Voice & Video Design Dr. Nawaporn Wisitpongphan 1

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Voice & Video Design

Dr. Nawaporn Wisitpongphan

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Traditional Voice Architecture

PBX (private branch exchange) route voice using TDM -- between company’s branches (on-net)-- between company and PSTN (off-net)

**PSTN = Public Switch Telephone Network

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PBX and PSTN Switches

• The PBX is located in the enterprise’s data center. • Each PBX up to thousands of phones.• Companies can deploy PBX networks to obtain enterprise

feature such as • extension dialing,• dialing privilege control• voice mail• Transfers• Conferencing• Private Line routing

• Companies with multiple large locations and lots of intersite calling can implement tie lines to reduce long distance charges.

• no toll charges• fixed costs associated with the circuits, which are provided by network

carriers/phone companies.

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Local Loop & Trunk

Local Loop: Pair of wires connects Central Office (CO) to office.

■ Interoffice trunk connects two CO switches. Also called a PSTN switch trunk.■ Tandem trunk connects central offices within a geographic area.■ Toll-connecting trunk connects the CO to the long-distance office.■ Intertoll trunk connects two long-distance offices.■ Tie trunk connects two PBXs. Also called a private trunk.■ PBX-to-CO trunk or CO-to-PBX business line connects the CO switch to the enterprise PBX.

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Voice Ports

• Foreign Exchange Station (FXS) – the port that actually delivers the

analog line to the subscriber. In other words it is the ‘plug on the wall’ that delivers a dialtone, battery current and ring voltage

• Foreign Exchange Office (FXO) – the port that receives the analog line. It

is the plug on the phone or fax machine, or the plug(s) on your analog phone system. It delivers an on-hook/off-hook indication (loop closure).

• Ear and Mouth (E&M) – connects private switches. It is an

analog trunk used to connect to a voice switch; it supports tie-line facilities or signaling between phone switches. E&M can be connected with two-wire and four-wire. E&M is also called Earth and Magnet.

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Voice Ports

• Channelized T1 (or E1) connect PBXs and the PSTN or 2 PBXs. There are 2 different formats:– Channel Associated Signaling (CAS) circuits provide 24 (for

T1) or 32 (for E1) channels (1 per DS0)• Signaling (dialed digits, caller ID, and so on) occurs in band along with

voice traffic• Signal is not far away from voice.. So sometimes called robbed-bit

signaling• It reduces data rate to 56 Kbps

– Common Channel Signaling (CCS) circuits also use T1/E1 circuits.• Signaling is done on a separate channel.• Provide more robust features• ISDN PRI uses CCS• Preferred connection between PSTN-to-PBX / PBX-to-PBX

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Major Analog and Digital Signaling Types

• Signaling provides state of the telephone:– Supervisory: on-hook, off-hook– Addressing: dialed digits– Informational: dial tone, busy tone, progress

indicator

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Loop-Start Signaling

Use between the telephone set and the CO, PBX, or FXS module.

Analog signaling technique used to indicate on-hook and off-hook conditions in the network

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Ground-Start Signaling

Alternative Analog signaling technique used to indicate on-hook and off-hook conditions.

Use in switch-to-switchconnections

Ground start requires the closing of the loop at both locations. It is commonly used by PBXs.

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E&M Signaling

• Often used in PBX-to-PBX tie lines• E&M is receive & transmit (Ear & Mouth)• Three forms of E&M dial supervision signaling to seize the E&M

trunk:– Immediate start:

• The originating switch goes off-hook, waits for a finite period of time (for example, 200 ms), and then sends the dial digits without regard for the far end.

– Wink start: • The originating switch goes off-hook, waits for a temporary off-hook pulse

from the other end (which is interpreted as an indication to proceed), and then sends the dial digits.

– Delay dial: • The originating side goes off-hook, waits for about 200 ms, and then checks

whether the far end is on-hook. If the far end is on-hook, it outputs dial digits. If the far end is off-hook, it waits until it goes on-hook and then outputs dial digits.

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Other Signaling

• Q.SIG : – signaling protocol used between PBX switches

• SS7:– Global ITU standard used between PSTN

switches– Implements call setup, routing, and control.

