Fundamentals Of Telephony 4l204r
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CCNA Voice A Brief History of Telephony
www.INE.com
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) Copyright © www.INE.com
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 enough TDM ‘channels’ or ‘slots’ for every single phone attached to the system to make a voice call
Copyright © www.INE.com
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 – s 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
Copyright © www.INE.com
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 Copyright © www.INE.com
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
Copyright © www.INE.com
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 circuitswitched 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 Copyright © www.INE.com
(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 Copyright © www.INE.com
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 Copyright © www.INE.com
(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)
Copyright © www.INE.com
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 Copyright © www.INE.com
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)
Copyright © www.INE.com
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
Copyright © www.INE.com
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
Copyright © www.INE.com
CODECs :: G.711 • G.711 – ITU standard – Uses Pulse Code Modulation (PCM) – 64kbs default rate – Excellent audio quality – For LAN connections
Copyright © www.INE.com
CODECs :: G.722 • G.722 – ITU standard – Optimized for wideband speech – 64kbs default rate – Superior audio quality to G.711 – For LAN connections
Copyright © www.INE.com
CODECs :: G.729 • G.729 – 8kbs default rate – High-complexity U / DSP – For WAN connections
• G.729A variant – Medium-complexity U / DSP
• G.729B variant – High-complexity U / 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) Copyright © www.INE.com
CODECs :: iLBC • Internet Low Bitrate Codec (iLBC) – 13.3kbps default rate – Optimized for narrowband speech and lossy WAN connections (e.g. Internet)
Copyright © www.INE.com
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)
Copyright © www.INE.com
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)
Copyright © www.INE.com
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) Copyright © www.INE.com
Real Time Protocol (RTP) • • •
•
•
RTP is the Layer 4 protocol that rides atop of UDP, and encapsulates all delaysensitive traffic, such as Voice and Video UDP Ports range from 16384 – 32767 Real Time Control Protocol (RT) 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, RT 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
Copyright © www.INE.com
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 MG SC
Copyright © www.INE.com
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 to a centralized dial plan server called a Gatekeeper Copyright © www.INE.com
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 to a centralized dial plan server called a SIP Proxy Server (other SIP servers)
Copyright © www.INE.com
MG– 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 with Server/Master and obey instructions – no independent operation
Copyright © www.INE.com
SC – 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) Copyright © www.INE.com
www.INE.com
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) Copyright © www.INE.com
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 enough TDM ‘channels’ or ‘slots’ for every single phone attached to the system to make a voice call
Copyright © www.INE.com
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 – s 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
Copyright © www.INE.com
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 Copyright © www.INE.com
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
Copyright © www.INE.com
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 circuitswitched 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 Copyright © www.INE.com
(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 Copyright © www.INE.com
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 Copyright © www.INE.com
(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)
Copyright © www.INE.com
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 Copyright © www.INE.com
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)
Copyright © www.INE.com
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
Copyright © www.INE.com
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
Copyright © www.INE.com
CODECs :: G.711 • G.711 – ITU standard – Uses Pulse Code Modulation (PCM) – 64kbs default rate – Excellent audio quality – For LAN connections
Copyright © www.INE.com
CODECs :: G.722 • G.722 – ITU standard – Optimized for wideband speech – 64kbs default rate – Superior audio quality to G.711 – For LAN connections
Copyright © www.INE.com
CODECs :: G.729 • G.729 – 8kbs default rate – High-complexity U / DSP – For WAN connections
• G.729A variant – Medium-complexity U / DSP
• G.729B variant – High-complexity U / 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) Copyright © www.INE.com
CODECs :: iLBC • Internet Low Bitrate Codec (iLBC) – 13.3kbps default rate – Optimized for narrowband speech and lossy WAN connections (e.g. Internet)
Copyright © www.INE.com
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)
Copyright © www.INE.com
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)
Copyright © www.INE.com
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) Copyright © www.INE.com
Real Time Protocol (RTP) • • •
•
•
RTP is the Layer 4 protocol that rides atop of UDP, and encapsulates all delaysensitive traffic, such as Voice and Video UDP Ports range from 16384 – 32767 Real Time Control Protocol (RT) 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, RT 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
Copyright © www.INE.com
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 MG SC
Copyright © www.INE.com
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 to a centralized dial plan server called a Gatekeeper Copyright © www.INE.com
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 to a centralized dial plan server called a SIP Proxy Server (other SIP servers)
Copyright © www.INE.com
MG– 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 with Server/Master and obey instructions – no independent operation
Copyright © www.INE.com
SC – 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) Copyright © www.INE.com
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