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.
- 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.
- 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.
- E1 (North America) – This is a channelized E1 or a Primary Rate ISDN (PRI).
- 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.
- 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.
- 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.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
- H.323 Sources on Wikipedia
- MGCP – RFC 3435
- SIP – RFC 3261
- Nyquist Theorem on Wikipedia
- MPLS on Wikipedia
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