• Addressing Digit Signaling– Pulse or rotary dialing– Dual-tone multifrequency (DTMF) dialing

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DTMF Frequencies

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Voice Engineering Terminology

• Grade of Service (GoS)– Probability that a call will be blocked by a voice gateway when attempting

to seize a circuit during the busiest hour.– P.01 GoS indicates a 1 percent probability of callers being blocked.– Sometimes called “Blocking Probability”

• Erlangs– Describe aggregate trunk usage– Example: a group of users makes/receives 20 calls in the average

busiest hour and each call lasts and average of 10 minutes, the Erlangs are calculated as follows:

• 20 calls per hour * 10 minutes per call = 200 minutes per hour • Traffic volume = (200 minutes per hour) / (60 minutes per hour) = 3.33 Erlangs

• Call Detail Records (CDR)– Call time, call duration, source phone number, dialed phone number,

amount billed– For Voice-over-IP, CDR also indicates source/destination IP

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Erlang Model

• Erlang B:– Use to estimates the amount of trunking (channels) for the

busiest hours.– Example: 3.33 Erlangs, GoS of 1% = 9 Erlang B 9 trunks is

required• Extended Erlang B:

– Add “retry” percentage. Assume that some blocked or failed calls will be reattempted.

– Results in more trunks.• Erlang C:

– Queue excess calls instead of blocking.– Use to calculate the number of agents required in a call.– This model is used to estimate the number of agents required in

a call center.

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Mr. Erlang started it all!!

Provide one trunk for every local phone line, so no call is ever blocked.

Provide just one trunk, and let callers wait until it’s available.

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Why Erlang B?

• Erlang B is derived from M/M/c/c queuing model

We get 3,200 calls a day (during 8-hr work time). That’s 400 calls an hour. Each call lasts three minutes, so each person can handle 20 calls an hour. So we’ll need 20 incoming lines and 20 people to answer the phones…. What’s wrong with my Logic???

Calls are not evenly distributed

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M/M/c/c

• M:Poisson arrival (arrival rate = calls/hr)• M:Poisson departure (departure rate = calls/hr)• Probability of occupying the channel is exponentially distributed• C: Finite number of channels (trunks)• C: Number of queue in the system (No queue for blocked calls)

0 1 c2

2

c

GoS

kA

CA

BlockingC

k

k

C

0 !

!]Pr[

(Trunks) Channels ofNumber

duration) (call timeholding call Average

rate arrival callmean total

C

H

HA

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System Capacity

• Each user generates a traffic intensity Au Erlangs given by:

Au = uHu

where Hu = Average duration of a call per user (hrs/call)

u = Average # of call requests per unit time per user• In a system of U users, the total offered traffic intensity (for an

unspecified # of channels)

A = UAu

• In a C channel trunked system, if the traffic is equally distributed among the channels, then

Traffic intensity / Channel = UAu / C

• Maximum possible CARRIED TRAFFIC = C (in Erlangs)• If offered traffic > max. capacity of the system PROBLEM

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Erlang B Table

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Example 1

An urban area has a population of 2 million residents. Three competing trunked mobile networks (systems A, B, and C) provide cellular service in this area. System A has 394 cells with 19 channels each, system B has 98 cells with 57 channels each, and system C has 49 cells, each with 100 channels. Find the # of users that can be supported at 2% blocking if each user averages 2 calls per hour at an average duration of 3 min. Assuming that all 3 trunked systems are operated a t maximum capacity, compute the percentage market penetration of each cellular provider.

• Solution:– System A:

• Given: Pr. Blocking = 2% = 0.02, # of ch.’s /cell, C = 19• Traffic intensity per user, Au = H = 2 x (3/60) = 0.1 Erlangs• For GoS = 0.02 and C = 19, from the Erlang B chart, the total carried traffic,

A, is obtained as 12 Erlangs.• Therefore, the number of users that can be supported per cell is:

U = A/ Au = 12/0.1 = 120• Since there are 394 cells, the total # of subscribers that can be supported by

system A is equal to 120 x 394 = 47280

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Example 1 (cont)

– System B:• Given: Pr. Blocking = 2% = 0.02, # of ch.’s /cell, C = 57• For GoS = 0.02 and C = 57, from the Erlang B chart, the total carried traffic,



system B is equal to 450 x 98 = 44100. – System C:

• Given: Pr. Blocking = 2% = 0.02, # of ch.’s /cell, C = 49• For GoS = 0.02 and C = 49, from the Erlang B chart, the total carried traffic,



system A is equal to 880 x 49 = 43120.– Therefore, total # of cellular subscribers that can be supported by these 3

systems are 47280+44100+43120 = 134500 users.

