sip trunking and voice over ip - scte san...

SIP Trunking and Voice over IP

VoIP Technology Confidential & Proprietary Information of VeEX Inc. 2

Agenda

What is SIP Trunking?SIP SignalingHow is Voice encoded and transported?What are the Voice over IP Impairments?How is Voice Quality measured?

Speech encoding is used to compress and encode human speech before it is transmitted through the network.

For the G.711 (PCM) codec, the voice is sampled at a rate of 8 kHz (8,000 samples per second) and digitized using 8-bit samples before being transmitted

Confidential & Proprietary Information of VeEX Inc.

Speech Encoding

Analog Voice Signal

Digitized Voice Signal

Voice Sampling

1

0

Digital Signal transmission

VoIP Technology Training 3

TRUNK: A trunk is a circuit that connects telephone switches.

Private Branch Exchange (PBX): PBXs make connections among the internal telephones of a private organization and also connect them to the via trunk lines

Trunking saves cost, because there are usually fewer trunk lines than extension lines, since it is unusual in most offices to have all extension lines in use for external calls at once

Useful Definitions

ExtensionsPBX

Trunk LinePSTN

The T1 signal takes 24 DS0 channels and multiplexes them into a single bit stream using Time Division Multiplexing

The multiplexing is octet (8 bits) oriented. 8 bits for each channel are contiguous.

Each channel uses 64kbps, the bandwidth is reserved for each channel whether there is an active call or not.


Time Division Multiplexing (TDM) T1

:::

8bits 8bits 8bits::: 8bitsTimeslot 0 Timeslot 1 Timeslot 23 Timeslot 0

T1 and ISDN Training 5

ISDN = Integrated Services Digital Network

With ISDN voice and data can be carried simultaneously

B Channel (Bearer) = 64 kbps for voice

D Channel (Data) = 16 or 64 kbps for signaling or data

ISDN Services: ISDN – PRI: Primary Rate Interface 23 B channels + 1 D channel ISDN PRI is carrier over T1 and is used for PBX interconnect


ISDN PRI

PRIBB

D

B

B

1.544 MbpsT1 and ISDN Training 6

SIP trunking: is a Voice over Internet Protocol (VoIP) service based on the signaling protocol Session Initiation Protocol (SIP)

The SIP trunk connects customers equipped with SIP-based IP-PBX to the telephone network using IP protocol

SIP Trunking

T1 and ISDN Training Confidential & Proprietary Information of VeEX Inc. 7

ExtensionsIP PBX

SIP Trunk LineIP

T1 lines and ISDN PRI leased lines are traditionally used by SMBs for PBX Voices Services

Telcos T1 line are dedicated 4-wire lines, these lines are expensive and hard to maintain

MSOs can use their existing HFC footprint to offer Business Voice Services at a lower cost.

Cable operators benefit from new business revenue while SMBs benefit from decreased operating costs compared to leasing and maintaining T1 lines.

Customer Premises installed with a DOCSIS modem and an Integrated Access Device (IAD), SMBs can keep their legacy ISDN PRI-PBX equipment. The IAD provides PRI to SIP “translation”.

Customer Premises installed with a DOCSIS modem can use their existing SIP-PBX.

Why SIP Trunking?


PRI Trunk

VoIP Technology Training Confidential & Proprietary Information of VeEX Inc. 9

PSTN

InternetVoice Gateway

POP - FirewallCMTS D3 Modem

IADT1 PBX

HFC Network

PRISIP

Customer PremisesMSO Network

• Capacity up to 46 trunk lines on HFC network (2 x PRI) or more on Fiber network

• Use customer existing PRI PBX equipment• IAD = Demarc converting PRI into SIP

SIP Trunk


PSTN


POP - FirewallCMTS D3 Modem

SIP PBX

HFC Network

SIP

Customer PremisesMSO Network

• End to End SIP • Capacity up to 46 trunk lines on HFC network (2 x PRI) or more on

Fiber network• Customer owned SIP PBX

VoIP Signaling – SIP (Session Initiation Protocol)

Confidential & Proprietary Information of VeEX Inc. 11VoIP Technology Training

Session Initiation Protocol (IETF RFC 3261)

Signaling protocol used for the setup and signaling of VoIP calls

SIP messages are exchanged between the SIP end points, which are the phones, and the SIP elements, which are the SIP servers, to establish the call

Session Initiation Protocol - SIP

SIP ServerSIP = Signaling

RTP= Media (Voice)

IP

IP

IP

VoIP Technology Training 12Confidential & Proprietary Information of VeEX Inc.

