Setting up a VoIP GW

Notes for CVoice exam. Got a lot of this from CVoice 8.0 - Implementing Cisco Unified Communications Voice over IP and QoS v8.0 by Andrew Froehlich

Table of Contents

Setting up Analog Voice ports

Practice configs

FXS (basic)

voice-port 0/0/0
    signal loopstart (or groundstart or E&M)
    cptone <2DigitCountryCode> (set's call progress tones)
    ring cadence pattern08 (or 'ring cadence define pulse interval)
    ring frequency 50 (or 25 (Hz))
    station_id number 3333 (caller ID)
    station_id name Joe Smith (caller ID)
    connection plar 4564 (auto ring-down to 4564)
    impedance 600c (ohms) (or 600r (real), or 900c (complex), etc..)
    input gain 2 (db) (scale back volume inbound)
    output attenuation -1 (db) (scale back volume outbound)
    echo-cancel coverage 32
    !shut/no shut if you change signal type

FXO (basic)

voice-port 0/0/0
    signal groundstart (or loopstart or E&M)
    dial-type dtmf (or pulse)
    connection plar opx 4564 (auto ring-down on off hook to 4564 - don't forget dial-peers to handle)
        without the opx you could get stutter ring down
        without the plar you would need to plan on 2-stage dialing or autoattendant plar
    ring number <1-10> (number of rings before answering call (not nec with plar above))
    !shut/no shut if you change signal type

FXS/DID w/ FXO outbound and related dial-peers

voice-port 0/0/0
    description FXS/DID
    signal loopstart
    signal did wink-start (or immediate-start, wink-start, delay-start)
    !shut/no shut if you change signal type

dial-peer voice 12 pots
    description inbound dialpeer
    direct-inward-dial (alternate is plar or 2-stage dialing on FXO)
    incoming called-number ....
    port 0/0/0

dial-peer voice 13 pots
    description outbound to phone for 3445 to port 0/0/1
    destination-pattern 3445
    port 0/0/1
    
voice-port 1/0/0
    description fxo port for outbound dialing
    signal groundstart
    !shut/no shut if you change signal type

dial-peer voice 9 pots
    description outbound to fxo port
    destination-pattern 9[2-8]..........
    forward-digits 10
    port 1/0/0

FXO CAMA with related dial-peers

voice-port 1/0/1
    description CAMA 911 port to PSAP
    signal cama KP-0-NPA-NXX-XXXX-ST (Type 2 CAMA Signaling)
                KP-0-NXX-XXXX-ST     (Type 1 CAMA Signaling)
                KP-2-ST              (Type 3 CAMA Signaling)
                KP-II-NPA-NXX-XXXX-ST-KP-NPA-NXX-XXXX-ST
                                     (Type 5 CAMA Signaling)
                KP-NPD-NXX-XXXX-ST   (Type 4 CAMA Signaling)
                      (may have to shut/no shut bounce port for CAMA signal config)
    ani mapping 0 555 (for NPD, must config digit (here it's 0) for area code (here it's 555))
    !shut/no shut if you change signal type

dial-peer voice 911 pots
    destination-pattern 911
    forward-digits all
    port 1/0/1

dial-peer voice 9911 pots
    destination-pattern 9911
    forward-digits 3
    port 1/0/1

E&M

voice-port 0/0/0
    type 2 (or 1 or 3 or...See below for types)
    operation two-wire (or four-wire)
    signal immediate-start (or wink-start, or delay-dial)
    !shut/no shut if you change signal type

Debugging Analog ports

term mon
debug voice ccapi inout
debug voip vtsp tone !was debug vtsp tone
debug vpm signal

gateway1#sh voice port sum
                                                                          IN        OUT
PORT            CH       SIG-TYPE           ADMIN        OPER             STATUS    STATUS        EC
=============== ==       ============       =====        ====             ========  ========      ==
0/3/0           --       fxo-ls             up           dorm             idle      on-hook       y 
0/3/1           --       fxo-ls             up           dorm             idle      on-hook       y

you can look at port status and see if .ADMIN. state is .up. and .OPER. state is .dorm. and .OUT STATUS.
is on-hook.  This is a normal condition of the port.