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Example 1 (cont2)

– Since there are 2 million residents in the given urban area and the total number of cellular subscribers in system A is equal to 47280, the percentage market penetration is equal to:

47280/2000000 = 2.36%– Similarly, the percentage market penetration of system B is equal to:

44100/2000000 = 2.205%– and the percentage market penetration of system C is equal to:

43120/2000000 = 2.156%– The market penetration of the three systems combined is equal to:

134500/2000000 = 6.725%

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M/M/c Model (Erlang C)

• Blocked calls delayed (Erlang-C) Model• In Erlang-C model a queue is provided to hold calls which are blocked

– If a channel is not available immediately, the call request may be delayed until a channel becomes available

– Probability of not having having immediate access

– It can be shown that the AVERAGE DELAY D for all calls in a queued system is:

1

0 !1!

]0Pr[C

k

kC

C

kA

CA

CA

Adelay

AC

HdelayD

]0Pr[

0 1 c+1c2

c

2

c

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Unified Network

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VoIP

■ Data is primary traffic on many voice service provider networks.■ Companies want to reduce WAN costs.■ PSTN architecture is not flexible enough to accommodate data.■ PSTN cannot integrate voice, data, and video.

With separate voice & data network

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Converged VoIP Network

■ To provide the same reliability and high availability traditionally associated with older voice technologies■ To offer lower cost of ownership than traditional telephony■ To offer greater flexibility than traditional telephony (remote worker, mobility)■ To leverage integration to provide new applications (presence, IVR, contact centers)■ To improve remote worker, agent, and work-at-home staff productivity■ To ensure backward compatibility with traditional systems and endpoints such as PBXs, faxes, and the PSTN

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Single-Site Deployment

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Multisite WAN with Centralized Call Processing Model

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Multisite WAN with Distributed Call Processing Model

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Sensitivity to Packet Loss

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Video Media Application Models

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Codecs

• Analog VoiceDigital Signal– Filtering (Human’s speech Frequency 300Hz-3400Hz)– Sampling (According to Nyquist’s Theorem @ 2 Max Freq)– Digitizing or Quantizing– Encoding

• Pulse-Code Modulation (PCM) (G.711)– Analog speech is sampled @8000 Hz– Quantized at 8 bits/sample– Thus, PCM produces 80008 = 64 kbps– Used as the primary with IPT over LANs where high bandwidth

is available.

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Codec Standard

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VoIP Control and Transport Protocol

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Other Related Protocols:DHCP, TFTP, SCCP

• Dynamic Host Configuration Protocol (DHCP): Used to provide device configuration parameters such as IP configuration (address, subnet mask, default gateway) and TFTP servers’ IP.

• TFTP: To obtain ring tones, backgrounds, configuration files, and firmware files.

• Skinny Client Control Protocol (SCCP): Used for call control for Cisco IP phones (Cisco proprietary). – Runs over TCP– Use less overhead than H.323.– IP Phone use SCCP to register with CUCM and to establish calls

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RTP vs. RTCP

• Real-time Transport Protocol (RTP): For voice stream (VoIP) station-to-station traffic in an active call. (Runs over UDP)

• Compressed RTP (cRTP)– Header Compression done on each link on hop-by-hop basis.– cRTP is recommended for slow link up to 768kbps.– Don’t really get used that much now since slow link is rare

• Real-time Transport Control Protocol (RTCP): For RTP control and reporting (accompanying stream to RTP between endpoints).

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MGCP

• Media Gateway Control Protocol (MGCP): A client/server protocol for control of endpoints and gateways. In the MGCP model, intelligence resides on the call agent (server), and the device is controlled by the agent.

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H.323

Framework for multimedia protocols for use over packet switched networks

• Terminals: Telephones, video phones, and voice-mail

• MCUs: a device used for joining together multiple audio/video streams into a single bridge, or conference. The MCU is responsible for taking streams from the different conference participants, mixing the streams together, and then sending the combined stream back to the participants.

