Bpms multiplier

Calculating bandwidth

Adding VAD to the mix

• Sometimes messes up quality....

MTPs

• SW MTP - install Cisco IP Voice Media Straming App on CM or IOS GW w/out using DSPs - cannot transcode (need HW MTP)
• HW MTP - NM modules on 2800 and 3800, WS-SVC-CMM-ACT, Cataluyst WS-X6608-T1, WS-X6608-E1

DSPs

• DSP does hw based calculations, transcoding, voice term, media termination point (MTP) (e.g. hw - translating between mu-law (in U.S.) and a-law, sw - supports (H.323) supplementary svcs (if the endpoint doesn't), conferencing.
• Cisco puts DSP use in rtrs...

Codec Complexity

Cisco Commands

Verifying Codec  Complexity

Calculator for Cisco DSP resources

DSP Configuration commands

SCCP config

show statements

show dspfarm profile 1 (profile #)
show dspfarm dsp all

CM config

Analog Signaling

• Supervisory signaling
• loop-start - close loop to go off hook - problems with glare and switch originated disconnect supervision
• ground-start - preferable
• E&M - RJ48 connector - 2 or 4 wire can have 6 to 8 physical wires
Cisco support I, II, III, and V.

Type	Wires	Comments
I	1 E, 2nd M, remaining 2 pairs audio PBX side - indicate off hook by connecting M to battery line side - indicate off hoook by connecting E to ground	most common in North America
II	1 E, 2nd M, 3rd signal ground, 4th signal battery PBX side - indicate off hook by connecting M to SB (signal battery) Line side - indicate off hook by connecting E to SG (signal ground)	used in sensitive environments - produces little interference
III	4 wires for signaling idle - E open, M connect to SG PBX off hook - move M from SG to SB line side off hook - ground E	not commonly used
IV	uses 4 wires for signaling idle - E and M open PBX off-hook, move M from SG to SB line side off-hook, move E to SG (grounded on PBX side)
V	similar to Type I. 2 wires (E & M) idle - both E&M are open. PBX off-hook - ground M line side - off hook - ground E	most common outside of North America
SSDC5	Similar to type V, but backwards - if line breaks, interface defaults to off-hook (busy)	often found in England

• address signaling
• pulse
• DTMF
• Informational signaling
• call progress tones - dial tone, busy tone, ring-back, congestion (LD), reorder (local - similar to congestion), receiver off-hook, no such number

• wait 150ms before sending DTMF digits after off-hook

• responding side sends wink goes off-hook 140-200ms.
• eliminates glare
• default and most commonly used

CAMA - Centralized Automated Accounting

Configure DID Trunks

Timers and Timing Configuration

• timeouts initial - dial-tone length
• timeouts interdigit - highend of time (maximum) (CM default 15sec, router 10sec)
• timeouts ringing
• timing digit - time that someone presses digit
• timing interdigit - lowend of time (minimum)
• timing hookflash-in and hookflash-out

Show commands

test commands

debug commands

OSI Ref Model

Comm Decisions

• how to convey or pass msg
• select language used for communication
• determine how best to prevent msg from mixing with other msgs
• determine how to deliver msg

Prereq knowledge

• Loc of other device
• medium needed to get msg to device
• proper use of medium

7 layer model

7. App layer - provides svcs to user
6. Presentation layer - translate to diff display formats
5. Session layer - keeps app organized
4. Transport layer - make sure networked conversations are maintained valid
3. Network layer - determine path msg travels
2. Data Link layer - organizes local/physical transmission
1. Physical layer - organizes low-level signals over medium

IP Suite

• App Layer -> app, preso, and session layer (e.g. Telnet, FTP, HTTP, SMTP)
• Trans Layer -> transport layer (e.g. TCP, UDP)
• Internet Layer -> network layer (e.g. ipv4, ipv6)
• Network Interface Layer -> data link and phys layer (e.g. ethernet, ppp, mpls, atm, etc.)

Transport Models

Ethernet

Carrier Class PBT/PBB

Voice over Frame Relay

• uses PVCs and SVCs for transmission
• CIR (Committed Info Rate) guarantees transmission. Above this DE (discard eligible) (bit) marked traffic can be dropped.
• 56 Kbps-45 Kbps

Voice over ATM

• 53-byte cells
• fixed channel or route between 2 points
• CBR (constant bit rate) - like a leased line
• VBR (variable bit rate) - data not sent evently. Popular for voice and video conferencing.
• UBR (Unspecified Bit Rate) - no guaranteed trhoughput levels. File transfer might be an application
• ABR (Available Bit Rate) - guaranteed minimum cap, allows birsting when higher caps are avail
• 25-622 Mbps
• small ATM cell size is much less subject to serialization delay than Frame Relay
• 
  

Voice over PPP

• full duplex over dedicated or circuit switched

Voice over Wireless LAN

• DCF (Distributed Control Function) - if medium is idle (Carrier Sense), WLAN devices wait for inter-fram spacing interval gbefore they transmit; this builds in delay for VoIP
• DCF can use either DIFS (Distributed Control Function Inter-frame Spacing, or SIFS (Short Interval Inter-frame Spacing). SIFS is shorter. PIFS (Point Control Function Inter-Frame Spacing) is rarely used.

