maintain the packet loss rate below 1%

IP through TCP has excellent retransmission capabilities for errored frames or packets but they are not practical for voice-over IP

15.4.4 Concealment of Lost Packets

A lost packet causes a gap in the reception stream or a single packet we are looking at a 20- to 40-msec gap The absolute silence of a gap may disturb a listener In this case, often artificial noise is inserted

Concealment techniques are most effective for about 40 to 60 msec of missing speech Gaps longer than 80 msec usually have to be muted

15.5 MEDIA GATEWAY CONTROLLER AND ITS PROTOCOLS

The gateway controller or media gateway controller (MGC) carries out the signaling function on VoIP circuits Some texts call an MGC a soft switch even though they are not truly switches but are servers that control gateways.There are four possible signaling protocol options between an MGC and gateways

ITU-T Rec H.323 This is employed where all network elements (NEs) have software intelligence

SIP (session initiation protocol [2]) is used when the end devices have software intelligence and the network itself is without such intelligence MGCP (media gateway control protocol) is another gateway control protocol

Megaco (ITU-T Rec H.248 [13]) is a gateway control protocol applicable when end devices are without software intelligence and the network has software intelligence

15.5.1 Overview of the ITU-T Rec H.323 Standard

In May 1996 the ITU ratified the H.323 specification, which defines how voice, video, and data traffic should be transported over IP-based LANs It also incorporates the ITU-T Rec T.120 [21] data-conferencing standard

A related protocol is ITU-T Rec H.324 [14] specification, which defines the transport of voice, data, and video over regular telephone networks

H.323 deals with three basic functional elements of VoIP:

Security It authenticates users of the H.323 network

It performs address translation between Internet addresses and ITU-T Rec E.164 addresses

H.323 determines call routing, to route through a gateway or be sent directly to the destination

It keeps track of the network’s bit rate capacity

15.5.3 Media Gateway Control Protocol (MGCP)

This protocol was the predecessor to Megaco and still holds sway with a number of carriers and other VoIP users MGCP assumes a call control architecture where the call control “intelligence” is outside the gateways and handled by external call control elements

In the MGCP protocol an assumption is made that the connection model consists of constructs that are basic points and connections End-points are sources or sinks of data and could be physical or virtual

end-15.5.4 Megaco or ITU-T Rec H.248 (Ref 13)

Megaco is a call-control protocol that communicates between a gateway controller and a gateway It evolved from and replaces SGCP (simple gateway control protocol) and

MGCP (media gateway control protocol) An MGC is sometimes

called a softswitch

A Megaco (H.248) is two principal abstractions relating to the model: terminations and contexts

A termination acts as sources and/or sinks for one or more data streams

A context is an association between a collection of terminations

Session Initiation Protocol (SIP)

SIP is based on RFC 2543 and is an application layer signaling protocol

SIP is closely related to IP and familiar HTTP (hypertext transfer protocol) An SIP message looks very much like

an HTTP message, especially with message formatting, header, and multipurpose Internet mail extension support

There are two modes with which a caller can set up a call with SIP These are called redirect and proxy, and servers are designed to handle these modes

The end

Thanks for your attention