• Gateways: A device that connects a VoIP network to a TDM network such as the PSTN (analog/T1 PRI). Gateways also provide translation services between H.323 endpoints and non-H.323 devices.

• Gatekeepers: Provide dial plan unification, CAC, device registration, and call routing services to the VoIP network. Gatekeepers are recommended when interconnecting more than two CUCM networks.

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Gatekeeper Provides Scalability

For large network: If a gatekeeper is not used, logical connections need to be configured on each gateway to connect to every single other gateway on the network.

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1

NNL

L = # of Logical ConnectionsN = # of Devices

Example: In a network with 7 devices, 21 logical connections must be configured to ensure that each device can communicate with every other device.

In a gatekeeper-controlled system, each device needs to be configured with only a logical connection to the gatekeeper.

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Protocols used with H.323

• H.245 call capability control– specifies messages for opening and closing

channels channels for media streams

• Q.931 call setup signaling

• H.225 call signaling– performs registration, admission, and status

(RAS) signaling for H.323 sessions.

• RTP/RTCP voice streams

• H.264: ITU-T standard that defines video compression algorithm.– identical to ISO/IEC MPEG-4 Part 10 and also

called Advanced Video Coding (AVC).– Upgradefrom H.263 – Found in Flash, YouTube, Google Video

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Session Initiation Protocol (SIP)

• Alternative to H.323. (More simple & lighter than H.323)

• A standard for VoIP networks defined by the IETF and used for gateways and endpoints.

• SIP is feature rich (native IM, presence, and video support), lightweight, and designed for easy troubleshooting (ASCII-based messages).

■ Resource Reservation Protocol (RSVP) for reserving network bandwidth and priority (low-latency) queuing■ RTP and RTCP for transporting real-time data and providing QoS feedback■ Real-Time Streaming Protocol (RTSP) for controlling delivery of streaming media■ Session Announcement Protocol (SAP) for advertising multimedia sessions via multicast■ Session Description Protocol (SDP) for describing multimedia sessions

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IP Telephony Design: Bandwidth

• G.729 call use 26 kbps• G.711 call use 80 kbps

• Bandwidth Estimation & Best Practice– Depends on codec used, Layer 2 protocol, whether voice-activity

detection (VAD) is enabled– VAD can suppress silence duration (saving at least 35% of the

bandwidth). – VAD is avoided in practice because of the quality issue– In the design, total bandwidth of voice, video, and data < 75% of

available bandwidth– Should not reserve more than 1/3 of the link for prioritized voice/video

traffic

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Voice Bandwidth Requirement with VAD and cRTP

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Calculating Voice Bandwidth

Assumptions• IP/UDP/RTP header uses 40 bytes.

• cRTP reduces the IP/UDP/RTP header to 2 or 4 bytes.

• The WAN Layer 2 header adds 6 bytes on a point-to-point circuit.

• Voice packet size = (Layer 2 header) + (IP/UDP/RTP header) + (voice payload).

• Voice packets per second (pps)

= codec bit rate / voice payload size.

• Voice bandwidth (bps)

= (voice packet size) * (pps).

Calculate the WAN bandwidth used at a site that will have 10 concurrent G.729 calls with cRTP and a default voice payload of 20 bytes.

• G.729 codec is used: 8 kbps codec bit rate.• cRTP = 2-byte IP/UDP/RTP header.• Default voice payload= 20 bytes * (8 bits/bytes) =

160 bits.• WAN header = 6 bytes.

• Voice packet size = 6 bytes + 2 bytes + 20 bytes = 28 bytes * (8 bits/byte) = 224 bits.

• PPS = 8 kbps / 160 bits = 8000/160 = 50 pps.

• BW per call = 224 (bits/packet) * 50 (pps) = 11200 bps = 11.2 kbps.

• BW for 10 calls = 11.2kbps * 10 = 112 kbps.

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Delay Component in VoIP• ITU’s G.114 recommendation specifies that the one-way delay between endpoints

– Commercial voice quality: < 150 ms– Usable voice quality: 150ms - 400ms– Unacceptable quality: > 400 ms

• Delay Components– Propagation delay– Processing delay (and packetization)– Serialization delay

• Variable delays are– Queuing delay– Jitter buffer delay

Processing delay includes coding, compression, decoding, and decompression delays.