Voice over MPLS

• To use for VoIP, make sure network is configured with Fast Route (also called MPLS local restoration or MPLS local protection). This allows recovery at local level, not at IP level, which could impact voice qual resulting in dropped packets and/or clipping.

Cable Modem

DSL

VoIP Control Protocols

• RTP - L5 protocol to carry voice/codec data.
  
• RFC3550
• Cisco ports typically UDP 16384-32767.
  
• Framework not intended as separate layer(?).
• Works well with RTCP
• Time stamp is autonomous clock source (based on NTP) that determines how many clock ticks have occurred
• time stamp supports VAD and jitter
• RTCP - augments RTP by providing end-to-end delivery service for real-time traffic.
• RFC3550
  
• RTCP XR (Real-time Transport Control Protocol Extended Reports - RFC3611 - allows user-agent endpoints to send metrics to eachother on VoIP call quality (e.g. jitter, packet loss, discards, delay, analog, voice quality,...)
• RTCP can be one-to-one or one-to-many
• RTCP packets
• Seder Report
• Receiver Report
• Source Description
• Goodbye
• app-specific
  

QoS

QoS Mechanisms for VoIP

• Header Compression
• Frame Relay Traffic Shaping (FRTS)
• FRF.12 - Link Fragmentation and Interleaving (LFI) - force break up of big packets on one side
• PSTN Fallback
• IP RTP Priority (Diffserv) - map L3 to L2 technologies
• and Frame relay IP RTP Priority
• IP to ATM class of service (CoS)
• Low Latency Queuing (LLQ) combo of 
• Priority Queuing, and 
• Weighted Queuing, and 
• Class based queuing
• Multilink PPP (MLP)
• RSVP

Objectives

• Supported dedicated BW
• Improve loss chars
• Avoid and manage net congestion
• shape net traffic
• set traffic prios

L2

802.1Q VLAN ID

• Cheat Sheet

Application/Network Service Classes

• IETF RFC4594 defines 12 ASCs (Application Service Classes) via DSCPs.
• These can be mapped to 8 NSCs (Network Service Classes) represented via 802.1q field

ASC to NSC mappings

• Some switches will map DSCP (ASC) mappings to 802.1q (NSC) mappings and ignore what is in that field
• Book says "If an IP packet arrives with a non-standard or experimental DSCP value that is not mapped to any svc class, then this packet must be treated as a DF-marked (default) packet."
• RFC4594 defines scheduler/queue types as rate or priority
• Priority Queuing looks at highest prio queue and sends packets if packets exist. Then goes to next, etc.
• Rate-based queuing including things like WRR and WFQ (see QoS cheatsheet)

NSC Traffic Svc Classes

• App traffic can be divided into 2 categories
• Network Control - net traffic such as routing updates
• User traffic
• Interactive (human to human) - sensitive to delay, loss, jitter. Delay perf on order of 10s of ms
• Responsive (human to server) - less affected by jitter. Can tolerate longer delays than interactive traffic. Delay perf on order of 100s of ms
• Timely (between servers or server to human) - daly tolerance significantly longer than Responsive traffic. delay perf less than 1 sec.
• NSCs are a superset of six QoS classes defined in ITU-T Y.1541.

Port-based prioritization

VLANs

Types

• Port based
• Policy based
• MAC based
• IP subnet based
• Protocol based
• Book forgot application based (e.g. application selects VLAN)

Queue types

Queuing guidelines

• Also, realize the on a low-bandwidth connection, WRR and WFQ may not prevent serialization delay. Implement fragmentation on these links.
• on high bw connections, strict queueing is recommended for VoIP queue.
• WRR and WFQ not recommended for VoIP over WAN

WAN Queuing considerations

• For PPP, consider PQ, RTP header compression, and fragmentation.
• For FR, with voice and data sharing a PVC, consider PQ with traffic shaping to ensure voice packets are note discarded or delayed.
• For ATM, with voice and data sharing a PVC, consider PQ to guarantee voice has prio over data

L3

RSVP

Traffic Shaping

IP Addr Prio

DiffServ and DSCP

• See the DiffServ section in the QoS Notes.
• PHB - Per hop behaviour recommended by IETF for DE, AF, and EF marked traffic.
• DiffServ Domains and Edge Nodes
  • Edge Node on edge of domain. Applies QoS policies for edge of DiffServ domain, and connects domain to non-DiffServ network.
  • Boundary Node between domains. Provides QoS policies between 2 or more DiffServ domains.
  • Interior Node is in trusted part of DiffServ domain. Less compleex traffic policing and conditioning.

