Transcript
SIP Trunking and Voice over IP
Agenda What SIP
is SIP Trunking?
Signaling
How
is Voice encoded and transported?
What How
are the Voice over IP Impairments?
is Voice Quality measured?
VoIP Technology
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Speech Encoding
Speech encoding is used to compress and encode human speech before it is transmitted through the network.
Voice Sampling 1
0
Analog Voice Signal
Digitized Voice Signal
Digital Signal transmission
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
VoIP Technology Training
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Useful Definitions
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
Trunk Line
PSTN
PBX Extensions
Time Division Multiplexing (TDM) T1
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.
T1 and ISDN Training
8bits Timeslot 0
8bits Timeslot 1
: : :
: : :
8bits Timeslot 23
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8bits Timeslot 0
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ISDN PRI
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
PRI
B B B B D
1.544 Mbps T1 and ISDN Training
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SIP Trunking
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 Trunk Line
IP
IP PBX Extensions T1 and ISDN Training
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Why SIP Trunking?
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 SIPPBX.
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PRI Trunk
MSO Network
Customer Premises
PSTN
HFC Network
Voice Gateway
Internet POP - Firewall
CMTS
D3 Modem
SIP
IAD T1 PBX
PRI
• 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
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SIP Trunk
MSO Network
Customer Premises
PSTN
HFC Network
Voice Gateway
Internet POP - Firewall
CMTS
D3 Modem SIP PBX
SIP
• 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 Technology Training
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VoIP Signaling – SIP (Session Initiation Protocol)
VoIP Technology Training
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Session Initiation Protocol - SIP
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
SIP = Signaling
SIP Server IP
IP
IP
RTP= Media (Voice)
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SIP Network Elements 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)
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Simple Example of SIP Call INVITE OK Terminal A
SIP
Terminal B
requires only 3 messages to establish a call.
INVITE:
ACK
initiates the call
200 OK: response from called party ACK: confirmation
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SIP URI
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]
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SIP Call Flow Calling Party
Registrar
Called Party
Proxy
REGISTER 200 OK
INVITE (with SDP) INVITE (with SDP) 100 Trying 100 Trying 180 Ringing 180 Ringing 200 OK (with SDP) 200 OK (with SDP) ACK ACK
RTP Media = Voice
BYE BYE ACK ACK
ISDN Technology Training
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Speech Encoding
VoIP Technology Training
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Speech Encoding
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
Voice Sampling 1
0
Analog Voice Signal
Digitized Voice Signal
Digital Signal transmission
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
VoIP Technology Training
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Speech Codecs – G.711 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 Packetization
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)
IP UDP Header Header
RTP Header
Speech Codec (p-time varies depending on codec)
UDP Header
RTP Header
Speech VoIP Technology Training
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IP Telephony Packets take different paths
Address E
Address A Voice is digitized and packetized
At the receiving side, voice packets are reassembled, reordered, and played out Address D
Address B
Address C
VoIP Technology Training
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)
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VoIP Impairments
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VoIP over HFC
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
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Common VoIP Impairments
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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Echo in VoIP Networks (1)
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 Acoustic Isolation Echo
Ambient Acoustic Coupling
Requires >45 dB isolation
Poor handset or headset design
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How is Voice Quality Measured?
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Subjective Voice Quality Measurement
Subjective Test: MOS (Mean Opinion Score): panel of listeners rate the call quality
Rating Speech Quality 1
Source
VoIP Technology
Channel
2
3
4
5
5
Excellent
4
Good
3
Fair
2
Poor
1
Unsatisfactory
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Objective Voice Quality Measurement
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
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Testing Topology
PRI Voice Trunk Testing
PSTN Voice Gateway
Internet IAD
D3 Modem POP - Firewall
CMTS
T1 PBX
D3 Modem
SIP PBX
Bi-directional PESQ Voice Quality Testing
SIP Trunk Testing VoIP Technology Training
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Examples of VoIP Testing
VoIP Technology
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Thank you. Any questions?
Tel: 1.510.651.0500 www.veexinc.com
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