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CCNA Voice
A Brief History of Telephony
www.INE.com
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Traditional PBX Overview • Traditional Post Branch eXchange (PBX)
– Station or (Line) side • Line card connected to phones – both analog and
digital (proprietary to vendor) – Trunk side
• Trunk cards connected to a local Central Office (CO) or to another PBX (called Tie-Trunks)
– Accessory cards (e.g. Voicemail, IVR, etc)
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Disadvantages to PBX Architecture • All voice traveled over dedicated, separate wires to PBX • All voice inside PBX must go through a single Time Division
Multiplexing (TDM) backplane • This can cause congestion at peak times, as the backplane
cannot support enough TDM ‘channels’ or ‘slots’ for every single phone attached to the system to make a voice call
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Analog Voice Circuits • FXS (Foreign eXchange Station) uses 2-wire Tip/Ring in a RJ-11 port
– Typically connects to analog phone or fax (although can connect to FXO port) – Provides power (-48V), call progress tones (ringback, ringing) and dial tone – Acts like a CO to the analog phone
• FXO (Foreign eXchange Office) uses 2-wire Tip/Ring in a RJ-11 port – Typically connects to the CO (although can connect to FXS port) – Provides supervised disconnect – Supports CallerID
• E&M (Ear & Mouth or Earth & Magnet) uses 4-wire Tip/Ring & Earth/Mouth in a RJ-12 port
– Tie-Line to another PBX – Music on Hold (MoH) source – Paging source
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Digital Voice Circuits • Digital circuits carry information in ‘Channels’ • Two main types of channels
– Bearer – Data
• Two main types of digital circuits: – Channel Associated Signaling (CAS) – Common Channel Signaling (CCS)
• Time Division Multiplexing (TDM) is used on all circuits – Many voice conversations are each sampled, then ‘cut up’ into slices
and interwoven with one another, & Data keeps track of conversations
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Digital Voice Circuits • Channel Associated Signaling (CAS)
– Voice and data are both carried on Bearer channels – Each Bearer channel is ‘robbed’ of bits in order to transmit data
alongside voice (why CAS is sometimes called Robbed Bit Signaling) – Example of this is T1 Wink Start or E1-R2
• Common Channel Signaling (CCS) – Voice alone is carried on Bearer channels – Data about the voice is carried on a single or ‘common’ Data channel – Example of this is T1 or E1 ISDN PRI as well as ISDN BRI
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Circuit Switched Network • Involves two nodes (phone-to-phone, phone-to-CO, etc) • They establish a dedicated channel or ‘circuit’ in order to talk • Circuit remains connected for the entire duration of the call • Circuit acts as if the nodes were physically connected as with
an electrical circuit • Public Switched Telephone Network (PSTN) is largest circuit-
switched network in the world – Sort of like the Internet but instead many Telephone Company (Telco)
‘switches’ connected together with TDM circuits using the Signaling System 7 (SS7) data protocol
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(Dis)/Advantages Circuit Switched • Advantages:
– Constant connection – Guarantee that if a channel is available, every bit from voice will arrive at other
side – Bitrate & delay both stay constant – Quality is inherently excellent
• Disadvantages – Not always enough channels available – Bitrate is limited – no (little) chance for newer technology to improve sound
quality – Must bond many channels together to allow video
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Packet Switched Network • Digital network that transmits data, irrespective of content into
blocks of data called packets • Layer 3 devices carry packets and encapsulate them into
Layer 2 ‘frames’ • Each device makes decisions independent from the previous
L3 routing or L2 switching device and determines where the packet or frame (respectively) should be sent
• Internet obviously the largest of these, but of course many private enterprise networks exist
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(Dis)/Advantages Packet Switched • Advantages:
– Bitrate stays constant – Bitrate is not limited – newer technology often improves sound quality – Eas(ier) to increase bandwidth for more features such as video – Quality of Service allows us to overcome ALL of the disadvantages
• Disadvantages – No dedicated connection as with circuit switched networks – No inherent guarantee the packet will ever arrive (Packet Loss) – Not always enough bandwidth available (although much easier to add) – Delay can occur at various router hops along the way – Delay can and often does vary greatly from hop-to-hop (Jitter)
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Unified Communication Networks • Everything flows over the same network as standard
Data, however QoS keeps UC packets prioritized – Voice Calls (hardware-based and Mac/PC soft-based) – Video Calls and Conferences – Voicemail – Presence and Instant Messaging – Directory Services – Phone-based applications
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Public Providers • UC systems can connect to traditional PSTN or to
newer VoIP providers – Internet Telephony Service Providers (ITSP) – Use either SIP (typical) or H.323 (older providers)
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Voice over IP • Voice over IP (VoIP)
– Taking analog voice (or video) and sampling and encoding it into a digital value for transmission in an IP data payload
