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Introduction to VoIP, RTP and SIP

Archana KesavanProduct Marketing Manager

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Telephony – A Brief HistoryMr. Watson – Come here – I want to see

you

Manual Switchboard

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Telephony – A Brief History

Business Office Callers

Local Exchange

PBX

Individual Callers

International Gateway Tandem Junction/Exchange

PSTN Network

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Voice-Over-IP

• Set of protocols designed to deliver communication services over the IP network

• Analog voice converted into data packets to be sent over the Internet.

• Two phases• Phase 1: Signaling (SIP)• Phase 2: Audio transport (RTP)

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IP Telephony or Voice-Over-IP

Local Exchange

PBX

Individual Callers

International Gateway Tandem Junction

PSTN

Net

wor

k

IP- PBXVoIP Servers

Ethernet (IP Network)

InternetSIP Trunk

Mobile Voice

Mobile Data

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Type of VoIP ServicesOn-Prem

Branch Office

Branch Office

DMZ

CRMWeb

Data Center

IP-PBXVoIP server

• All hardware and software owned and managed by the enterprise.

• IP-PBX and adjoining systems reside in the datacenter or on premise.

• Voice packets go through the LAN and WAN

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Type of VoIP Services

Hosted

• IP-phones are owned by the enterprise

• All other equipment and software located in the service provider data centerand provided as a service

• VoIP packets travel over the Internet or dedicated WAN connectivity to the hosted site

Data Center

Branch Office

Branch OfficeDMZ

CRMWeb

IP-PBXVoIP Provider Data Center

Branch Office

Branch OfficeDMZ

CRMWeb

IP-PBXVoIP Provider Data Center

Data Center

Enterprise A

Enterprise B

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So how does VoIP work?

• Session Initiation Protocol (SIP)• Pre-requisite for the voice call– RFC 3261: Standard protocol (however

propriety versions exist to force vendor lock-down)

– Application level protocol residing above TCP/IP stack

– TCP or UDP– Text-based protocol like HTTP– Encrypted with TLS– Response Codes indicates the state of

the request message

Phase1: SignalingVoIP Phone A SIP Server/Proxy VoIP Phone B

SIP RegisterSIP Register

SIP INVITE

100 Trying SIP INVITE

180 Ringing180 Ringing

200 OK200 OK

AUDIO CALLSIP BYE

200 OK

SIP ACK SIP ACK

SIP BYE

200 OK

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So how does VoIP work? Phase1: Signaling

VoIP Phone A SIP Server/Proxy VoIP Phone B

SIP RegisterSIP Register

SIP INVITE

100 Trying SIP INVITE

180 Ringing180 Ringing

200 OK200 OK

AUDIO CALLSIP BYE

200 OK

SIP ACK SIP ACK

SIP BYE

200 OK

REGISTER

INVITE

CONNECT

DISCONNECT

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• Real Transport Protocol (RTP)– Analog voice signals converted into data

packets and sent over UDP– Audio frames are encapsulated in RTP

packets – RTP packets are encapsulated in UDP

packets – UDP packets are encapsulated in IP

packets

So how does VoIP work? Phase 2: Audio Data

IPheader

UDPheader Frame 1RTP

header Frame 2

RTP Audio Stream

SIP Network

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• How voice traffic is encoded and decoded• Determines the quality of the VoIP conversation• G.711, G.722, SILK

Key VoIP Concepts

• MoS• Latency• Jitter (De-Jitter buffer)• PDV

Codecs

VoIP Metrics

• Prioritization of VoIP Traffic • DSCP codes – Traffic shaping, firewall and LB configuration– 3 bits for class: Best effort, Assured Forwarding,

Expedited Forwarding, Voice Admit

QoS

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Time

Packet delay (from sender to receiver)Latency

Packet 1 Packet 2 Packet 4Sent at

Packet 1 Packet 2 Packet 4Received at Packet 3Packet 3

Latency LatencyLatency Latency

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Time

Variation of the latency

Jitter

Packet 1 Packet 2 Packet 4Sent at

Packet 1 Packet 2 Packet 4Received at Packet 3Packet 3

Min Latency Max Latency

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Time

99.9th percentile of the packet delay variation

Packet Delay Variation

Packet 1 Packet 2 Packet 4Sent at

Packet 1 Packet 2 Packet 4Received at Packet 3Packet 3

Played at

Delayed playback

Min Latency Max Latency PDV = max latency – min latency

De-jitter buffer should be able to accommodate PDV.