L4 and beyond

• Best-effort networks - try creating 3 prio levels, Prem for voice traf, Platinum for VoIP signalling, BE for rest
• L4 TCP/IP classification - look at TCP or UDP port #s. May not always work if duplicate ports are in use.
• 
  
• prio - reserve a RTP port range and prio on this
• Packet Frag - can add processing delay to frag, interleav, and re-int. Be careful when apps set Do Not Frag bit. Reduced MTU puts data on WAN sooner, but makes transmissions less efficient.
  • Frame Relay (FRF.12) - Frag smaller pieces and interleave real-time frames with non-real-time frames
  • ATM - packets frag automatically into 53-byte cells
  • PPP - split large packets into smaller ones and encapsulate into PPP frames before queing and trans. Hi prio packets can interupt and trans ahead of lower prio packets.
  • IP - only 1st packet in series of frag packets contains L4 and higher proto info.
• Policy Mgmt - control traf flow based on traf condition, VLAN ID, user prio val, DSCP val, proto type. Schedule the time when policy is effective. What happens to packets in various stes.

• Policy mgrs can ID traffic flows, and mark flows for prio based on packet info.
• Policy mgr administers rules or policies for net behavior to alilgn prios with busines.
• can be applied across multiple devices simultaneously

Common Open Policy Services - Provisioning

Net Mgmt

• Perf Mgmt
• Fault Mgmt

Cisco QoS Implementation Technologies

• Header compression
• Frame Relay Traffic Shaing (FRTS) - Delays excess traffic using buffer or queuing mechanism when data rate of source is higher thhan expected
• FRM.12 (and Higher) -interleaves delay-sensitive voice traffic on one virtual cirucit with fragments of a long fram from another VC utilizing same interface
• PSTN Fallback - based on network congestion
• IP RTP Prio and FR IP RTP Prio - queuing for slower links, works with Weighted Fair Queuing (WFQ) and Class-Based WFQ (CBWFQ).
• IP to ATM CoS - maps QoS characteristics between IP and ATM
• Low Latency Queuing (LLQ) - provides strict prio queuing on ATM VCs and serial ints.  provides priority queuing in conjunction with CBWFQ.  Cisco recommends.
  
• MLP - Allows large packet to be multilink encap'ed and fag'ed to satisfy delay reqs
• RSVP
  

WLANs

CSMA/CA

• Uses CSMA/CA (Carrier Sense Multiple Access/Collision Avoidance). Wired ethernet uses CSMA/CD (.../Collision Detection)
• Physical Carrier sense -
  • is a station transmitting. If so synchronize...
  • CCA (Clear Channel Assessment) - is there too much ambient radio frequency to indicate that another station is transmitting (or something else is happening
• Virtual carrier sense -
  • In every transmission there is a Duration/ID field with value between 0 and 32,767. Values above 0 are copied to the stations NAV (Network Allocation Vector) countdown timer
  • NAV continues to countdown whether busy or idle.
  • station will transmit when:
    • NAV is 0
    • CCA indicates medium is free

Interframe Spacing

Contention Window

• SIFS is ready to go, OR
• DIFS is ready to go, NAV is 0, CCA is clear

RTS/CTS

• Receiving station (typically access point) transmits CTS to station that sent RTS.
• In mixed mode (802.11b/g), stations send RTS and CTS at slower speed. All stations can wait for NAV value, but 802.11g stations can transmit data packets at faster speeds.
• ACKs must be sent after every Data frame. Otherwise sending station assumes that packet was lost.

Radio Waves

• Low freq waves (100-150khz) propagate as ground waves
• Mid-range frequency waves propagate as space waves and bounce off the troposphere or the ionosphere. They can travel several thousand kilometers.
• Waves with frequencies above 3GHz propagate as direct waves, and can travel only as far as the optical horizon.
• WLAN signals genarlly fade in strength with the square of the distance they travel. Obstacles in the way make it worse (e.g. by the fourth power of the distance they travel, could be totally blocked).
• remember to do site survey, not just guesstimate what the coverage is going to be.

• Lobe - coverage area of RF antenna.
• Multi-path - signals reflect/bounce off of other objects, and cause noice or out-of phase interferance

WLAN components

• Mobile Units (MUs) or stations
• Access Points
• Access Ports - switches to coordinate access points, make them intelligent, allow for easier mgmt.
• WLAN switches
• GWs and VPN term points
• Supporting servers (DHCP, DNS, etc...)

Service Sets

• BSS (Basic Service Set) - 1 AP and 1 or more MUs. All traffic must go thru AP
• ESS (Extended Service Set) - Multiple APs and 1 or more MUs. All APs have same SSID (same service set identifier). Roaming from one AP to another is allowed.
• IBSS/ad hoc (Independent Basic Service Set) - no AP. Units talk directly to each other

Activities

• Beacons
  • Beacon frames sent out at frequent and periodic intervals.
  • Intended to synchronize the MUs via time stamp.
  • Contains channel info and SSID. Some APs will allow SSID advertising to be turned off.
  • Contains TIM (traffic indication map) that lets clients know if AP has data to send to it.
  • Contains info about which data rates are supported by AP
• Scanning - passive - listen for beacons and attempt to associate when you see a matching SSID
• Scanning - active
  • Send probe request frame with SSID of net that MU wishes to join (or wildcard). All APs that match, respond.
  • frame is simlar to beacon frame, but no TIM.
  • if multiple responses received, MU joins strongest signal strength
  • continue to gather info about nearby APs, so roaming is easier
• Authentication and Association - after MU determins AP to connect to:
  • MU sends Auth frame to AP. AP can auth or hand off req to back0end server
  • Auth establishes verfied L2 connection. AP sends auth response to MU after ID verification
  • After authentication, MU associates with AP (sends a association req frame and gets a association resopnse frame back from AP allowing or disallowing association request. Once associated, data can be sent
  • pre-authentication makes roaming smoother

