VoIP Conception and Implementation

transcript

VoIP Conceptionand

Implementation

LANtel Telecommunication Corp.Senior Product Manager

Jeremy Chan

Agenda

What’s VoIP and IP Telephony (IPT) VoIP Applications VoIP QoS Issue VoIP Architecture VoIP Signaling Fax over IP (FoIP)

What’s VoIP and IP Telephony (IPT)

VoIP : VoIP, Voice over Internet Protocol, is the

technology that uses the Internet Protocol to transmit voice conversation over a data network.

The primary advantages of moving voice over a data network are increased efficiency and decrease cost.

IPT (IP Telephony)

IPT (IP Telephony) : An IP Communications System that

provides a high availability and scalability telephony system.

Provide support industry standards such as H.323, MGCP, SIP, JTAPI, TAPI, … etc. VoIP signal protocol.

Enterprise Voice Solution

PBXPBX

Router/GW Router/GW

IP WAN

Soft-Switch Soft-Switch

Router/GW Router/GWIP WAN

Application Servers

PBXPBX

Packet Voice Technology

VoIP Applications

Branch Office Application

Packet Network

Branch 1

Branch N

Server Farm

Telephone

*IWF: Interworking function

Interoffice Trunking Application

TelephonePBX

Telephone

Packet Network

Interoffice Trunking Application

VoIP QoS Issue

Delay Algorithmic Delay Processing Delay Network Delay

Jitter Lost-Packet Compensation Echo Compensation

Algorithmic Delay Collect a frame of voice samples to be processed by

the voice coder. G.726 adaptive differential pulse-code modulation

(ADPCM) (16, 24, 32, 40 kbps)—0.125 microseconds G.728 LD–code excited linear prediction (CELP)(16

kbps)—2.5 milliseconds G.729 CS–ACELP (8 kbps)—10 milliseconds G.723.1 Multirate Coder (5.3, 6.3 kbps)—30

milliseconds

Processing Delay Actual process of encoding and collecting

the encoded samples into a packet for transmission over the packet network.

The encoding delay is a function of both the processor execution time and the type of algorithm used.

Network Delay Physical medium and protocols used to

transmit the voice data and by the buffers used to remove packet jitter on the receive side.

Network delay is a function of the capacity of the links in the network and the processing.

Delay Causes Problems Echo

Signal reflections of the speaker's voice from the far-end telephone equipment back into the speaker's ear.

Round-trip delay becomes greater than 50 milliseconds. (G.131)

Talker overlap one talker stepping on the other talker's speech the one-way delay becomes greater than 250

milliseconds. (G.114)

Jitter

Variable interpacket timing caused by the

network a packet traverses. Removing jitter: collecting packets and

holding them long enough to allow the slowest packets to arrive in time to be played in the correct sequence.

Causes additional delay

Lost-Packet Compensation

Lost packets can be an even more severeproblem, depending on the type of packet network that is being used. Interpolate for lost speech packets by

replaying the last packet received during the interval.

Send redundant information. Use a hybrid approach with a much lower

bandwidth voice coder to provide redundant information.

Avoiding and Managing network congestion

Normal Telephony Call

Normal Telephony Call with an Echo

Echo Compensation

Signal reflections generated by the hybrid circuit that converts between a four-wire circuit (a separate transmit and receive pair) and a two-wire circuit (a single transmit and receive pair).

The round-trip delay through the network is almost always greater than 50 milliseconds.

ITU standard G.165 defines performance requirements that are currently required for echo cancellers.

VoIP Architecture

VoIP–Embedded Software Architecture

Voice Packet Software Module digital-signal processor (DSP)

Telephony-Signaling Gateway Software Module Translating signaling into state changes used by the

packet protocol module to set up connections. Packet Protocol Module

processes signaling information and converts it. Network-Management Module

Voice-management interface to configure and maintain the other modules

VoIP Signaling

Signaling – H.323 H.323

Umbrella standard covering multimedia communications over LANs that do not provide a guaranteed Quality of Service. (H.323 v1)

Entities Terminals Gateways Gatekeepers MCUs

Protocols Parts of H.225.0 - RAS, H.225 (Q.931) H.245 RTP/RTCP Audio/video codecs

