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Here are some of the notes I’ve been taking while reading over the ONT book. I hope it benefits somebody.  Feel free to correct any stupid mistakes as a paraphrase to avoid a lawsuit.

There’s way too much info here.  I’ll refine the process a little better for the next topics.

Benefits of Packet Telephony Networks

• More efficient use of bandwidth and equipment – Packet telephony networks don’t dedicate channels or a static bandwidth to a call; it’s just another network application.
• Consolidate network expense – The common infrastructure (IP-based networks) keeps you from having to support another distinct network for voice like in traditional PBX implementations.
• Improved employee productivity – The phone can be used for more than just phone calls by utilizing the XML interface to run applications or provide content from the network.
• Access to new communications devices – IP phones can communicate with computers, network gear, PDAs, etc., and not just the PBX.

Packet Telephony Components

• Phones – These include analog phone, digital phones, IP phones, softphones, etc.
• Gateways – These devices connect the different devices that cannot access the IP network.  For example, making a 911 call from your IP phone requires a gateway that switches and converts your VOIP conversation to the PSTN.
• Gatekeepers – These are devices that handle call routing (resolving an IP to an extension/phone number) and call admission control (CAC, grants permission to make the call).
• Multipoint control units (MCUs) – These are conference bridges that connect a bunch of streams together and present it to all participants.  Some can do video as well.
• Call agents – These are devices used in a centralized model that handle the call routing, address translation, call setup, call maintenance, and call termination.
• Application and database servers – These provide required and optional services to the packet telephony network and include TFTP servers for configuration and OS download and XML servers for application use.
• Digital signal processors (DSPs) – These guys converts signals from one form to another.  They convert analog to digital signals, digital to packetized data in the form of a codec, from codec to codec, etc.

Analog Interfaces

• Foreign Exchange Office (FXO) – These are interfaces that expect to connect to a CO or equivalent.  You connect these to your wall jack to get access to the PSTN.
• Foreign Exchange Station (FXS) – You connect your analog devices (phones, modems, faxes, etc.) to these guys to get dial tone.
• Ear and Mouth (E&M) – These are the old-school way to connect PBXes together.

Digital Interfaces

• Basic Rate ISDN (BRI) – These give you 2 64kbps channels (bearer channels) to run voice over.  It also includes a 16kbps D (delta) channel with 48kbps of framing overhead to give you 192kbps.

• T1 (North America) – This is a channelized T1 or a Primary Rate ISDN (PRI).
  • Common Channel Signaling (CCS) – The D channel is dedicated to signaling, giving you 23 64kbps channels.
  • Channel Associated Signaling (CAS)  – There is no D channel, but every bearer channel dedicates a few data bits for its own signaling.

• • E1 (North America) – This is a channelized E1 or a Primary Rate ISDN (PRI).
    • Common Channel Signaling (CCS) – The D channel is dedicated to signaling, giving you 30 64kbps channels.
    • Channel Associated Signaling (CAS)  – There is still a dedicated D channel, so you still have 30 64kbps channels to use.

VOIP Signaling

• H323. – ITU Standard that uses a whole mess of RFCs; distributed model
• Media Gateway Control Protocol (MGCP) – IETF RFC 3435; centralized model
• Session Initiation Protocol (SIP) – IETF standard; distributed model

Phone Call Stages

• Call setup – connects the call between the endpoints
  • Call routing – figures out where the call is going
  • CAC (optional) – Do you have enough resources (i.e., an available channel or bandwidth) to make the call?
  • Call negotiation – negotiates the source and destination IPs, source and destination UDP ports, and codec.
• Call maintenance – collects call statistics for on-demand or historical use
• Call teardown – hanging up and terminating the connection

Digitizing Analog Signals

• Sampling – Periodic capturing and recording of voice resulting in a pulse amplitude modulation (PAM) signal
• Quantization – Assigning numerical values to the PAM signal
• Encoding – Converting the quantization to binary
• Compression (optional) – compressing the binary stream
• Pulse code modulation (PCM) converts analog to digital, but it doesn’t use compression.  It takes 8000 samples per second and converts each sample to an 8-bit number, giving 64kbps of capacity.

Digital to Analog

• Decompression (optional)
• Decoding and filtering – binary is converted back to a PAM signal; filtering removes any noise from the conversion
• Reconstructing the analog signal

The Nyquist Theorem

• The number of samples required to accurately encode (and decode) a signal is twice the highest frequency of the signal.
• Since telephone lines can only transmit up to 3400 Hz (4000 Hz for simplicity), the sample rate should be 8000 samples/second.

Measuring Compression Qualities

• Mean opinion score (MOS) – ITU standard technique for measuring quality of codec; subjective score from 1 to 5
• Perceptual speech quality measurement (PSQM) – Another ITU standard technique for measuring quality of codec; test equipment score from 0. to 6.5
• Perceptual analysis measurement system (PAMS) – Developed by BT; predictive system
• Perceptual evaluation of speech quality (PESQ) – Another ITU standard; combines PSQM and PAMS; objective measurement of factors including subjective values

Digital Signal Processors (DSPs)

• Provide 3 major services – voice termination, transcoding, conferencing
• Also performs compression (codec), echo cancellation, voice activity detection (VAD), comfort noise generation (CNG), and jitter handling
• Conferencing among participants with the same codec is called a single-mode conference.
• Conferencing among participants with different codecs is called a mixed-mode conference.

