Voice Over IP (VoIP) Technology Reference Chart

Provider Hosted VoIP Service “IP Centrex” Enterprise VoIP to the Desktop Enterprise VoIP “Toll Bypass Model” Residential Voice Over CATV Infrastructure

Small Enterprise Site A Large Enterprise Site B Small Enterprise Site A Large Enterprise Site B
Small Enterprise Site A Large Enterprise Site B Phone Call Management Server (CMS)

1 6 7 1 6 7
PSTN via T1, PRI, etc. 2
Gateway Gateway PSTN via T1, PRI, etc.
Standard Phone HFC Access Network CMTS Managed
IP Network (LAN) IP Network (LAN)
IP Router IP Router
Voice (DOCSIS) IP Network
2 2 1 1
4 5
6 2 2 6
IP Network (LAN) IP Network (LAN) TDM PBX 4
IP Network (LAN) TDM PBX Tap and Ground
IP Router IP Router 7 4 Media
7 SYSLOG Server Gateway
ATA IP Card 4
IP Network WAN 2 eMTA Key Distribution (MG)
1 6 7 Voice Card Center (KDC)
VoIP Call
IP Phones
5
Management IP Phones Router SW/Router
Computer Data
Device VoIP Call Private WAN
Service Provider Management IP Phones DHCP Server
Device PSTN

2 IP Phones Standard Phones Video Record Keeping

IP Phones Data WAN Server
Gateway 5
IP Router DNS Server
3 PC
Or PBX upgraded with IP function Switch or Router with voice ports 4
5 to become a hybrid Television
IP-based Centrex Services Class 5 with Centrex PRI, 303
TFTP/HTTP Server Provisioning Server

The CommunicationsTest & Measurement Segment of JDSU

4 DSAM
3 PVA-1000 A handheld CATV service and installation meter for VoIP verification and
5 OPERA - Voice Quality Analysis 6 T-BERD®/MTS-4000 SmartClass Triple Play Service (TPS),
7
1 HST-3000 VoIP 2 DA-3400 An enterprise and service support software tool for IP telephony problem troubleshooting over coax cables. DSAM contains an eMTA and allows A voice and audio quality analyzer for end-to-end quality testing over any VoIP A handheld platform designed for installation and troubleshooting of ADSL, and Ethernet Testers
An IP telephony installation field tool for service turn-up and verification. A 7-layer network analyzer for VoIP and Data network monitoring and capture and analysis. PVA-1000 software provides full analysis of VoIP for efficient "find and fix" capabilities of both IP and FR issues in an HFC network, OPERA, measuring to ITU-T standards, objectively evaluates and ensures Access/FTTx networks and Triple-Play services. The layered approach of SmartClass handheld test tools that combine intelligence, power, and
The HST-3000 emulates IP phones, validates VoIP connectivity, feature troubleshooting. The DA-3400 is ideal for VoIP call quality monitoring, telephone calls, including jitter, packet loss, and audio playback. network. the quality of compressed speech (P.862 PESQ) and wideband audio signals the GUI simplifies service acceptance and troubleshooting. portability required to deliver triple-play services. Economical, yet easy-to-use
availability, and end-user voice quality. signaling analysis, and problem identification and isolation. (BS.1387 PEAQ). handheld point solutions suitable for tier-1 field technicians.

