Packet Pushers Podcast Episode 11: If You Can’t Be Replaced, You Can’t Be Promoted

Study Questions

• The obvious first question involves the common LSA types and their function.  Can you list them?

Type-1 – Router – Lists each router their connected IP addresses
Type-2 – Network – Lists all the transit, or multiaccess, networks
Type-3 – Net Summary – Defines a  host route for interarea routes; this is from the ABR
Type-4 – ASBR Summary – Defines a host route for an external (to OSPF) route; this is from an ASBR
Type-5 – AS External – Lists the networks advertised into OSPF from external sources (redistribution)
Type-7 – NSSA External – External routes injected into a not-so-stubby area

• What information about the OSPF area does a router’s OSPF database contain?

Everything, basically.  The database includes all the routers in the area, the IPs of every OSPF-enable interface, all the networks, and the costs of each hop.  The SPF algorithm uses all this information to figure out the best path to all networks advertised.

• Define the reference bandwidth in OSPF.  What’s the default value, and how do you change it?

The reference bandwidth is the bandwidth that is divided by an interface bandwidth to get a the cost for that link.  The default value is 100Mbps, which can be changed with the auto-cost reference-bandwidth OSPF subcommand.

• How can you override the cost on an interface?

You can change the reference bandwidth, but that will change the costs of all interfaces.  The ip ospf cost X command on an interface will do the trick.  You can also give it the old bandwidth change (the one that tends to break or influence other things).

• What is the formula for calculating the cost of an interface?

reference bandwidth / interface bandwidth

• What are the five OSPF message types, and what do they do?

Hello – Establishes neighbor relationships
Database Description (DBD) – Send summaries of the LSAs a router has
Link State Requests (LSR) – Sent to a router to ask for more details on an LSA
Link State Update (LSU) – Reply to an LSR that includes the details of the requested LSA
Link State Acknowledgment (LSAck) – An acknowledgement of the DBD

• How often does a router send its full OSPF database to its neighbors?

It doesn’t.  It does, however, send any self-originated LSAs (LSAs that it generated) every 30 minutes (1800 seconds).

• R1 is an ABR to area 1 with an area 0 route to a network with a cost of 100.  R2, also the same ABR setup, advertises the same route to area 1 (and, thus, R1) with a cost of 10.  Which route does R1 take?

ABRs always take an intra-area route over and interarea route, so the path with a cost of 100 will be chosen.

• You see a type-2 LSA in a router’s database.  Without knowing what the details of the LSA are, list some things you can assume have happened.

Type-2 LSAs mean a transit network (multiaccess network) is turned up somewhere.
This transit network has two or more OSPF routers on it.
An election for DR and BDR has taken place.
The DR has started acting as a pseudonode for the transit network.
All other area routers have been told about that transit network.
[There are many others, I'm sure.]

• What configuration is required to enable unequal-cost load balancing in OSPF?

This isn’t EIGRP; you can’t do that.

• It seems that your OSPF database has 95 equal-cost paths to the same network.  By default, how many show up in the routing table?

Four.  You can change this with the maximum-paths directive under OSPF.

• Your 700 series router has been elected the DR on a transit network.  How do you make sure your 12000 series is elected instead?

On the 12000′s interface, set the priority higher with the ip ospf priority x command.

What Command Was That

What command…

…shows all the type-1 (router) LSAs that a router has seen?

show ip ospf database router

…shows all the type-2 (network) LSAs that a router has seen?

show ip ospf database network

…shows all the type-3 (network summary) LSAs that a router has seen?

show ip ospf database summary

…shows the maximum paths OSPF will send to the routing table?

show ip protocols

…shows what transit networks exist in the area?

show ip ospf database network

…shows all the routers in the area?

show ip ospf database router

…shows what router advertise a particular transit network?

show ip ospf database

…shows the DR and BDR for a transit network?

show ip ospf interface

…shows the reference bandwidth?

show ip protocols

…shows how many times the SPF algorithm has been run in an area?

show ip ospf
- OR -
show ip ospf statistics

…shows how many of each message type a router has sent?

show ip ospf traffic

Study Questions

• What settings must match between two routers in order to become EIGRP neighbors?

Both routers must be in the same primary subnet
Both routers must be configured to use the same k-values
Both routers must in the same AS
Both routers must have the same authentication configuration (within reason)
The interfaces facing each other must not be passive

• What are the default hello and hold times in EIGRP?

On links with bandwidth > 1.544Mbps:
Hello:  5 sec
Hold:  15 sec

On links with bandwidth <= 1.544
Hello:  60 sec
Hold:  180 sec

• How do you change the hello and hold times?