Protocols

• VOIP calls run over Real Time Protocol (RTP).
• RTP provides sequence reordering, time-stamping, and multiplexing
• Rides on UDP ports 16384-32767
• Voice does not need the reliability (retransmission) of TCP since retransmitted data is no longer useful (I already said that).
• VOIP packets headers:
  • IP – 20 bytes
  • UDP – 8 bytes
  • RTP – 12 bytes
  • L2 headers vary depending on technology (Ethernet = 12 bytes, MPLS, etc.)
• 2 10-ms packages are usually in one packet (20ms of voice)
• G.711 (64kbps) produces 160 bytes from 20 ms of voice.
• G.729 (8kbps) produces 20 bytes from 20 ms of voice.

cRTP

• Compressed RTP (cRTP) reduces the headers
• After the first packet lands, the IP, UDP, and RTP headers won’t change, so why send them again?
• The headers are reduced to a hash.
• cRTP reduces the headers to 4 bytes with a UDP checksum and 2 bytes without a UDP checksum.
• Slow links only
• Processing overhead
• Finite delay in packetization

Packet Size Effect on Bandwidth

• The size of a voice frame depends on:
  • Packet rate and packetization size – rate is inversely proporational to size
  • IP overhead – RTP, UDP, IP, cRTP overhead
  • L2 overhead -
  • Tunneling overhead – IPSec, GRP, MPLS, etc.
• Codecs have different bandwidth
  • G.711 (PCM) – 8000 samples per second @ 8 bits per sample = 64 kbps
  • G.726 (Adaptive Differencial PCM – ADPCM) – Variable bit rate of 32 kbps, 24 kbps, or 16 kbps
  • G.722 (Wideband Speech Encoding) – 2 subbands using modified ADPCM of 64 kpbs, 56kbps, or 48 kbps
  • G.728
  • G.729 – 10 samples per 10-bit code = 8 kbps

Calculating Total Bandwidth

• Step 1 – Determine codec and packetization period: What does the codec require in bandwidth?  How many samples per packet (usually 2)?
• Step 2 – Determine link-specific overhead:  Encapsulation?  cRTP?
• Step 3 – Calculate packetization size:  Size of voice payload; codec bandwidth * packetization period / 8 = voice payload in bytes
• Step 4 – Calculate total frame size: IP + UDP + RTP + Tunneling + data link + packetization size
• Step 5 – Calculate packet rate: 1 / packetization period (ex., 20ms packetization period is 1/0.020 = 50 packets per second)
• Step 6 – Calculate total bandwidth:  Total frame size * packet rate

VAD and Bandwidth

• Common for 1/3 of conversation to be silence
• VAD bandwidth savings depends on:
  • Type of audio: regular phone call (two-way), conf call (one-way), music on hold (MOH)
  • Background noise: noise may be detected as voice
  • Other factors:  language, culture may influence amount of silence

Enterprise VOIP Implementations

• Consists of gateways, gatekeepers, Cisco Unified CallManagers (CCM), Cisco IP Phones
• Routers can provide the voice gateway function by connecting the IP network to the WAN (and other gateways), PSTN, PBXes, etc.
• Survivable Remote Site Telephony (SRST) allows local calling and use of PSTN while services are down

Functions of CCM

• Call processing – routing, signaling, accounting
• Dial plan administration -  call routing
• Signaling and device control – configuration and instruction in case of events
• Phone feature administration – button programming, profiles, etc.
• Directory and XML
• API for interface – allows custom programming for IP phones

Enterprise Deployment Models

• Single-site: You have one site, and everything is there.
• Multisite with centralized call processing: You have multiple sites, but the main site has the CCM cluster.
• Multisite with distributed call processing: You have multiple sites, and each site has its own CCM cluster.
• Clustering over WAN: You have multiple sites, and each site has a part of one big CCM cluster.

IOS Voice Commands

----- R1 -----
! FXS on 1/1/2
Dial-peer voice 1 POTS
 destination-pattern 120
 port 1/1/2

! Extension 230 is on R2
Dial-peer voice 2 R2
 destination-pattern 230
 session target ipv4:10.1.1.2

----- R2 -----
! FXS on 2/2/1
Dial-peer voice 1 POTS
 destination-pattern 230
 port 2/2/1

! Extension
Dial-peer voice 2 R2
 destination-pattern 120
 session target ipv4:10.1.1.1

Call Admission Control (CAC)

• QoS can guarantee bandwidth but can only reserve so much (say, for 2 simultaneous calls).
• CAC make sure that resources are available (denies a new call if 2 calls are already placed).
• Dropped packets affect every call – not just the new ones

—–

Additional Reading

1. H.323 Sources on Wikipedia
2. MGCP – RFC 3435
3. SIP – RFC 3261
4. Nyquist Theorem on Wikipedia
5. MPLS on Wikipedia