Antennas

• omni directional - standard or high-gain
• highly directional
• semi directional (i.e. 1 half of a omni directional)

WLAN Power

• Decibels per mW (dBm) measure abs power. 1mW of power is amped to 10 mW would hav a gain of +10dBm.
• Gain of antena is dBi (decibels / isotrobic radiator or perfect antenna). AP at 20 mW, gain is 10dBI, trans signal is 200mW.

FHSS

• Frequency Hopping Spread Sprectrum
• 2.4GHz ISM (Industrial, Scientific, and Medical) band
• channel 2 through 79 - 1MHz wide sections of band
• 802.11 hops ~10x per second.
• hop sequence specified freq hopping sequence
• Dwell time defines how long to stay on specific frequency
• Hop time time takes to hop from on freq to another. Usually measured in ms.

DSSS

• Apply quick large chipping sequence to info bit, to result in large number of bits sent across spread spectrum simultaneously
• DSSS uses Barker Codes of (+1 +1 +1 -1 -1 -1 +1 -1 -1 +1 -1) and (+1 +1 +1 +1 +1 -1 -1 +1 +1 -1 +1 -1 +1).

Standards

Delays and ARQs

WLANs use positive acks; ARQs (Auto Repeat Reqs) are common. These happen when drrors are detected and can be complete retrans or retrans of orig info using idff mod and coding scheme. This happens outside of typical QoS queues and prios.

You also have to wait for interframse spacing, NAV countdown, waiting for IDFS, etc. Every frame has to be ack'ed.

Voice packet typically experiences (at least):

• 50 microseconds - DIFS
• 80 microseconds - Avg Backoff
• 192 microseconds - Synch/Preamble
• 171 microseconds - full packet transmit
• 10 microseconds - SIFS
• 192 microseconds - preamble
• 11 microseconds - MAC ACK

VoIP Security

Firewalls

Types of External Threats

• DoS attacks
• Intrusion attacks
• Information theft

Tech Solutions

• intrusion - fw combined w/ auth svcs
• data theft - fw and encrypt sw
• data mod - encrypt and gi sig sw

Types of FWs

• Simple filtering routers
• Stateful packet filters
• Circuit-level GWs - checks for TCP and UDP connections, then nothing.
• App GWs (ALGs) - book describes these as proxy, thinks these aren't transparent to users.
• Bastion Hosts

Topology elements

• Dual-homed Host
• Screened Host - Bastion Hosts off to side of fw.  All traffic passes through them.
• DMZ and Perim Nets - Bastoin Hosts/ALGs behind filtering router.  All traffic passes through them.
• Combined Rtr and FW
• Isolated Perm Network - Only one rtr/fw combo with DMZ arm and protected nets arm.

NAT

NAT Traversal solutions

• UPnP w/ NAT device as to what exteral address/port will be.  Can only traverse one NAT.
• STUN (Simple Traversal of UDP through NAT) - Send multiple STUN query requests to STUN server outside of firewall.  Determines external address(s) and what type of NAT it is behind.  Doesn't work with Symmetric NATs.
• TURN (Traversal Using Relay NAT) - TURN server on Internet becomes a relay server changing NAT addresses to it's own and forwarding.  Also handles media.  Expensive solution.  Can be bottleneck and single point of failure.
• NAT SIP ALG - ALG inspects every SIP paragrpah, performs necessary translations, and updates SIP msg before forwarding packet.
• NAT Hairpinning - 2 private side addrs use public addr in SIP and RTP/RTCP.  Box translates and/or forwards as appropriate.

Crossing Autonomous System Boundaries

B2BUA

SBC

Securing SIP Signalling

• HTTP Digest authentication, MD5-Session or MD5 (MD5 default)
• RFC 3261 Section 22

TLS

• Transport Layer Security (RFC 4346)
• Allows secure channel between 2 machines, on a hop-to-top basis.
• Does not relay on user input.  Computers on each end of hop rely on scerts issued by CA (Cert Auth) to auth each other.  Cannot use self-created certs to create secure chans across pub net.

TLS Auth

Secure SIP

• Each TLS chan provides single hop of encryption and auth.  
• RFC3261 requires each hop to be secured or send 416 error msg.  Exceptsions made for first and last hop (can be secured in other ways (such as IPSec).

SRTP

• Secure Real-time Transport Protocol (RFC3711) - bump in the stack (encrypt/decrypt the
  
•  at both endpoints).
• Session keys derived from Master key.  Master key is random bit string provided to SRTP by external key mechanism.  RFC3711 references MIKEY, KEYMGT, SDMS, and KINK as external key mgmt protos.
• SRTP can enctyp payload and include auth hash, or just include auth hash.  auth hash is 180 bit truncated to 80 bit HMAC-SHA-1.
• SDP indicates RTP/SAVP or SRTP/SAVPF in media (m=) line.  
• RFC refers to SDP attr referred to as "crypto".  Only use the crypto attr to exchange key with another enctyping tech such as IPSec, TLS, or SIP S/MIME.

