ONT Notes – VOIP Networks

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

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

Aaron Conaway

I shake my head around sometimes and see what falls out. That's what lands on these pages.

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