H.323 Protocol Stack

PresentationSessionTransport

Data LinkPhysical

Network

Audio SignalAudio SignalG.711G.722

G.723.1

G.728G.729 Video SignalVideo Signal

H.261 H.263

T.127DataData

RAS RTPT.124

T.125/T.122Supplementary ServicesSupplementary ServicesH.450.3 H.450.2

H.450.1

ControlControlH.245 H.225 TCP

X.224.0

H.323 protocols H.225 Covers narrow-band visual telephone services H.225 Annex G H.235 Security and authentication H.245 Negotiates channel usage and capabilities H.450.1 Series defines Supplementary Services for H.323 H.450.2 Call Transfer supplementary service for H.323 H.450.3 Call diversion supplementary service for H.323 H.450.4 Call Hold supplementary service H.450.5 Call Park supplementary service H.450.6 Call Waiting supplementary service H.450.7 Message Waiting Indication supplementary service H.450.8 Calling Party Name Presentation supplementary service H.450.9 Completion of Calls to Busy Subscribers supplementary service H.450.10 Call Offer supplementary service H.450.11 Call Intrusion supplementary service H.450.12 ANF-CMN supplementary service H.261 Video stream for transport using the real-time transport H.263 Bitstream in the RTP Q.931manages call setup and termination RAS Manages registration, admission, status RTCP RTP Control protocol RTP Real-Time Transport T.38 IP-based fax service maps T.125 Multipoint Communication Service Protocol (MCS).

H.323 Architecture

Typical H.323 Deployment

Signaling – MGCP, MAGACO Media Gateway Control Protocol

Using packages model and providing an centralized architecture where call control and services.

Controlling Telephony Gateways from external call control elements called media gateway controllers or call agents.

Entities MGC (Media Gateway controller / Call agent) MG (Media Gateway)

Protocols MGCP v1 – RFC 2705 H.248 (H.248 / MAGACO) – RFC 3525 SDP (Session Definition Protocol) - RFC 3407

MGCP Architecture

FXO/FXSE&M

Call Agent

MGCPVoice Gateway

IP Phone( MGCP Client )

Signaling – SIP Session Initiation Protocol

Multimedia protocol that could take advantage of the Internet model for building VoIP networks and applications. Using distributed architecture.

Entities User Agent Gateways Proxy Server Redirect Server Registrar Server

Protocols (RFC 2543 v1, RFC 3261 v2) SDP ( Session Definition Protocol ) URLs DNSs TRIP ( Telephony Routing Over IP

SIP Architecture

ENUM “ENUM protocol is defined by RFC 2916, aiming at

translating the numbers stemming form the ITU-T E.164 Recommendation into Internet Domain Names; ENUM is an opportunity for developing the information society.”

“As a matter of fact, ENUM allows to use a traditional telephone number in the context of different communications media, in particular those rising from the development of IP networks (e-mail, VoIP, …) and therefore, could facilitate the penetration of new applications into the mass market easily ( this market is accustomed to E.164 numbers).”

ENUM (Cont.)

ENUM is part of Convergence ENUM is part of series of technical initiatives

underway in both the IETF and ITU to develop Internet Telephony Standards.

Call Setup – H.323 – SIP Quality of Service – DIFFSERV – INTSERV – MPLS PSTN – IP Interworking H.248/MEGACO FAX – T.37, T.38 – RFC 2503 Mobile – 3GPP related

ENUM is about new service creation It must address naming and numbering

issues

VoIP Signaling Comparison

Fax over IP

FAX over IP

ITU and Internet Engineering Task Force (IETF) are working together to continue to evolve both the real-time FoIP network standard (T.38) as well as the store-and-forward FoIP network standard (T.37).

T.38 is the fax transmission protocol selected for H.323.

FoIP QoS Timing

network delay processing delay IWF must compensate for the loss of a fixed timing

of messages over the packet network. Jitter

collect packets and hold them long enough so that the slowest packets to arrive are still in time to be played in the correct sequence.

Lost-Packet Compensation repeating information in subsequent frames using an error-correcting protocol

Reference Cisco

Introduce H.323 SIP Presentation

REDCOM H.323 Tutorial

IEC Voice and Fax over Internet Protocol (V/FoIP)

ENUM.ORG Study Group A Presentation on ENUM

IETF ftp://ftp.isi.edu/in-notes/rfc2916.txt -- ENUM Core Protocol ftp://ftp.isi.edu/in-notes/rfc3261.txt -- SIP ftp://ftp.isi.edu/in-notes/rfc2705.txt -- MGCP

Thank You

VoIP Conception and Implementation

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