If OPER state shows .UP. then that means port is detecting off-hook/seizure or is stuck in disconnect mode.

test voice port x/x/x  si-reg-read 29 1
you will normally get either a 0X00 or 0XNN (where NN is some number other than zero).
value of 0x00 indicates there is no voltage being seen on the line/voice port. 
You need the term mon to see the output of this command.

test voice port 0/3/0 si-reg-read 29 1
gateway1#
Apr 16 13:50:39.285: 
Values read from SiLabs Codec connected to DSP 0, channel 0:
--------------------------------------------------------------
Register 29 = 0x00

simulating a call....................................

csim start <dn>
gateway1# csim start 916175551212
csim: called number = 916175551212, loop count = 1 ping count = 0
                                                                                csim err csimDisconnected recvd DISC cid(80) 
Apr 16 13:51:02.462: htsp_timer_stop3 htsp_setup_req
Apr 16 13:51:02.466: htsp_process_event: [0/3/0, FXOLS_ONHOOK, E_HTSP_SETUP_REQ]fxols_onhook_setup
Apr 16 13:51:02.466: [0/3/0] set signal state = 0xC timestamp = 0
Apr 16 13:51:02.466: htsp_timer - 1300 msec
Apr 16 13:51:02.722: htsp_process_event: [0/3/0, FXOLS_WAIT_DIAL_TONE, E_DSP_SIG_1100]fxols_power_denial_detected
Apr 16 13:51:02.722: htsp_timer2 - 1000 msec
Apr 16 13:51:02.722: htsp_timer_stop 
csim: loop = 1, failed = 1  
csim: call attempted = 1, setup failed = 1, tone failed = 0

gateway1#
Apr 16 13:51:03.722: htsp_process_event: [0/3/0, FXOLS_WAIT_DIAL_TONE, E_HTSP_EVENT_TIMER2]fxols_power_den_disc
Apr 16 13:51:03.722: htsp_timer_stop 
Apr 16 13:51:03.722: htsp_timer_stop2 
Apr 16 13:51:03.722: [0/3/0] set signal state = 0x4 timestamp = 0
Apr 16 13:51:03.722: htsp_process_event: [0/3/0, FXOLS_ONHOOK, E_HTSP_RELEASE_REQ]fxols_onhook_release
gateway1#

sh call hist voice command will show a call attempt on that port.

Analog circuit cheatsheet

analog cheatsheet

Setting up digital ports

Practice configs

T1 (basic)

controller T1 2/1
    framing esf (or sf)
    linecode b8zs (or ami)
    clock source line (or internal)
    ds-group 0 timeslots 1-12 e&m-wink-start (or e&m-immediate-start, or fxs-loop-start, etc...)

T1 channel cross-connect to FXS port

controller T1 2/1
    framing esf (or sf)
    linecode b8zs (or ami)
    clock source line (or internal)
    ds-group 1 timeslots 13 fxo-loop-start

voice-port 2/1:1
    signal loopstart
voice-port 0/0/0
    description fax
    signal loopstart
    
connect fax1 voice-port 0/0/0 t1 2/1:1  (cross connect ds-group 1 to fxs port on 0/0/0)

dial-peer 211
    incoming called number .
    port 2/1:1
dial-peer 3000
    destination-pattern 5554443000
    port 0/0/0

show connection all (verify list of connections, mappings, current state)

PRI

isdn switch-type primary-dms100 (or primary-5ess, or primary-qsig, etc...)
controller T1 2/1
    framing esf (or sf)
    linecode b8zs (or ami)
    clock source line (or internal)
    pri-group timeslots 1-24 (could be fractional (e.g. between 1 to 5-15)

interface serial 2/1:23
    isdn incoming-voice voice (set it to be processed by DSPs)

dial-peer voice 9 pots
    destination-pattern 9[2-8].........
    forward-digits 10
    port 2/1:23

show voice port summary (show status on all voice ports)

