• Congestion
  • Speed mismatch – traffic leaves a lower-bandwidth interface than the one it came in on
  • Aggregation problem – lots of links with one egress of equal bandwidth
  • Confluence problem – a bunch of traffic needs to egress out of the same interface
• Queuing
  • Transmit queue (TxQ) – hardware queue; there’s only one you can’t touch
  • Software queue – where packets wait to be sent; there are many queue-types that you modified to police traffic
• FIFO
  • If I beat you to the router, I leave the router first.
  • Possible long delays, jitter, and starvation
• Priority queuing (PQ)
  • Four queues
    • High-priority
    • Medium-priority
    • Normal-priority
    • Low-priority
  • Scheduler starts from high and work to low
  • When the high queue is empty, it processes a packet from medium, then starts all over
  • Can you say starvation?
• Round robin queuing (RR)
  • One packet from this queue, one from the next, etc., then start over again
• Custom queuing (CQ)
  • Weighted round robin
  • Queues are given weights (bandwidth guarantees)
• Weighted Fair Queuing (WFQ)
  • Default queuing on slow links ( < E1 )
  • Divides traffic into flows
  • Equal bandwidth is given to each flow
  • Provides faster scheduling to low-volume flows
  • Provides more bandwidth to higher-priority flows
  • Flows identified by a hash
    • Source IP
    • Destination IP
    • Protocol number
    • ToS
    • Source port
    • Destination port
  • Each unique has is a new flow
  • No way to allocate bandwidth among the flows
  • By default, up to 256 queues are made, but that is changeable to a power of 2 between 16 and 4096
  • If the max number of flows is reached, queues are reused for other flows
  • If a queue is full, a packet may be dropped.
  • WFQ early dropping drops packets when the queue reaches the congestive discard threshold (CDT)
  • Advantages
    • Simple configuration
    • No starvation
    • Guarantee processing of all flows
    • Drops packets from big-hitter flows
    • Faster service no low-hitters (interactive) flows
    • Standard on (nearly) all IOS devices
  • Disadvantages
    • Classification and scheduling are not configurable
    • Only on slow links
    • No guarantee of bandwidth or delay
• Class-based Weighted Fair Queuing (CBWFQ)
  • User-defined queues for flexibility
  • Configured with class-maps via MQC
  • Weights are calculated based on values give in class-map
    • Bandwidth – guarantee this much bandwidth
    • Bandwidth percent – give me this much of the available bandwidth
    • Bandwidth remaining percent
  • Advantages
    • User-defined traffic classes
    • Each queue gets its own bandwidth
    • Scalability
  • Disadvantages
    • No delay guarantee (not good for real-time application like voice)
  • Configuring

    class-map TESTCM1
     match access-group 100
    !
    class-map TESTCM2
     match access-group 200
    !
    policy-map TESTPM
     class TESTCM1
      bandwidth 64
     class TESTCM2
      bandwidth 128

• Low-latency Queuing
  • Includes strict priority queue for delay-sensitive data
  • Strict priority queue is policed to avoid starvation of other queues
  • Configured the same way as normal CBWFQ, but with the priority keyword
  • This configuration makes TESTCM2 a priority queue

  class-map TESTCM1
   match access-group 100
  !
  class-map TESTCM2
   match access-group 200
  !
  policy-map TESTPM
   class TESTCM1
    bandwidth 64
   class TESTCM2
    priority bandwidth 128

• Classification is done with traffic desriptors
  • Ingress interface
  • CoS value on ISL or 802.1P frames
  • Source/destination IP address
  • IP Precedence or DSCP value
  • MPLS EXP
  • Application type
• Layer 3 QoS
  • Type of Service (ToS) is 8-bit field.
  • First 3 bits of ToS are the IP precedence.
  • First 6 bits of ToS are the DSCP value.
  • Last 2 bits of ToS are explicit congestion notification (ECN).
• Layer 2 QoS
  • Ethernet
    • Class of Service (CoS)
    • On 802.1P frame
    • 3-bit priority (PRI) field
      • 000 – Routine – Best-effort
      • 001 – Priority – Medium priority
      • 010 – Immediate – High priority
      • 011 – Flash – Call signaling
      • 100 – Flash-Override – Video conferencing
      • 101 – Critical – Voice bearer
      • 110 – Internet – Reserved
      • 111 – Network – Reserved
  • Frame Relay
    • 1-bit discard eligible (DE) field
  • ATM
    • 1-bit cell loss priority (CLP) field
  • MPLS (layer 2 1/2)
    • 3-bit experimental (EXP) field
    • By default, the 3 most significant ToS bits (IP Precedence bits) are copied to EXP
• Per-hop Behavior (PHB)
  • “an externally observable fowarding behavior of a network node toward a group of IP packets that have the same DSCP value”
  • In other words, treat packets with the same DSCP value in the same manner – scheduling, queuing, policing, etc.
  • Behavior aggregate (BA) is a group of packets with the same DSCP value
• DSCP
  • DSCP is chopped up into 4 PHBs
    • Class selector PHB – (000) old IP precedence compatibility
    • Default PHB – (000) best effort
    • Assured forwarding (AF) PHB – (001, 010, 011, 100) guarantee bandwidth
      • Provides 4 queues for 4 classes of traffic (AF1-4)
      • Also specifies drop preference (ex., AF41, A13) where second number is preference (higher is more probable to be dropped)
      • Each queue must have (W)RED to avoid drops
      • No queue is any better than the other
      • Backward compatible with IP precedence
  • • Expedited forwarding (EF) PHB – (101) low delay
      • Minimum delay
      • Bandwidth guarantee
      • Policing
• Trust boundaries
  • Establish DSCP values as close to the source as possible
    • On the device (IP phone), access switch, or distribution switch
    • The core should never assign DSCP values
  • Only trust DSCP values from devices you trust
  • Examine and rewrite values from untrust sources
• Network-based Application Recognition (NBAR)
  • Protocol discovery – discovers what protocols you’re running on your network
  • Traffic statistics collection – keeps tracks of stats on each protocol
  • Traffic classification – NBAR protocols can be used in class-maps to define traffic to be services
  • Packet description language models (PDLMs) – table of what protocols NBAR recognizes
  • Limitations
    • Doesn’t work on EtherChannel interfaces
    • Only handles 24 URLs, hosts, or MIME types
    • Only analyzes first 400 bytes of the packets
    • Requires CEF
    • Doesn’t work on HTTPS, multicasts, or fragments
    • Ignored traffic destined for the router itself
  • NBAR commands
    • Router(config)# ip nbar pdlm pdlm-name : Update the PDLM table
    • Router(config)# ip nbar port-map protocol-name [tcp|udp] port-number : Adds an entry to the PDLM table
    • Router# show ip nbar port-map protocol-name : Shows what’s in the PDLM table
    • Router# show ip nbar protocol-discovery : Shows what’s been discovered
    • Router(config-cmap)# match protocol name : a class-map match for an NBAR-discovered protocol
  • Special protocol matching
    • Can match beyond the port number with deep packet inspection
    • Matches HTTP hostname, URL, or MIME type
    • Matches fast-track P2P
    • Matches RTP content