You set these values at the interface.

R1(config-if)#ip hello-interval eigrp 1 X
R1(config-if)#ip hold-tim eigrp 1 X

• How do you keep an interface from being used for EIGRP discovery?

Don’t configure a network statement that includes that interface
Make the interface passive
Configure static neighbors for that interface

• Why might NTP be a good thing to use in regards to EIGRP?

EIGRP uses key chains for authentication.  Key chains can be configured with a range of valid dates and times.  If the time on two routers was off by even a few seconds, some keys would expire, causing neighbor relationships to drop.

• How do you configure EIGRP authentication?

In each interface participating in EIGRP, you configure the authentication mode and the key chain to use.

R1(config-if)#ip authentication mode eigrp 1 md5
R1(config-if)#ip authentication key-chain eigrp 1 KEYCHAIN1

• What are the k-values that EIGRP uses?

k1 = bandwidth
k2 = load
k3 = delay
k4 = reliability
k5 = MTU

• How does a router choose its router ID in EIGRP?

First, it looks for a configured router-id in the EIGRP configuration.  If none exists, it uses the highest (largest) IP address configured on a loopback interface.  If no loopbacks exist, it uses the highest IP configured on the other interfaces.

What Command Was That?

What command tells you…

• …whether a neighbor was discovered or statically configured?

show ip eigrp neighbor detail

• …what interfaces are involved in EIGRP?

show ip eigrp interfaces

• …what k-values your router is using?

show ip protocols

• …how long your router has been neighbored with another router?

show ip eigrp neighbors

• …what your router ID is?

show ip eigrp topology
show ip eigrp accounting

• …a summary of the configured network statements?

show ip protocols

• …the configured hello interval?

show ip eigrp interface detail

• …the configured hold time?

There’s isn’t a way to do it directly.  You have to check your neighbors several times over the course of a few seconds to see where the hold timers drop to before resetting.

To check it out a bit, I set up a small network with four routers connected only to a single Ethernet segment.  I set up one router to run EIGRP, OSPF, and BGP to each one of the other routers just so I could see the output for the different routing protocols.  Here’s what puked out after struggling with GNS for a few minutes.

R1#sh ip protocols
Routing Protocol is "eigrp 1"
  Outgoing update filter list for all interfaces is not set
  Incoming update filter list for all interfaces is not set
  Default networks flagged in outgoing updates
  Default networks accepted from incoming updates
  EIGRP metric weight K1=1, K2=0, K3=1, K4=0, K5=0
  EIGRP maximum hopcount 100
  EIGRP maximum metric variance 1
  Redistributing: eigrp 1
  EIGRP NSF-aware route hold timer is 240s
  Automatic network summarization is in effect
  Maximum path: 4
  Routing for Networks:
    192.168.0.0
  Routing Information Sources:
    Gateway         Distance      Last Update
  Distance: internal 90 external 170

Routing Protocol is "ospf 1"
  Outgoing update filter list for all interfaces is not set
  Incoming update filter list for all interfaces is not set
  Router ID 192.168.0.101
  Number of areas in this router is 1. 1 normal 0 stub 0 nssa
  Maximum path: 4
  Routing for Networks:
    192.168.0.0 0.0.0.255 area 0
 Reference bandwidth unit is 100 mbps
  Routing Information Sources:
    Gateway         Distance      Last Update
  Distance: (default is 110)

Routing Protocol is "bgp 65001"
  Outgoing update filter list for all interfaces is not set
  Incoming update filter list for all interfaces is not set
  IGP synchronization is disabled
  Automatic route summarization is disabled
  Neighbor(s):
    Address          FiltIn FiltOut DistIn DistOut Weight RouteMap
    192.168.0.104
  Maximum path: 1
  Routing Information Sources:
    Gateway         Distance      Last Update
  Distance: external 20 internal 200 local 200

The EIGRP section shows some important details, including what k-values are used, networks configured, and administrative distance (AD) of the various route types (internal and external).  The OSPF section shows the router ID, number of areas on the router, and number of area types (normal, stub, NSSA), as well as the networks configured and the AD.  The section regarding BGP shows summarization status, neighbors (along with any filter lists, distribution lists, local weights, and route-maps if they were configured), and the ADs again.

That’s good stuff to know.  I’ll have to put that command in usual repertoire.

I had a VTP server and client in the same VTP domain, and, when I cabled up the trunk, the client overwrote the VLAN database on the server.

The moral of the story is that the best revision number will win no matter what the operating mode of the switch.

• 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

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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