PRI - additional sample

network-clock-participate wic  4
network-clock-select 1 T1 0/3/0

card type t1 0 3

controller T1 0/3/0
 cablelength long 0db
 pri-group timeslots 1-24

interface Serial0/3/0:23
    no ip address
    encapsulation hdlc
    isdn switch-type primary-ni
    isdn incoming-voice voice
    isdn supp-service name calling
    ! send an alerting message before the connect message
    isdn send-alerting
    no cdp enable
    no shutdown
!
voice-port 0/3/0:23
    echo-cancel coverage 64
    no shutdown

additional port configs

Practice configs

echo cancellation

voice-port 1/1:0
 echo-cancel enable (usually enabled by default)
 echo-cancel coverage 32 (ms, usually 64 by default)

VAD and comfort noise

voice vad-time 750 (ms - how long to wait before vad kicks in - 250ms in default)

dial-peer voice 100 voip
 no vad (default enabled on dial-peers, but not on POTS interfaces)

voice port 1/0
 vad (default not enabled on POTS interfaces)
 comfort-noise (turn on white noise locally during VAD)

Dial Peers

Practice configs

Inbound dial peer

PUT ONE OF THESE IN FOR VOIP AND/OR POTS TO AVOID DEFAULT

dial-peer voice 1 voip
    incoming called-number .
    dtmf-relay h245-alphanumeric
    codec g711ulaw

Inbound Dial Peer rules

DNIS - incoming called-number

ANI - answer-address

DNIS again - destination-pattern

Inbound port - port

Default Dial Peer 0 - matches if nothing else matches

1. G.729r8
2. no RSVP
3. QoS preference 0
4. Fax-relay disabled
5. no DTMF relay
6. no DID forwarding
7. no IVR support for POTS dial peers

GOOD IDEA TO SET UP AN INBOUND DIAL-PEER with INCOMING CALLED-NUMBER . to avoid default dial peer

Outbound POTS Dial Peers

dial-peer voice 6001 pots
    destination-pattern 6001
    port 0/0/0

Outbound VoIP Dial Peer

dial-peer voice 6002 voip
    destination-pattern 6002
    session target ipv4:<ip addr> (could be ipv6: or dns:)

Outbound Dial Peer rules

DNIS - destination-pattern

Sample Destination strings

dial-peer voice 6003 voip
    destination-pattern 6... (6 followed by 3 digits)
      or
    destination-pattern 6[3-5][3,5][3-5,7] (6, 2nd dig 3-5, 3rd dig 3 or 5, 4th dig 3-5 or 7)
      or
    destination-pattern 65(12)?  (65 or 6512 will match)
      or
    destination-pattern 65(12)%  (65 or 6512 or 651212 or 65121212 or ... up to 32 digits)
      or
    destination-pattern 65(12)+  (6512 or 651212 or 65121212 or ... up to 32 digits)
      or
    destination-pattern 9T  (any dial pattern beginning with 9, up to 32 digits)
    session target ipv4:<ip addr> (could be ipv6: or dns:)

Digit Manipulation (in dial-peer)

digit-strip, forward-digits, prefix may not work in voip dial peer
for voip dial peer you may have to use translation pattern

dial-peer voice 6004 pots
 destination-pattern 6...
  (default strips named digits in pots)
  no digit-strip
    or
  forward-digits 4
    and/or 
  prefix 9,5554444 (comma pauses...)
 port 0/0/0:23

Number Substitution

num-exp 2... 4000 (change 2000-2999 to 4000 - direct it out 0/0/2 below)
dial-peer voice 4000 pots
  destination-pattern 4000
  port 0/0/2

Translation Rules

http://www.cisco.com/c/en/us/support/docs/voice/call-routing-dial-plans/64020-number-voice-translation-profiles.html