VPNs and VPN Components

• Authentication svrs - such as Cert Authorities and RADIUS svrs, gurantee ID
• Manage svrs - contorl monitoring alerting reporting
• Phys transport 
• VPN GWs 
• VPN client sw 

Security Reqs

• Integrity - make sure end to end is valid.  Dig Sigs and 1-way hash or digests
• Privacy - Ectypt to ensure priv
• Encryption
• Authorization - Access rights/privelages LDAP, RADIUS, ID the user,etc...
• Authentication - Username/pw
• Non-repudiation - keep sender from saying they did not send msg
• Non-replay - keep unauth users from recreating session by impersonating respondants

Encryption and Key mgmt

• Symmetric (1 key both sides) can be less secure/scalable
• DES
• 3DES
• RC4
• AES
• Asymetric (1 key for encrypt, 1 key for decrypt) can be slower/more cumbersome.
• RSA
• ECC (Elliptic Curve Crypto, good for handhelds)

VPN Tunneling Protos

• GRE (not encrypt?)
• L2F
• PPP (RFC1661)
• PPTP (RFC2637)
• L2TP (RFC2661)

IPSec AH/ES

Authentication for IPSec Tunnels

• HMAC 128 or 160bit - msg auth mechs using crypto hash functions, such as MD5, SHA-1
• MD5 128bit
• SHA-1 160bit

VPN Session Authentication Protos

• PAP
• ChAP
• MS-CHAP
• MS-CHAPv2

SSL/TLS VPNs

• Secure web browsing, email ,other data transfers
• Secure gw can term conns and provide policy enfocement and access ctrl.
• Encrypts HTTP data payload
• Complete start and stop proxy.  App specific algos
• Easy to use RAS
• TLS handshake establishes symmetric keys that server and client use for session
• Can provide app level access for VoIP across NAT, FWs, and provides drop-out resiliency.

User Authentication Services

• RADIUS - External
• LDAP - Internal or External db or external proxy (preferred)
• Certs
• SecurID using a RADIUS server proxy

E911

• 911/122 call goes to PSTN and then to Selective Router
• Selective Router matches ANI/CLID agaisnt MSAG (Master Street Address Guide).  Routes to appropate PSAP.
• PSAP dips into ALI DB and pops info to call taker's computer terminal

L2 Switch port discovery

• L2 DM (Discovery Manager) uses SNMP traps and queries to learn where phones connect ot network.

L3 Subnet Discovery

• Based on VoIP phones IP address, assign acceptable ERL (Emergency response Location)/zone.
• ERLs/zones should be 7000 square feet or less
• ERL associated with specific ELIN to be usesd as CLID/ANI for 911 calls.

E911 Considerations

• Support 911 as well as AC+911, (e.g. 9911).  Allow for misdials.
• Allow for onsite notification of 911 events, but do not intercept call from getting to PSAP.
• Set up dedicated 911 trunk facilities.

Network Assessment

Pre-Sales Planning phase

1. Who's the network assessment prime.
2. ID the PCs used as endpoints for assessment (kit or customer).  Make sure they meet baseline assumptions.
3. Obtain physical and logical network diagrams.
4. Obtain list of equipment and telephony currently provided.
5. Obtain additional net info
• Voice quality targets
• loc of users
• link types
• QoS strategy
• etc...
6. Verify net assessment kits have appropriate tools

Net Assessment Work FLow

1. Readiness audit.  Security.  Power and Wiring.  Protocol considerations.
2. Estimate traffic.  Use or generate existing numbers of possible.
3. Develop Solution
4. Work with customer to Implement.

Cisco Stuff

Reasons for using VoIP

1. Cost savings
2. Flexibility
3. Advanced features
• Advancced call routing
• Unified messaging
• Integrated Info Systems
• Long-distance toll bypass
• Security
• Customer relationships
• Telephony application services - pixel-based display stuff..

VoIP Network Components

• IP Phones
• Gatekeeper
• Gatway
• MCU (Multipoint Control Unit)
• Call agent
• App servers - voice mail unified messaging, Attendant COnsole
• Vidconf station

VoIP Functions

• Signaling
• Database Services - access to toll-free nums, caller ID, etc requires cap to query db.  billing info, caller name delivery, toll-free db svcs, calling card svcs
• Bearer control
• Codecs

H.323 Protocols

Gateways

• Analog station gateways.
• Analog trunk gateways.
• Digital gateways.
• H.323 and SIP gateways don't need a call control agent (like UCM).
• 2600XM or 3700 rtrs support H.323 and MGCP 0.1 beginning with IOS 12.2(11)T and UCM Rel 3.1+.
• SMDI standard for integrating VM systems with PBXs or Centrex.  Uses FXS or PRI conections, use SCCP or MGCP to work with specific line/ports.
• Adv GW functionality
• DTMF relay
• supp svcs - hold transfer, conference, etc.
• Rehoming to secondary UCM in event of failure
• call survivability
• QSIG support
• fax/modem support
• 2801, 2811, 2821, 2851
• 38xx
• 6500 w/ Communication Media Module (CMM) - can handle up to 144 T1/E1 connections using 8 CMMs with 18 ports each
• 1751-V, 1760-V (Modular access rtrs
• 2600XM
• 3600
• 37xx

