Post on 15-Feb-2020
SIP Trunking and Voice over IP
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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
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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.
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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
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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
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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?
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PRI Trunk
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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
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PSTN
InternetVoice Gateway
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)
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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
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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
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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
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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 joe@domain.com joe@192.168.102.1 POTS: 5106500500@domain.com POTS: 5106500500@192.168.102.1
SIP URI
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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
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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
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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.
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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)
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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
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At the receiving side, voice packets are
reassembled, reordered, and played out
VoIP Impairments
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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
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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
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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.
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Common VoIP Impairments
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
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Common VoIP Impairments
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
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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?
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Subjective Test: MOS (Mean Opinion Score): panel of listeners rate the call quality
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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
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Testing Topology
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PSTN
InternetVoice Gateway
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
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Thank you.Any questions?
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Tel: 1.510.651.0500www.veexinc.com