Create rule

Convert dial strings before inbound dial-peer or after dial-peer rule match

Each translation rule set can have up to 15 individual rules

Only one translation rule (set) incoming per voice-port or dial-peer

Only one translation rule (set) outgoing per voice-port or dial-peer

voice translation-rule 1
  rule 1 /3456/ /4444/ (replace 3456 with 4444)
  rule 2 /^\(...\)555\(....\)/ /\1444\2/  (replace nxx of 555 match with 444) 
  rule 3 /^$/ /3000/  (replace anything (including NULL) with 3000)

test voice translation-rule 1 6665553333

apply translation rules in profiles

voice translation-profile testtrunks-out
  translate called 1 (this could be calling or redirected-called)

dial-peer voice 101 pots
  destination pattern 5....
  port 0/0/0:23
  translation-profile outgoing testtrunks-out

voice-port 1/0:1
  translation-profile incoming testtrunk-in

Debug dial-peers, translations, and profiles

show dial-peer voice summary
show dial-peer voice <#>
show dialplan number 53322 (or any other number you want to test - show all settings that dialpeer will kick out)
debug voip dialpeer
debug voice translation

Caller ID

dial-peer voice 101 pots
 clid network-number 5554441212 second-number strip

Destination pattern wildcard chars

Test Dialpeer, number manipulation creation

Practice configs

Codecs

Practice configs

voice-card 1
 codec complexity high (or medium - C549 choices)

high is default

high can handle medium or low (at high amounts of dsp resource usage)

voice-card 1
 codec complexity flex (or high, medium, or secure - C5510/PVDM2 and PVDM3 choices)

flex is default, flex adjusts to use right resouce for right complexity

high and medium as above

secure handles sRTP

show voice dsp (shows summary status of dsps installed

DSPs

SCCP controlling the DSPs

Practice Config

voice-card 1
    dsp services dspfarm

dspfarm profile 10 transcode
    codec g711ulaw
    codec g729ar8
    codec g729abr8
    codec g729r8
    maximum sessions 5
    associate application SCCP
    no shutdown

dspfarm profile 15 mtp (e.g. conferencing
    codec g711ulaw
    maximum sessions hardware 2
    maximum sessions software 2 (e.g. use cpu)
    associate application SCCP
    no shutdown

sccp ccm 10.20.30.40 identifier 1 priority 1 version 7.0+
    (priority - more than one can have diff priorities, version is CUCM version)
sccp local FastEthernet 3/0 (port that will communicate with CUCM?)

sccp ccm group 1
    bind interface FastEthernet 3/0 (?)
    associate ccm 1 priority 1
    associate profile 15 (dspfarm profile above)
    register TheDSPFARM1 (needs to match up on CUCM config)

CUCM - Media Resources/Conference Bridge menu

show voice port [<port num> | summary] (show details of analog voice port)
show controller [t1 | e1] (controller configuration)
show voice dsp (current dsp usage status)
test voice port [detector | inject-tone | loopback | relay | switch] 
                (test phys level pots port)
csim start <DN> (initiate outbound call, test dial-peers, translation rules, etc.)
debug dialpeer (review dialpeer matching in real time)

H.323 GW

Practice Config
H.323 is default protocol on Cisco GWs

voice service voip
  (shutdown [forced]) (shutdown service, forced means even if there are calls up)
  h323
    call start slow (this is global, default is fast)
    session transport udp (default is tcp, not recommended to modify)
    session transport tcp calls-per-connection 20 (1-9999, default is 15)
    h225 timeout tcp call-idle value 5 (how long till tear down on idle call)
                                       (0-1440 seconds, default is 10)

voice class h323 15 (set up a special class)
  call start slow (slow start vs. fast start (default in later Cisco GWs))
  h225 timeout tcp establish 3 (amount of time to hear response from remote gw - default is 15 seconds - use to speed up failover to backup gw)
  h225 timeout tcp setup 3 (amount of time to hear response (to H.225 setup msg) from remote gw - default is 15 seconds - use to speed up failover to backup gw)
voice class codec 20
  codec preference 1 g711ulaw bytes 160
  codec preference 2 g711ulaw bytes 240
  codec preference 3 g729br8
dial-peer voice 15 voip
  voice-class h323 15 (use the elements in this voice class)
dial-peer voice 30
  ! remember initiating router will request perticular codec
  ! don't forget dial-peer 0 for inbound (all codecs supported)
  voice-class codec 30 (can be used with H.323 MGCP, or SIP)

interface loopback0
  ip address 10.11.12.13 255.255.255.0
  h323-gateway voip bind srcaddr 10.11.12.13 (any outbound h.323 will use this address)