Standalone Voice Gateways

• VG224 (and used to be VG248)
• AS5300 (T1/E1 gw)
  
• AS5400 (enhances fax/modem)
• AS5850 Universal gateways (T3/T1/E1)
• 826-4V ADSL Rtr
• ATA 186 - supports up to 2 voice portHA, fault tolerant infra.
• 7200

Deployment Models

Single-Site Deployment

• max 30,000 SCCP/SIP phones or SCCP vid endpoints, max 1100 H.323 or MGCP devices (gws, MCUs, trunks, clients)

Best Practices

• know calling patterns
• Use G.711 if possible (reduce DSP use)
• Use SIP, SRST, and MGCP GWs for PSTN  H323 possibly or SS7 or Non fac assoc signal (NFAS)

Multisite WAN w/ Centralized Call-Proc Deployment

• Same Max endpoints
• hi bw in site, lower between sites
  
• min 768kbps WAN recommended (for video)
• SRST set up 4.0+ for video
• Unified CME an alternative for SRST (more features)
• H.323 devs require gatekeeper such as Cisco IOS Gatekeeper (IOS 12.3(8)T+)
• Need to allow for QoS for signaling and RTP over WAN
• Use CAC (Call Admission Control) to limit calls over WAN so quality is acceptible (via QOS)
• locations based CAC - calls from hq to site1 use this much BW (statically set)
• Use regions (HQ, site1, site2) and relationships
• HQ<->HQ G711
• HQ<->S1 G729
• HQ<->S2 G729
• AAR (Automated Alternate Routing) use if WAN bw is exceeded
• Secure Survivable Remote Site Telephony (SRST)
• basic to full-capability
• phone keep-alives to primary, secondary, and tertiary call manager fail
• SRST router monitors keep-alives going between phone and CM
• go to SRST mode
• can use MGCP fallback to config basic dial peers configured
• recommend basic dial plan for MGCP fallback
• build expanded dialplan #s when in SRST mode
• Call Mgr Express in SRST mode 
• 2800 < 100 phones SRST sales model
• 3800 > 100 phones SRST sales model

Best Practices/Design Guidelines

• minimize WAN delay (e.g. clipping, reduce voice cut-through delays)
• user HSRP for backup GWs
• use locations mechanism in CUCM to pfrovide CAC into and out of remote branches
• SRST on IOS GW supports 720 phone whereas Unified CME running in SRST supports 240.
• num of IP phones and line appearsances supported SRST mode at each remote site depends on branch router platform
• SCCP phones - use SRST on IOS GW or Unified CME
• SIP phones use SIP SRST
• MGCP phones, use MGCP GW Fallback.
• Can all reside on same GW.
  

Multisite WAN w/ Distributed Call-Proc Deployment

• same max endpoints / cluster
• without Gatekeeper - Intercluster trunks need to be built between CMs
• with Gatekeeper - Intercluster trunks only need to be built to Gatekeeper from each CM
• Use IOS Gatekeeper for (CAC) Call Admission Control between UCM clusters, 
• use gatekeeper pairs, gatekeeper clustering, alt gatekeeper suport resiliency HA/HSRP between Gatekeeper pairs/cluster members
• Use 11 type of codec on WAN  - H323 dosn't allow for header overhead in bw request.  Simplifies capacity planning.
• use 1 type of codec (simplify conig)
• Provide redundacy for SIP proxies
• ensure SIP proxie have capacity for call rate
• Transparent use of PSTN if IP WAN is unavailable

Clustering over IP WAN Deployment

• Local Failover Deployment
• Remote Failover Deployment (may need higher BW for intracluster traff flows)
• < 40ms round trip delay or DB breaks, can recover - manual effort
• QoS to min jitter for IP Precedence 3 ICCS traffic (CoS 3)
• provide sufficient prio bandwidth for all ICCS traffic, especially prio ICCS traffic
• minimize jitter related delay, packet loss/errors especially for Intra-Cluster Communicatoin Signaling (ICCS) traffic
• Prvide sufficient bw for expected voice/video traffic
• configure QoS appropriately
  

VoIP Design Elements

Call Manager Publisher/Subscriber

• Changes are made on publisher
• Subscriber picks up changes

Networking/Audio Clarity

• Fidelity - accurate recreation of signal. human speech bw 100-10,000Hz, although 90 percent of speech is between 100-3000hz
• Echo - usually due to impedance mismatch
• Jitter - variation in arrival of voice packets - runthrough dejitter/playout buffer
• Delay - G.114 says<150ms acceptable...for private networks 200ms is reaonable, 250ms is limit...remember to calculate coder, packetization, queuing buffering, serialization, dejitter buffer, as well as network delay...
• Packet Loss - loss of packets on the network, shoot for < 1%
• Side tone - hear your own voice in the earpiece
• Background noise - Compensate for voice activity detection (VAD) issues with comfort noise generation (CNG).
  