Protocol Stack Real-Time Protocol (RTP) Signaling Quality Impairments (Transport-Related) VoIP Test Phases
0 1 2 3 4 5 6 7 Octet
V P X CSRC Count 1 Equipment and/or Network Design Verification Network Audit/VoIP Pre-Assessment Service Installation/Turn-up
H.323
• Defined in RFC 3550 and 3551 M Payload Type 2 Delay – Latency Equipment manufacturers test their products for full functionality, Before adding VoIP as an application on an existing LAN/WAN During service installation, the underlying physical and datalink
Voice Sample/ Sequence Number 3 • ITU-T H.225 Call control signaling for multimedia systems
• Used by H.323, SIP, MGCP/NCS, MEGACO, and PacketCable • End-to-end delay from speaker to listener performance under load, regression, and conformance to specifica- infrastructure, evaluate the network for readiness via a detailed layers should be fully tested for any marginal performance issues
Signaling (SIP, H.323, SCCP, NCS, etc.) CODEC G.7XX RTCP Y/N Time Stamp 4 • CODEC operation
• Transports real-time voice and video content • ITU-T H.235 Security and encryption for H-series terminals tions. Providers and Enterprises implementing VoIP should repeat baseline and through an active simulation of the deployed VoIP followed by a test of the VoIP service itself. Perform inbound, out-
SSRC 5 • Lower bit rate = higher delay
• ITU-T H.245 Media transport channel setup and control many of the same tests in their specific environment before adding service. Test equipment requirements include network discovery bound, on-net and off-net calls to verify provisioning and perform-
RTP CSRC 6
ance across the IP/PSTN boundaries. Test equipment requirements
or upgrading equipment. Test equipment requirements for this and mapping tools, protocol analysis for statistics collection, and
RTP Structure • ITU-T H.450 H.323 supplemental services
TCP UDP phase include conformance testers, load generators, voice quality distributed software or hardware VoIP agents for generation and include software-based or handheld portable tools suitable for
V This stands for version. It identifies the RTP version. • ITU-T T.38 Real-time fax over IP analyzers, and protocol analyzers. (Products: DA-3400, Opticom measurement of VoIP performance. (Products: DA-3400, the field technician. (Products: DSAM, T-BERD/MTS-4000, and
IP P This stands for padding.When set, the packet contains one or more additional padding octets at the end. • IETF RFC 3550 RTP for media transport over IP
Jitter OPERA and PVA-1000) HST-3000, T-BERD/MTS-4000, and PVA-1000) HST-3000)
Padding octets are not part of the payload. • Variations in inter-packet delay