Dial-peer for H.323

dial-peer voice 3000 voip
  destination pattern 3...
  session target ipv4:192.168.2.1

show and debug commands (H.323)

show gateway (what state and version is H.323 in)
show h323 gateway h225 (show how many setup , alert, Progress, etc commands were sent received, and failed)
clear h323 gateway h225 (clear the counters for the show command)

SIP GW

Practice Config

voice service voip
  sip
    session transport udp (or tcp)
    bind control source-interface Loopback 0 (source-interface interface-id)
    bind media source-interface Loopback 0
    early-offer forced (force early offer every time, needs SW or HW MTP)
    url sips (configure sip secure at global level)
    exit
  srtp (configure secure rtp at global level)
  srtp fallback (configure fallback to rtp if srtp not supported)
  CallerID - all can be in dial peer
  signaling forward unconditional (forward display name with caller ID from PRI to terminating gw)
                                  (make sure d-chan (23) interface has isdn supp-service name calling on it)
  clid substitute name (clid number for name if none present)
  no shutdown

sip-ua
  authentication username <username> password <password> (digest auth)
  registrar <name> expires <secs> (enable sip gw to reg e.164 nums on ext phones)
  sip-server {dns:<hostname> | ipv4:<ipaddr>:[<portnum>]}
    (with above you can specify session target sip-server in dial peer as opposed to interface addresses)
  timers trying 1000 (wait for INVITE response for 1000ms)
  timers connect 1000 (wait for ACK response for 1000ms)
  timers expires 1000 (INVITE is valid for 1000ms)
  retry {invite <number> | response <number> | bye <number> | cancel <number>}
  bind all source-interface fa0/1 (can be control, media, or all - everything use this interface/ip addr)

Dial-Peers for SIP

dial-peer voice 40 voip
  session protocol sipv2
  destination pattern 3...
  session target sip-server (defined in sip-ua section above)
  voice-class sip url sips (configure sip secure for this dial peer - takes precedence over global)

dial-peer voice 90 voip
  destination-pattern 9T
  session target ipv4:<ipaddr>
  session protocol sipv2
  dtmf-relay {rtp-nte [digit-drop] | sip-notify} (out of band dtmf - good for low bw codecs)
  clid strip pi-restrict (strip clid if it's designated as private on ISDN)

timers and retries

sip-ua
    timers ?
    more important

 
    buffer-invite  Time to buffer the INVITE while waiting for display info
    connect        Time to wait for confirmation a session connected
    connection     Connection related timers
    disconnect     Time to wait for confirmation a session disconnected
    expires        Time to wait for the expiration of an INVITE request
    hold           Time to wait during hold before disconnecting
    info           Time to wait before INFO retransmission
    keepalive      Options keepalive related timers
    notify         Time to wait before NOTIFY retransmission
    options        Time to wait before OPTIONS retransmissions
    prack          Time to wait before starting PRACK retransmission
    refer          Time to wait before REFER retransmission
    register       Time to wait before REGISTER retransmission
    rel1xx         Time to wait before starting reliable 1xx retransmission
    trying         Time to wait for provisional response to INVITE
    update         Time to wait before starting UPDATE retransmission

    retry ?
    more important

 
    bye        BYE retry value
    cancel     CANCEL retry value
    info       INFO retry value
    invite     INVITE retry value
    keepalive  KEEPALIVE retry value
    notify     NOTIFY retry value
    options    OPTIONS retry value
    prack      PRACK retry value
    refer      REFER retry value
    register   REGISTER retry value
    rel1xx     Reliable 1xx response retry value
    response   Response Methods retry value
    subscribe  SUBSCRIBE retry value
    update     UPDATE retry value