User Agent (UA): SIP endpoint. For example: an SIP phone or “soft phones” application running on PC

Registrar: Server that receives the register request from users and keeps track of where to locate users

Session Border Controller (SBC): Can be used in the network to provide services to the UAs connected to it, for security topology hiding or NAT traversal

Gateways: Provide interconnect function between SIP and other networks (PSTN or H.323)

SIP Network Elements


SIP requires only 3 messages to establish a call. INVITE: initiates the call 200 OK: response from called party ACK: confirmation

Simple Example of SIP Call

Terminal A Terminal B

INVITE

OK

ACK


All the SIP messages contain a TO and FROM fields, those fields are the SIP URI (Universal Resource Identifier)

URI are in the format user@domain [email protected] [email protected] POTS: [email protected] POTS: [email protected]

SIP URI


ISDN Technology Training Confidential & Proprietary Information of VeEX Inc. 16

SIP Call Flow

REGISTER

200 OK

INVITE (with SDP)

100 Trying

Calling Party Registrar Called Party

RTP Media = Voice

Proxy

INVITE (with SDP)

100 Trying

180 Ringing180 Ringing

200 OK (with SDP)200 OK (with SDP)

BYEBYE

ACKACK

ACKACK

Speech Encoding

Confidential & Proprietary Information of VeEX Inc. 17VoIP Technology Training

Speech encoding is used to compress and encode human speech before it is transmitted through the network. It is used in IP and mobile telephony

For the G.711 (PCM) codec, the voice is sampled at a rate of 8 kHz (8,000 samples per second) and digitized using 8-bit samples before being transmitted


Speech Encoding

Analog Voice Signal

Digitized Voice Signal

Voice Sampling

1

0

Digital Signal transmission

VoIP Technology Training 18

G.711 Pulse Code Modulation (PCM), is a very commonly used waveform codec.

Two different versions are available: G.711 µ-law used in North America, G.711 A-law used in the rest of the world

Sampling frequency is 8kHz, bandwidth is 64 kbps

Each voice packet contains 20ms speech

Different encoding techniques achieve different results. There is usually a tradeoff between speech quality and bandwidth usage, computational delay and complexity.

G.711 PCM encoding produces good quality speech but relatively poor bandwidth usage (64 kbps). It is generally seen as the toll-quality codec and has a MOS of 4.2 when no external impairments (e.g. delay, packet loss) are present.

Other codecs such as G.723.1 or G.729 achieve better bandwidth performance at the detriment of speech quality. Their MOS score is 3.8 and 3.9 respectively when no other impairments are present.


Speech Codecs – G.711

Encoded speech is broken down into frames containing usually 20 ms or 30-40 ms of speech. The amount of speech contained in a frame is described as p-time (packetization time). Speech frames are transported on the network using RTP (Real Time Transport Protocol) and UDP (User Datagram Protocol)


Speech Packetization

Speech Codec (p-time varies depending on codec)

RTP Header

UDP Header

IP Header

RTP Header

UDP Header

Speech

IP Telephony

Voice is digitized, packetized and sent over the IP network.

As voice travels through the network, packets may be impaired as a result of network impairments (i.e., jitter, delay, Packet loss)

Packets take different paths

Address B

Address E

Address D

Address C

Address A

Voice is digitized and packetized


At the receiving side, voice packets are

reassembled, reordered, and played out

VoIP Impairments

Confidential & Proprietary Information of VeEX Inc. 22

Voice trunking requires more HFC bandwidth than the traditional residential services

Up to 46 simultaneous trunk lines can be in use

Careful traffic engineering at the CMTS and in the backbone is required

VoIP over HFC


Network delay High levels of delay (generally over 200 milliseconds round trip) can cause

problems with conversational interaction. Delay can also make echo problems more obvious and annoying The sources of delay on a VoIP include codec encoding/decoding delay,

packetization time, network transmission delay


Common VoIP Impairments

Jitter Jitter is the variation in packet transit delay it is cause by queuing, congestion, network

route changes Although some level of jitter is expected and taken care of in the jitter buffer, excessive

jitter will cause packet discard, degrading speech quality.



Packet Loss Occurs due to a variety of reasons including Link failure, high levels of congestion, router

buffer overflow, physical links problems … Most codecs are equipment with Packet Loss Concealment algorithm that mask the

effects of lost packets. These algorithms are inefficient if packet loss comes in bursts and bursty packet loss has a severe impact on voice quality even if the average packet loss rate for the call is low.

Out of order or Duplicate packets Occurs due to a variety of reasons including Link failure, high levels of congestion, router

buffer overflow, physical links problems … Out of order or duplicate packets are taken care of by the jitter buffer, but excessive

count could lead to packet discard and indicate network issues



Echo Sources of Echo: Reflection on the 2/4 Wire interface (Hybrid echo) Acoustic Echo created by Created by the voice reflected back from the microphone to

the speaker due to device issues or reflective environment

Echo reported by Echo Return Loss (ERL) 55 dB ERL represents a low echo 15 dB ERL represents a high echo


Echo in VoIP Networks (1)

Acoustic Isolation Echo

Poor handset or headset design

Requires >45 dB isolation

Ambient Acoustic Coupling

How is Voice Quality Measured?


Subjective Test: MOS (Mean Opinion Score): panel of listeners rate the call quality


Subjective Voice Quality Measurement

Rating Speech Quality5 Excellent

4 Good

3 Fair

2 Poor

1 Unsatisfactory

1 2 3 4 5

Source Channel

Objective Tests (machine tests): ITU-T P.862 (PESQ) determine the distortion introduced by a transmission system or

codec by comparing an original reference file sent into the system with the impaired signal that came out

Objective Voice Quality Measurement


Testing Topology


PSTN


POP - FirewallCMTS

D3 Modem IAD

T1 PBX

PRI Voice Trunk Testing

D3 Modem SIP PBX

SIP Trunk Testing

Bi-directional PESQ Voice Quality Testing

Examples of VoIP Testing


Thank you.Any questions?


Tel: 1.510.651.0500www.veexinc.com

sip trunking and voice over ip - scte san...

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