Modulated Data over IP Nets

Fax

• Fax Relay - T.30 fax from PSTN demod'ed, enveloped into TCP packets, sent over net, remodulateed to T.30 on far end.
• Cisco default is Cisco Fax Relay (proprietary)
• T.38 standard can also be configured.
• H.323, SIP, MGCP
• Fax relay packet loss concealment
• MGCP-based fax (T.38) and DTMF relay
• SIP T.38 fax relay
• T.38 fax relay for T.37/T.38 fax gateway
• T.38 fax relay for VoIP H.323
• Fax pass-through - passes in-band end-to-end over IP net.  Preferred method of sending
  
• G711 with no VAD and No Echo Cancellation, or
• clear-channel codec, or
  
• G.726/32 (?)
• GW notes fax tones, and changes codec configured to G.711 with not VAD, no EC for the duration of the fax session.  Usually changes packetization to 10ms.
  
• voice-band-data referes to transport of fax modem signals over oice channel thru packet net.
• redundant encodng or packet redundancy often used to mitigate packet loss.
• Does not work when codec is G.Clear (GW cannot detect fax tone).
• Supported in H.323, SIP, MGCP.
  
• Store-and-forward fax - uses separate process.  ITU-T T.37. Converts to TIFF. Sends via SMTP. Can be delivered/received between computers rather than machines.

Modem

• Modem pass-through - similar technique to fax-pass-through
• Can use packet redundancy (e.g. @ 10ms sample size) to mitigate jitter, loss, etc
• In Cisco, static jitter buffer of 200ms kicks in
• Modem relay - modem signals demodulated at gw, converted to digital, carried in Simple Packet Relay Transport (SPRT).
• SPRT runs over UDP
• On detection of modem answer tone, GWs switch to modem pass-thorugh, and then if call menu (CM) signal is detected, into modem relay mode
• Defaults to 20ms packet size

DTMF

• DTMF tones are distorted when gateways use compression on slower WAN links or compression oriented codecs
• DTMF relay addresses problem
• Cisco Proprietary - DTMF digits encoded differently and RTP packets IDed as type 121
• H.245 Alphanumeric - send through H.245 signaling channel as User Input Indication msgs. Guaranteed delivery
• H.245 Length - Similar to H.245 Alphanumeric, but includes info on length digits are pressed
• NTE - (H.323 is RFC2833) - Separate Codec type negotiated that cannot be compressed for DTMF digits. Still uses RTP.
• NSE - (MGCP is RFC2833)
• SIP DTMF can use Cisco Proprietary method using Notify msgs - similar to DTMF relay described in RFC2833

Notes from CVoice Class

• Cost Savings
• Flexibilty 
• Advanced features
• Advanced call routing (e.g. PSTN fallback)
• Unified msging
• Integrated info systems (e.g. integrate AD users into VoIP systems)
• LD toll bypass (e.g. Tel-End-Hopp-Off (TEHO) - toll by-pass to local #)
• Voice security
• Customer relationship
• Telephony application services

• H.323 - Rtr decides what to do, lots of config on rtr (all dial plan), not much on Call Mgr, more distributed
• MGCP - call mgr makes decision, very dependent on call agent such as call mgr, lots of config on Call Mgr (all dial plan), a little on rtr, more centralized
• SIP - Rtr decides what to do, in this class, lots of config on rtr, less on Call Mgr
• SCCP/"Skinny" - call mgr makes decision - default for phone

• Distributed configuration

• IETF RFC 2705 - 1999
• Centralized configure on CM - backhauls used for signaling!
• make sure IOS and CUCM have compatible version
• Allow call agent to control interfaces, call agent (e.g. CM) controls all
• if CM controlling MGCP PRI  backhaul controls interfaces...
• BRI backhaul available with recent versions

• P2P
• SIP GWs never registered with CUCM; only IP addr available to confirm comm is possible
• 2800, 3800
• Distributed configuration

• Proprietary terminal control protocol
• stimulus protocol -  for every event end device sends msg to CUC
• Can be used to control GW FXS port (e.g. VG224)
• Proprietary nature allows quick additions and changes
• SEP (Selsius E Phone)
• Centralized
• CM maintains dialplan
• 224 ata 186, 2800 with FXS

• Latency - delay
• Jitter - Doesn't re-arrange sequence
• BW
• Packet loss
• Reliability
• Security

Call Types

• Local
• On-net
• Off-net
• PLAR - Private Line Automatic Ringdown - auto connects tlephone to 2nd phone
• PBX-to-PBX - originates at one PBX and terminates at another
• Inter-Cluster Trunk calls (H.323):  Occurs when calls are routed by 2 separate CMs Device/Trunk/Add New
• On-net-to-off-net

Dial Peers

• Inbound call leg matches inbound dial peer
• outbound call leg matches outbound dial peer
• POTS/PSTN dial peer - type 1
• VoIP dial peer - type 2...
• match where it came from 1st