Sample SIP GW cfg

voice service voip
  sip
    bind control source-interface Loopback0
    bind media source-interface Loopback0
    registrar server expires max 600 min 60
!
dial-peer voice 1 voip
    description test01 DID
    preference 1
    destination-pattern +180055512..
    progress_ind setup enable 3
    modem passthrough nse codec g711ulaw
    voice-class codec 1
    session protocol sipv2
    session target ipv4:192.168.2.10
    incoming called-number .
    dtmf-relay rtp-nte
    no vad
!
dial-peer voice 2 voip
    description Test02 DID
    preference 2
    destination-pattern +180055512..
    progress_ind setup enable 3
    modem passthrough nse codec g711ulaw
    voice-class codec 1
    session protocol sipv2
    session target ipv4:192.168.3.10
    dtmf-relay rtp-nte
    no vad
!
sip-ua
    retry invite 2
    retry response 2
    retry bye 2
    retry cancel 2

show and debug commands (SIP)

show sip service
show sip-ua status (options, proxy, redirect, sdp settings)
show sip-ua register status
show sip-ua timers (default sip timers)
show sip-ua retry (default sip retries)
show sip-ua connections
show sip-ua calls (SIP info on calls that are up)
show sip-ua calls brief (lists all the call legs that are up)
show call active voice summary
show sip-ua statistics (SIP successes and failure counters)

debug asnl events - verify SIP Subscription svr is up

debug voip ccapi inout - call control - very active - use sparingly

debug voip ccapi protoheaders - displays msgs sent to/from orig and term gws

debug ccsip all all ccsip - very active - use sparingly

debug ccsip calls

debug ccsip errors

debug ccsip events - events such as call setups, connections, disconnections. Good place to start.

debug ccsip info - SIP Svc Prov Interface info

debug ccsip media - media streams

debug ccsip messages - SIP message headers between client and svr

debug ccsip preauth

debug ccsip states

ccsip transport - TCP or UDP process

MGCP GW

Practice Config

mgcp package-capability ?
  as-package      Select the Announcement Server Package
  atm-package     Select the ATM Package
  dtmf-package    Select the DTMF Package
  fm-package      Select the FM Package
  fxr-package     Select the FXR Package
  gm-package      Select the Generic Media Package
  hs-package      Select the Handset Package
  it-package      Select the IT Package
  line-package    Select the Line Package
  mdr-package     Select the MDR Package
  mf-package      Select the MF Package
  pre-package     Select the PRE Package
  res-package     Select the RES Package
  rtp-package     Select the RTP Package
  script-package  Select the Script Package
  srtp-package    Select the SRTP Package
  sst-package     Select the SST Package
  trunk-package   Select the Trunk Package

Residential GW

mgcp
ccm-manager mgcp
mgcp call-agent 192.168.1.2 service-type mgcp
  or
mgcp call-agent 192.168.1.2 2427 service-type mgcp version 0.1
mgcp package-capability line-package (default package for residential gws - clid, hook flash, reorder tones...)
mgcp package-capability dtmf-package (dtmf tones)
mgcp package-capability gm-package (generate media events, signal generic events such as congestion, fax tones, ringback)
mgcp package-capability rtp-package (generage rtp event msgs such as continuity tones, tests, jitter buffer mod, RTP/RTCP timeouts

Trunking GW

mgcp
ccm-manager mgcp
mgcp call-agent 192.168.1.2 service-type mgcp
  or
mgcp call-agent 192.168.1.2 2427 service-type mgcp version 0.1
mgcp package-capability trunk-package 
mgcp package-capability dtmf-package 
mgcp package-capability gm-package 
mgcp package-capability rtp-package

controller t1 1/0/1
    ds0-group 0 timeslots 1-24 type none service mgcp
        or
    pri-group timeslots 1-24 service mgcp

show and debug commands (MGCP)

show mgcp profile (basic config settings)
show mgcp (status, timers, packages, codecs)
show mgcp statistics (how many of each command successful and failed)
show ccm-manager (make sure you're registered to CUCM)