- G.729 has a delay of 15 ms - G.711 PCM has a delay of 0.75 ms.

Serialization delay = frame size in bits / link bandwidth in bps

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Network Delay Summary

A reasonable design goal is to keep the serialization delay experienced by the largest packets or fragments on the order of 10 ms at any interface.

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Packet Loss

• It causes voice and video clipping and skips.

• Causes:– congested links– improper QoS configuration,– bad packet buffer management, – routing issues. – packets received outside of the dejitter buffer

range, which are packets that are discarded.

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Echo Cancellation

• Echo cancellation Process– recognizing the originally transmitted signal that

reappears, with some delay, in the transmitted or received signal.

– Once the echo is recognized, it can be removed by subtracting it from the transmitted or received signal.

• ITU-T defines that – Echo delays more than 15 ms should be suppressed

with echo cancellers. – Echo delays up to 15 ms do not need to be

suppressed.

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QoS

• Classification: Process that identifies the class or group a packet belongs to.• Marking: Process of marking packets with differentiated service codepoint

(DSCP) values for QoS.• Congestion avoidance: Using Weighted Random Early Discard (WRED) and

Distributed WRED (DWRED).• Traffic conditioners:

– Traffic Shaper: delays excessive traffic by using a buffer or queuing mechanism– Traffic Policer: drops traffic or reclassifies excessive traffic to a lower priority.

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QoS and Bandwidth Management

• cRTP: – compress the header

• IEEE 802.1P – OSI Layer 2 standard for prioritizing network traffic at the data link/MAC sublayer.– IEEE 802.1P is a spin-off of the 802.1Q VLAN trunking standard. (Define 3-bit Prioritization field in

802.1Q)• RSVP

– reserves bandwidth for the application– resources is reserved in each node along the data path.

• Link Fragmentation and Interleaving (LFI)– QoS mechanism used to reduce the serialization delay.– If the large data packet arrives at the interface first, the VoIP packet has to wait until the large data

packet has been serialized.– Large data packet is fragmented into smaller packets, the VoIP packets can be interleaved

between the data packets.

• FRF.12– Similar to LFI, but is used in Frame Relay Networks

• Low Latency Queueing (LLQ)– Provide strict priority queue for VoIP Traffic

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LFI

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LLQ

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IP Telephony Design Recommendation

• Use separate VLANs and IP subnets for IP phones and data to provide ease of management and simplified QoS configuration.

• Use private IP addresses for IP phones subnets to allow for more security to voice devices.

• Place CallManager and Unity servers on filtered VLAN/IP subnets in the server access in the data center.

• Use IEEE 802.1Q trunking and 802.1P to allow for prioritization at Layer 2.

• Extend QoS trust boundaries to voice devices but not to PCs and other data devices.

• In the access layer, use multiple egress queues to provide priority queuing of RTP voice streams.

• Use DSCP for classification and marking. [Differentiated Service Code Point]• Use LLQ on WAN links.• Use LFI on WAN links less than 768 kbps.• Use CAC to avoid oversubscription of circuits.

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Service Classes (as specified in RFC 5865)

■ Network Control: For routing and network control functions■ Operations, Administration, and Management (OAM): For network configuration and management functions■ Telephony: Includes VoIP and circuit emulation■ Signaling: For peer-to-peer and client/server signaling, such as SIP, MGCP, H.323, and H.248■ Multimedia Conferencing: For applications that can change their encoding rate, such as H.323/V2■ Real-Time Interactive: For RTP/UDP streams for video conferencing applications that cannot change the encoding rate

■ Multimedia Streaming: For variable-rate elastic streaming media applications and webcasts■ Broadcast Video: For inelastic streaming media with low jitter and low packet loss, such as broadcast TV, video surveillance, and security■ Low-Latency Data: For data processing applications, such as web-based ordering■ High-Throughput Data: For store-and-forward applications, such as FTP■ Standard: For traffic that has not been identified for any preferential treatment■ Low-Priority Data: For traffic types that do not required any bandwidth assurance

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Cisco’s Setting Recommendation

**PHB = per-hop behavior

Documents

Voice & Video Design Dr. Nawaporn Wisitpongphan 1