HDLC (PPP, FR) ATM ETHERNET X Thisstandsforextensionbit.Whenset,thefixedheaderisfollowedbyexactlyoneheaderextensionwithadefinedformat. Packets Equally Spaced Variation in Packet Spacing Service Troubleshooting Performance Monitoring
IP
CSRC Count This contains the number of CSRC identifiers that follow the fixed header. SIP If the VoIP service is inoperable, intermittent, or degraded, identify the Larger deployments with stringent service expectations warrant a
PHYSICAL (Ethernet, T1, T3, DOCSIS, etc) M This stands for marker. The interpretation of the marker is defined by a profile. It is intended to allow • IETF RFC 3261 Session Initiation Protocol root cause before taking corrective action. Tools for troubleshooting vary distributed, passive analysis system to monitor and alarm on various
for significant events, such as frame boundaries, to be marked in the packet stream. widely with the nature and architecture of the VoIP Network, but they can performance thresholds. Some of these requirements may be met by
• IETF RFC 3550 RTP for media transport over IP
Payload Type This identifies the format of the RTP payload (G.729, G.711, etc.) and determines its interpretation by the
Packet Loss include fixed or portable, passive or active test systems, and dispatchless monitoring functions in the VoIP equipment itself, but they are often
The variables Constant since this is VoIP application. A profile specifies a default static mapping of payload type codes to payload formats. software. Functionality should include VoIP end-point emulation, call augmented by purpose-built peripherals and software that integrate
• Packets that are not delivered to the destination
Additional payload type codes may be defined dynamically through non-RTP means. MGCP capture, playback, and QoS scoring via MOS. (Products: DA-3400, DSAM, with operations support systems (OSS) used by the network opera-
Three Packets In Two Packets Out
Sequence This increments by one for each RTP data packet sent. It may may be used by the receiver to detect packet • IETF RFC 3435 Media Gateway Control Protocol IP HST-3000, T-BERD/MTS-4000, and PVA-1000) tions center (NOC). (Products: DA-3400)
Number loss and to restore packet sequence.
• IETF RFC 3550 RTP for media transport over IP
Voice Encoding Time Stamp This reflects the sampling instant of the first octet in the RTP data packet. The sampling instant must be
Test Methodologies - Signal Degradation Analysis Test Methodologies - Packet Transport Analysis
derived from a clock that increments monotonically and linearly in time to allow for synchronization
accuracy and for measuring packet arrival jitter. PacketCable Out-of-Sequence Packets
Example: G.729
SSRC This identifies the synchronization source. The identifier is chosen randomly with the intent that no two • Packets that are not delivered in order
Sample = 10 ms Voice • NCS Network-Based Call Signaling protocol Signal Degradation Analysis Packet Transport Analysis
Analog Signal 2 Samples per Packet IP-UDP-RTP 2x Samples synchronization sources within the same RTP session will have the same SSRC identifiers. In Sequence Out of Sequence
• IETF RFC 3550 RTP for media transport over IP IP This out-of-service (objective) test evaluates distortion across all network segments from end to end This in-service (subjective) test computes the impact of packet transport problems on call quality
CSRC This identifies the contributing sources for the payload contained in this packet.
• Receiver compares received sample against known original sample • Measures jitter, packet loss, and latency
A-to-D Conversion CODEC Compression Buffer Samples
• Complex signal analysis identifies network induced distortion • Focuses on monitoring and troubleshooting of customer problems
Compressed RTP (CRTP) MEGACO/H.248
Echo • PESQ MOS quality scale from 1 (lowest) to 4.5 (highest) • MOS quality scale from 1 (lowest) to 5 (highest)
• CRTP greatly reduces the overhead for Voice applications over slow links. • ITU-T H.248 H.248 joining of IETF and ITU-T VoIP signaling standards • Reflection of speaker’s voice to speaker’s ear • R-Factor values from 0 (lowest) to 100 (highest)
CODEC CODEC Bandwidth IP Bandwidth IP Bandwidth
• IETF RFC 3525 MEGACO joining of IETF and ITU-T VoIP signaling standards • Not a true digital or VoIP problem Analog PESQ - PAMS PESQ
NO Silence Suppression 30% Silence Suppression (Digital) P.862 Perceptual Evaluation of Speech MOS – Mean Opinion Score
• Compresses the IP/UDP/RTP header in a RTP data packet from 40 bytes to approximately 2 to 5 bytes. • Aggravated by network latency • Uses pre-recorded voice samples ITU ITU-T G.107 MOS Scores for VoIP
Gateway
• IETF RFC 3550 Real-time multimedia encapsulation over IP • In-service monitoring of telephone calls
G.711 PCM 64 Kbps 80 Kbps 56 Kbps IP Network • No network monitoring capability PAMS
(Analog) P.861 Perceptual Analysis and Measurement System
• RTP header compression is supported in point-to-point networks (FR, HDLC, PPP, etc.) • Measures packet transport impairments ETSI TS 101 329-5 R Factor Values for VoIP
G.729 CS-ACELP 8 Kbps 24 Kbps 16.8 Kbps Proprietary
Pre-recorded Sample Pre-recorded Sample
Before RTP Header Compression: After RTP Header Compression: Multi-Tandem Distortion Transmitter Receiver
Acceptable MOS and R-Factor Values for Narrowband CODECs
G.723.1 ACELP 5.6 Kbps 16.27 Kbps 11.39 Kbps • Cisco Skinny Client Control Protocol (SCCP)
• Multiple transitions between CODECs User Option R-Factor MOS Score
20 bytes 8 bytes 12 bytes 2-5 bytes VoIP Network Maximum obtainable for G.711 93 4.4
• Nortel UniStim G.711 • Each transition adds distortion G.711
Very satisfied 90-100 4.3-5.0
G.723.1 MP-MLQ 6.4 Kbps 17.07 Kbps 11.95 Kbps VoIP Network
Satisfied 80-90 4.0-4.3
IP UDP RTP Payload Payload • Avaya custom H.323
G.729 G.729 Some users satisfied 70-80 3.6-4.0
G.726 32 Kbps 40 Kbps 29 Kbps G.728 Analyzer Analyzer Many users dissatisfied 60-70 3.1-3.6
20 to 180 bytes 20 to 180 bytes Nearly all users dissatisfied 50-60 2.6-3.1
Not recommened 0-50 1.0-2.6
Header IP/UDP/RTP Header

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