Notes not necessarily in exams

Debugging RTP

debug voip rtp sess name
show voip rtp connections

Debugging fax

debug fax relay t30

QoS configuration

class-map match-any media
  match ip dscp ef
class-map match-any control
  match ip dscp cs3
  match ip dscp af31
class-map match-any qos-gateway-critical-traffic
  match ip dscp cs6
!
!
policy-map voip
  class media
    bandwidth percent 50
  class control
    bandwidth percent 5
  class qos-gateway-critical-traffic
    bandwidth percent 5
  class class-default
    fair-queue

dial-peer voice 29 voip
destination-pattern [2-9].........
  preference 1
  session protocol sipv2
  session target ipv4:<ipaddr>
  voice-class codec 1
  dtmf-relay rtp-nte sip-notify
  ip qos dscp cs3 signaling
  ip qos dscp ef media
  voice-class sip options-keepalive
  no vad

voice class codec 1
  codec preference 1 g711ulaw
  codec preference 2 g729r8

Studying for IIUC 640-460

Configuring Voice VLAN

conf t
    vlan 20
        name Data
    vlan 30
        name Voice

    int fa0/2 (or interface range fa0/5 - 12)
        switchport mode access
        switchport access vlan 20
        switchport voice vlan 30

show vlan brief

Test yourself
http://ciscocertstudyblog.blogspot.com/2010/06/voice-vlans.html

Studying for CVoice

Traditional Telephony

Tie Trunk

CO trunk

Interoffice trunks - between COs

Key system - lines shared across all phones - external phone call focus

PBX

IPT Unified Communications Model/Tiers

Endpoints, applications, call processing agents, network infrastructure

Endpoints

Wired IP Phones

small business
SPA 300 phones
SPA 500 phones
- support SPCP (Smart Phone Control Protocol)- Cisc Proprietary
- work with UC500 series platform
- also suport SIP (e.g. offered by a ITSP (Internet Telephony Service Provider))

enterprise
9900 series
8900 series
7900 series
6900 series
3900 series - basic needs, public access areas

Wireless

7921G and 7925G

Softphones

IP Communicator (Windows)
Personal Communidcator - integrates voice voicemail, im, other features - WIndows and Mac
Mobile Communicator - software package (integrates UC environment) on iPhone, Blackberry

Video Phones/Tablets

7985G - hi res camera, color LCD
9951 - ehternet or wifi, touchscreen, hi res camera
Video Advantage - application on Windows
Cius

Analog to IP adapter

ATA 180 - 2 port adapters
VG200 series adapters - VG224, 248 2, 4 - scheduled to replace

Applications

CER
Unity/Unity ConnectoinCisco Converenc Connection suite
billing applications

Call Processing Agents

Call Manager/Unified Communications Manager
UC Manager Business Edition - limit500 endpoints on eadchch appliance, no hgh availabilty/redundancy, integrated voicemail (unity connection)
UC Manager Express - runs on Cisco routers - geared to business with up to 250 endpoints

Network Infrastructure

Voice Gateways

use H.323, SIP, MGCP, or SCCP
older

1700

2600XM

3700

slightly newer

1800 ISR, 2800 ISR, 3800 ISR

newer

other

100 ASR
9000 ASR
6500
7200
7600
12000
AS5400
AS5800

UC Deployment models

Centralized - single building

Distributed - hub and spoke - remote sites have high speed wan links - Call Manager at central site - SRST at remote locations

Inter-Networking - Multiple large and geographically dispersed sites, or multiple sites with unreliable WAN links - Call Manager at each site

Goegrphical Diversity - Multiple sites, good connectivity. Call Manager at each site, all in one cluster

Analog and Digital ports

FXS

FXO

E&M - typically interconnect PBX systems - 6 or 8 wires, 2 wor 4 wire for signaling, 2 pairs for voice comm

Analog Signaling

Address Signaling

Pulse dialing

DTMF dialing

Informational Signaling

Call Progress tones

Dial tone, Busy, Number not in service, Call waiting, ring-back, re-order, congestion, receiver off-hook

Supervisory Signalling

Loop-Start

Common use - Home Telephones - FXS/FXO

Go off hook - ring connected through to tip by phone creating loop back to PSTN CO

2 problems

• glare
• FXO ports using loopstart may not probperly disconnect

Ground-Start

Common use - PBX-to-PBX and PBX-to-PSTN - FXS/FXO

Go off hook - PBX side will ground ring, PSTN will ground tip

• when both sides grounded, circuit connected

E&M

Wires

Wiring Types

E&M Line Seizure types