– Uses both a Signaling protocol and a Media protocol to comprise the overall VoIP conversation, and they are transmitted separately Signaling protocol is used to ‘setup’, ‘teardown’, and control information about the call
– Signaling being successfully negotiated results in Media transmission – Media protocol samples Voice/Video using a CODEC, is completely
independent from signaling, and is sent using UDP
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CODEC – COmpressor / DECompressor • Codec carries the Voice or Video • COmpressor used in Tx direction • DECompressor used in Rx direction • Codec carried in the Media stream (RTP stream) • Codecs include (but not limited to)
– G.711 – G.722 – G.729 – iLBC – iSAC
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CODECs :: G.711 • G.711
– ITU standard – Uses Pulse Code Modulation (PCM) – 64kbs default rate – Excellent audio quality – For LAN connections
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CODECs :: G.722 • G.722
– ITU standard – Optimized for wideband speech – 64kbs default rate – Superior audio quality to G.711 – For LAN connections
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CODECs :: G.729 • G.729
– 8kbs default rate – High-complexity CPU / DSP – For WAN connections
• G.729A variant – Medium-complexity CPU / DSP
• G.729B variant – High-complexity CPU / DSP – Addition of Voice Activity Detection (VAD, and VAD is BAD) – Addition of Comfort Noise Generation (CNG) to ‘makeup’ for the loss of
transmission of voice with VAD (people freak out at pure silence)
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CODECs :: iLBC • Internet Low Bitrate Codec (iLBC)
– 13.3kbps default rate – Optimized for narrowband speech and lossy WAN
connections (e.g. Internet)
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CODECs :: iSAC • Internet Speech Audio Codec (iSAC)
– 10-32kbps adaptive, variable rate – Optimized for wideband speech and packet jitter
(differences in delay) over WAN connections (e.g. Internet)
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Digital Signal Processors (DSPs) • Converts Analog-to-Digital or Digital-to-Analog by
– Sampling (listens) analog voice (or video) – Quantization (approximates continuous range of sampled
values) – Encoding (converts quantization values into digital values
(CODECs) and packetizes them into IP data payload) – Compression (optionally compresses packets using
hashing algorithms)
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Packet Data Voice Modules (PVDMs) • Cisco-proprietary DSP’s
– Used in IP Phones for Voice Termination – Used in VoIP gateways for Voice Termination – Used in VoIP DSPFarms for
• Conferencing • Media Termination Point (MTP) • Transcoding (is also an MTP)
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Real Time Protocol (RTP) • RTP is the Layer 4 protocol that rides atop of UDP, and encapsulates all delay-
sensitive traffic, such as Voice and Video • UDP Ports range from 16384 – 32767 • Real Time Control Protocol (RTCP) is a RTP control and quality & statistics
protocol and is sent along side or ‘paired’ with each RTP stream, and uses same randomly chosen port as the RTP stream, but +1 (e.g. RTP stream chooses UDP port 20743, RTCP will be sent alongside on UDP port 20744)
• Security by means of encrypting the voice data payload using the AES cipher (does not encrypt the entire packet as IPSec would) is performed by Secure RTP (SRTP) using Transport Layer Security (TLS aka SSL v3.1)
• Encapsulation: – Layer 2 Layer 3 IP Header UDP Header RTP Header Voice Payload
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Signaling Protocols • Signaling protocol used to setup, teardown, and control
information about call, such as ‘Supplementary Services’ which include things like Call Forward, Transfer, Hold, Busy, Redirect, Call Park, Call Pickup, Presence, MWI, etc, etc, etc
• Most common protocols used for Cisco UC – H.323 – SIP – MGCP – SCCP
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H.323 • International Telecommunications Union (ITU) • Evolved from ISDN Q.931 Layer 3 signaling (from H.
320 video-conferencing to be more exact) • Peer-to-Peer Protocol • Endpoints (called Gateways) are intelligent, have
independent dial plans (no registration required) • Endpoints can register to a centralized dial plan
server called a Gatekeeper
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SIP – Session Initiation Protocol • Internet Engineering Task Force (IETF) • Peer-to-Peer Protocol • Endpoints are intelligent, have independent dial
plans (no registration required) • Endpoints can register to a centralized dial plan
server called a SIP Proxy Server (other SIP servers)
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MGCP– Media Gateway Control Protocol • Internet Engineering Task Force (IETF) • Designed specifically for IP-to-PSTN voice gateways • Client/Server (Master/Slave) Protocol • Client endpoints are non-intelligent • Clients/Slaves must register with Server/Master and
obey instructions – no independent operation
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SCCP – Skinny Call Control Protocol • Designed by Selsius Systems who built CallManager
(acquired by Cisco in 1998) • Based on H.323, but H.323 was too ‘fat’ with too many chatty
messages, thus the ‘skinnier’ protocol • Cisco proprietary, though some of base of language is open
to public to understand and use, but not change (Asterisk uses)
• Specifically used for Cisco devices such as IP phones, analog gateways, voice ports (Unity, Unity Connection)