Major issues for converged enterprise networks

• Available bandwidth: competition among applications
  • Fixes
    • Increase bandwidth: More power!
    • Properly queue based on classification and marking: QoS
    • Compress: cRTP, TCP header compression, etc.
• Delay: Lead time to get a packet to the destination
  • Types of delay
    • Processing delay: routing, switch delay
    • Queuing delay: how long a frame stays in an output queue
    • Serialization delay:  how long to put the frame on the wire
    • Propagation delay: the time to cross the physical medium
• Jitter (delay variation): Variation is the delay
  • Different delays mean different arrival times
  • De-jitter buffers save up packets to reduce jitter (like the old CD writers)
  • Fixes
    • More bandwidth
    • Prioritize sensitive data and forward first
    • Remark (reclassify) packets based on sensitivity
    • Enable L2 payload compression: make sure compression delay isn’t worse than the jitter
    • Use header compression
• Packet loss: Packets are lost in the network somewhere
  • Fixes
    • More bandwidth
    • Increase buffers space: more room for the queue on the interface
    • Provide guaranteed bandwidth: Queuing and QoS
    • Congestion avoidance
      • Random Early Detection (RED) and weighted RED (WRED) drop packets before the queue is full
      • Selective dropping is better than FIFO or LIFO dropping

QoS History

• Priority queuing: gives certain data the right-of-way for transmission
• Weighted Fair Queuing (WFQ): prevents small packets from waiting too long for big packets
• RTP priority queuing: Gives voice packets the right-of-way
• CAC: Makes sure we don’t fill up the queue or pipe with voice traffic

Implementing QoS

• Step 1: Identify traffic types and requirements
  • Network audit
  • Business audit
  • Define bandwidth requirements for each class found
• Step 2: Classify the traffic
  • Common classes
    • VOIP
    • Mission-critical
    • Signal traffic: for VOIP
    • Transactional application: SAP, ERP
    • Best-effort: Everything else
    • Scavenger: Crap you don’t care about like P2P and your boss’s email
• Step 3: Define policies for each class
  • Tasks for each class
    • Set max bandwidth
    • Set min bandwidth
    • Assign relative priorities
    • Apply congestion avoidance, congestion management, etc.

QoS Models

• Best-effort: no QoS
  • Scalable
  • Easy
  • No service guarantee: doesn’t care what you’re trying to do
  • No service differentiation: all traffic is equal
• Integrated Service (IntServ)
  • Hard-QoS
  • Uses RSVP to guarantee bandwidth through the entire path
  • Requires
    • Admission control
    • Classification
    • Polices the traffic (ceiling)
    • Queuing
    • Scheduling
  • Advantages
    • End-to-end resource admission control
    • Per-request policy admission control
    • Signaling of dynamic ports
  • Disadvantages
    • Continuous signaling
    • Not scalable
• Differentiated Services (DiffServ)
  • Soft-QoS
  • Configured on each hop
  • Traffic is classified
  • Enforces different treatment on different classes
  • Defined based on business requirements
  • Benefits
    • Scalable
    • Supports lots of service levels
  • Drawbacks
    • No absolute guarantee of service
    • Complex configuration throughout network

QoS Implementation Methods

• CLI
  • Old school
  • Not used any more
• Modules QoS CLI (MQC)
  • Step 1: class-map
  • Step 2: policy-map
  • Step 3: service-policy
• AutoQoS
  • Automatically generates classes and policies based on traffic it sees
  • Super-simple
  • Requires CEF, NBAR, and correct bandwidth statements
• SDM QoS Wizard
  • Next, next, next
  • Can be used to implement, monitor, or troubleshoot QoS

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