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E-Model (ITU-T Recommendation G.107, 1998-2014)Based on a mathematical model in which the individual transmission parameters are transformed into different individual "impairment factors” such as codec characteristics, delay, loss ratio, discard ratio, etc., to obtain a quality metric called R factor:

Mean Opinion Score (MOS)

Basic signal-to-noise ratio

Delay impairment

Equipment impairment

Advantage factor

(expectation)

• Network latency• De-jitter buffer size

• Ie (codec)• Packet loss robustness (codec)• Packet loss probability

• Network latency

Simultaneous impairment

4 Rec. ITU-T G.107 (12/2011)

There are three different parameters associated with transmission time. The absolute delay Ta represents the total one-way delay between the send side and receive side and is used to estimate the impairment due to excessive delay. The parameter mean one-way delay T represents the delay between the receive side (in talking state) and the point in a connection where a signal coupling occurs as a source of echo. The round-trip delay Tr only represents the delay in a 4-wire loop, where the "double reflected" signal will cause impairments due to listener echo.

7.1 Calculation of the transmission rating factor, R According to the equipment impairment factor method, the fundamental principle of the E-model is based on a concept given in the description of the OPINE model (see [b-ITU-T P-Sup.3]).

Psychological factors on the psychological scale are additive.

The result of any calculation with the E-model in a first step is a transmission rating factor R, which combines all transmission parameters relevant for the considered connection. This rating factor R is composed of:

AIe-effIdIsRoR +−−−= (7-1)

Ro represents in principle the basic signal-to-noise ratio, including noise sources such as circuit noise and room noise. Factor Is is a combination of all impairments which occur more or less simultaneously with the voice signal. Factor Id represents the impairments caused by delay and the effective equipment impairment factor Ie-eff represents impairments caused by low bit-rate codecs. It also includes impairment due to randomly distributed pack losses. The advantage factor A allows for compensation of impairment factors when the user benefits from other types of access to the user. The term Ro and the Is and Id values are subdivided into further specific impairment values. The following clauses give the equations used in the E-model.

7.2 Basic signal-to-noise ratio, Ro The basic signal-to-noise ratio Ro is defined by:

( )NoSLRRo +−= 5.115 (7-2)

The term No [in dBm0p] is the power addition of different noise sources:

»»¼

º

««¬

ª+++= 10101010 10101010log10

NfoNorNosNc

No (7-3)

Nc [in dBm0p] is the sum of all circuit noise powers, all referred to the 0 dBr point.

Nos [in dBm0p] is the equivalent circuit noise at the 0 dBr point, caused by the room noise Ps at the send side:

( )214004.0100 −−−+−−−= DsOLRPsDsSLRPsNos (7-4)

where OLR = SLR + RLR. In the same way, the room noise Pr at the receive side is transferred into an equivalent circuit noise Nor [in dBm0p] at the 0 dBr point.

2)35(008.0121 −++−= PrePreRLRNor (7-5)

The term Pre [in dBm0p] is the "effective room noise" caused by the enhancement of Pr by the listener's sidetone path:

»»¼

º

««¬

ª++= 10

)–10(

101log10LSTR

PrPre (7-6)

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• Data records that contain specific information about a call. For eg, timestamp, call duration etc

• CDRs are generated at specified triggers• Record call quality, loss, latency experienced • Billing, Law Enforcement

Monitoring Techniques for VoIP

• Packet sniffer that can record every SIP and RTP transaction• Can typically decode speech and replay for call quality analysis• Detect MOS score and other voice metrics • Maximum overhead

Call Detail Records

Packet Capture

• Simulate VoIP traffic from strategic vantage points in periodic intervals

• Quickly pinpoint when and where an issue occurs • Real time detection of voice quality degradation• Less overhead

Active Monitoring

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Demo

• Dip in MOS score due to DSCP change• https://earhhpng.share.thousandeyes.com

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VoIP Metrics

Average of packet delays

99.9th percentile of packet delay

variation

Packets dropped by the de-jitter buffer

Packets dropped by the network

MOS Score (1-5)

Audio codec used

Source

Destination

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Thank You!