• dial peer is an addressable call endpoint
• dial peers establish logical connections, called call legs, to complete an end-toend cal

POTS Dial Peers

VoIP Dial Peers

Inbound Dial Peers

Outbound Dial Peers

Digital Voice Ports

Configure commands

PRI QSIG

BRI

PRI

Busyout channels (e.g. H323)
interface Serial0/1:23
isdn service b_channel 0-23 state 2 soft
  b_channel <range>
  state <0=InService, 1=Maint, 2=OutOfService>
  soft - place in defined state when chan is idle

Troubleshooting

QSIG

H323 GW Config

H.323 Fax Passthrough

DTMF Relay

Verifying H323 GW

H323 GK services

technology prefix

Debug

GK Cfg steps

GK Zone BW Operation

Dial Peer CAC

RAI (Resource Availability Indicator)

Debug Commands

MGCP

Advantages of MGCP GW

MGCP GWs

Calls and connections

Residential GW cfg

Trunk GW cfg

Fax passthrough and relay

MGCP debugging

CM steps

SIP

SIP Advantages

SIP Architecture

SIP Servers

Commands

SIP Addresses

SIP DTMF on CM

SIP GW cfg

Debug commands

Dial Plan on GWs

Digit manipulation commands

• num-exp (num-exp 55.. (prefix number with 55) global command)
• digit-strip (no digit-strip to not strip)
• prefix
• forward-digits (e.g. forward-digits 2 or forward digits all (right justified))
• voice translation-profile
• clid (modify your caller id)

voice translation-rule 1
   rule 1 /^4085551/ /1/
voice translation-profile pstn-in
   translate called 1
voice-port 0/0/0:23
   translation-profile incoming pstn-in
   
voice translation-rule 2
	rule 1 /^2/ /4085552
voice translation-profile pstn-out
	translate calling 2
voice-port 0/0/0:23
	translation-profile outgoing pstn-out

clid

Calling Privileges

Call Coverage - Hunt groups

• can manipulate ANI based on TON
  

Digit manipulation

Debugging dialpeers and dialplans

Digit Manipulation Order

• Best Practice - do digit manipulation as soon as possible
• POTS dial peers by default strips left-justified digits (deestination-pattern 555.... strips 555)
• disable with no digit-strip or forward digits all
• VoIP dial peers - by default router fwds all digit pairs
  

Inbound POTS

Inbound VoIP

Outbound POTS

Outbound VoIP

Digit collection

• rtr collects one at a time until match an outbound dial peer
• after match made, rtr immediately places call, no futhrer digits collected

Caller ID Name/number manipulation

Voice translation rules/profiles

Regular expressions

Voice Translation Profiles

voice translation-rule 1
	rule 1 reject /312555*/
voice translation profile block
	translate calling 1
dial-peer voice 111 pots
	call-block translation-profile incoming block
	call-block disconnect-cause incoming invalid-number

dialplan-pattern

Basic digit manipulation - Quick ref

Call Routing and Path Selection

Best practices

Tail-End Hop-Off (TEHO)

COR (Class of Restriction?)

Enter COR cfg mode
dial-peer cor custom
	Name the CORs
	name 911
	name local
	name ld
	name intl
Configure Outbound Corlists
dial-peer cor list 911call
	member 911
dial-peer cor list localcall
	member local
dial-peer cor list ldcall
	member ld
dial-peer cor list intlcall
	member intl
Configure Inbound Corlists
dial-peer cor list 911
	member 911
dial-peer cor list local
	member 911
	member local
dial-peer cor list ld
	member 911
	member local
	member ld
dial-peer cor list intl
	member 911
	member local
	member ld
	member intl
Assign Corlists to PSTN dial peers
dial-peer voice 911 pots
	destination-pattern 911
	forwrd-digits all
	corlist outgoing 911call
	port 0/0/0:23
dial-peer voice 9911 pots
	destination-pattern 9911
	forward-digits 3
	corlist outgoing 911call
	port 0/0/0:23
dial-peer voice 9 pots
	destination-pattern 9[2-9]......
	corlist outgoing localcall
	port 0/0/0:23
dial-peer voice 91 pots
	destination-pattern 91[2-9]..[2-9]......
	prefix 1
	corlist outgoing ldcall
	port 0/0/0:23
dial-peer voice 9011 pots
	destination-pattern 9011T
	prefix 011
	corlist outgoing intlcall
	port 0/0/0:23
Assign Corlists to incoming dial peers
dial-peer voice 1003 pots
	destination-pattern 1003$
	port 1/0/0
	corlist incoming local
	corlist incoming 911
dial-peer voice 1004 pots
	destination-pattern 1004$
	port 1/0/1
	corlist incoming 911
	corlist incoming locla
	corlist incoming ld
	corlist incoming intl
(optional) assign to SRST cfg
call-manager-fallback
	cor {incoming | outgoing } cor-list-name  [cor-list-number starting-number - ending-number | default]
		(e.g)
	cor incoming intl 1 2000 - 2100

Debugging COR

Border Control (Cisco UBE)

Call Manager Express example

SIP Trunk Cfg example

Debug Commands