1. VoIP H323 basics , Fax management / H323, statistics

2. VoIP Typical Architecture
Media Gateway
PSTN
SS7/ISUP
H.323 Voice Gateway
ONE 400
ISDN T0/T2 IP
H.323 Voice Gateway
ONE 200
Softswitch
Gatekeeper
ISDN S0 or FXS
Copyright © OneAccess Networks - All rights reserved -

3. H.225: Call signalling channel
VoIP Basics
H.323 v4
H.323 Annex D for FAX (T38)
H.225: Call signalling channel
Q931: Call signalling
RAS: Registration, Admission, Status (with gatekeeper)
RTP/RTCP:
media streams packetization and synchronisation
H.245: Call control channel
(capabilities, master/slave, logical channels)
G.7XX: audio compression algorithms

4. H.323 Protocol Architecture
VoIP Basics
H.323 Protocol Architecture
Q.931
Terminal Control and Management
Data App
AV App
Conference Manager
G.7XX
H.26X
Terminal to Gatekeeper Signaling
H Call Signaling
T.125
H.245
T.124
RTP
RTCP
(RAS)
TPKT
T.123
Unreliable Transport (UDP)
Reliable Transport (TCP)
Network Layer (IP)
Link Layer
Physical Layer

5. H.323 Protocol Overview VoIP Basics Gatekeeper RAS (UDP) Q.931 (TCP)
Registration
Registration
Gatekeeper
RAS
(UDP)
Admission
H.323 Terminal
SETUP
SETUP
Q.931
H.323 Terminal
CONNECT
(TCP)
CONNECT
(H245 Address)
H.245 Messages
Open Logical Channels
H.245
(TCP)
(RTCP address)
(RTCP & RTP addresses)
(RTCP address)
(RTCP & RTP addresses)
RTP stream
Media
RTP stream
(UDP)
RTCP stream

6. ONEx00 (Voice gateway) interworking with: IP phones
VoIP Interoperability
ONEx00 (Voice gateway) interworking with:
IP phones
Softphone on PC (Netmeeting, Openphone,…)
Interactive Voice Response server (IVR)
Multi Conference Unit (MCU)
Gatekeeper / Softswitch
Nortel CS2K, Cirpack, Italtel, Sonus, Netcentrex,…
Media Gateway

7. Use of a gatekeeper: parameters for registration Gatekeeper identifier
VoIP: Support of H323 Gatekeepers
Use of a gatekeeper: parameters for registration
Gatekeeper identifier
Gatekeeper IP address
H.323 identifier (alias)
Gateway prefix
Alternate gatekeepers for backup
Pre-defined list (static mode)
List defined by the gatekeeper (dynamic mode)

8. Dialling: In-bloc VoIP: Dialing Modes ISDN terminal ONE200/400
Gatekeeper
Remote End
SETUP
SETUP ACK
INFO Digit 3
INFO Digit 0
INFO Digit 2
CALL PROCEEDING
ARQ N° 302
ACF
SETUP N° 302
SETUP N° 302
SETUP ACK
CALL PROCEEDING
ALERT
ALERT
ALERT
CONNECT
CONNECT
CONNECT

9. Dialing: Overlap VoIP: Dialing modes ISDN terminal ONE200/400
Gatekeeper
Remote End
SETUP
SETUP ACK
INFO Digit 3
INFO Digit 0
INFO Digit 2
ARQ N° 302
ACF
SETUP N° 302
SETUP N° 302
SETUP ACK
INFO Digit 1
INFO Digit 1
INFO Digit 1
INFO Digit 7
INFO Digit 7
INFO Digit 7
INFO Digit 9
INFO Digit 9
INFO Digit 9
CALL PROCEED.
CALL PROCEED.
CALL PROCEEDING
ALERT
ALERT
ALERT
CONNECT
CONNECT
CONNECT

10. Three types of devices are supported:
VoIP: FAX/Modem processing
Fax / Modem processing
Three types of devices are supported:
G3 FAX (up to V17 – bps)
Super G3 FAX (V34): processed as a Modem
Modem (up to V90)
The ONE200/400 detects the type of communication: voice, G3 FAX, Modem/SG3 Fax , (analysis of the FAX answer tone)
Requirements:
For Modem / SG3 FAX, the coder must be G.711 and the echo canceller must be disabled , SG3toG3 facility is also available to force the SG3 FAX in G3 mode
For G3 FAX: the Non Linear Processor is disabled. Transport in G.711 coding mode or FAX Relay T38

11. VoIP: FAX/Modem Processing
In-band processing
The coder used is G.711 Selected for all the calls or for specific subscriber numbers only (routing table)
Echo canceller automatically disabled in case of Modem
NLP automatically disabled in case of G3 Fax
FAX / Modem pass-through
Direct switchover to G.711 coder upon fax and/or modem detection (called side only). ( oneway G729 to G711 fallback) or H245 request mode or TCS null or NSE RTP packets (Cisco)
FAX Relay T38
V27 ter (4800 bps) and V29 (9600 bps)
T.30 messages analysis
Transparent transport for data (UDP) with optional redundancy
SG3toG3 fallback / allows to use same bandwidth for G3 and SG3 FAX

12. DTMF processing VoIP: DTMF Processing Transparent mode
If G.711 coder used (Voice is not compressed, DTMF signal is transported as a voice signal)
In band transmission (RTP frames carrying the DTMF tone indication instead of voice)
RFC2833 ( dedicated RTP packets with specific packet type )
H.245
Use of H.245 message ( out of band transmission )
Alphanumeric format : DTMF code. Configurable level & duration
Signal format : DTMF code + duration & level are transmitted
Detection, Suppression, Regeneration

13. Configuration diagram (1)
VoIP Configuration
Configuration diagram (1)
H.323 Gateway
Voice-port
Voice Routing
Dial-peer
Voice POTS
Dial-peer
Voice Voip
Physical Port
Interface BRI/PRI
VoIP coder
profile

14. Configuration diagram (2) 1 - Physical voice ports
VoIP Configuration
Configuration diagram (2)
1 - Physical voice ports
2 - Interface if BRI or PRI
3 - Dial-peer voice POTS
4 - Dial-peer voip
5 - VoiP coder profile
6 - Voice routing
7 - H323-gateway

15. 1 - Physical voice ports (1/3)
Voice Port Configuration
1 - Physical voice ports (1/3)
BRI (2 or 4 or 8 ports) or FXS (4 or 8 ports)
PRI
CLI# configure terminal
CLI(configure)# voice-port 5/0
CLI(voice-port)# exit
CLI(configure)# voice-port 5/1
. . .
CLI(configure)# voice-port 5/7
one200>conf t
one200(configure)>voice-port 5/0
one200(voice-port)>exit

16. 1 - Physical voice ports (2/3)
VoIP Configuration
1 - Physical voice ports (2/3)
one200(voice-port)> ?
analog-aoc-type Analog AOC type (FXS port only)
aoc-d-service method for AOC-D behaviour
aoc-e-service method for AOC-E behaviour
call-hold Set VOIP call hold
call-waiting Set VOIP call waiting
caller-id Set VOIP caller id
cas-conf Configuration of cas signal analysis
clock-source Synchro source options (global over voice ports)
coder-law Set coder law
dialing-timer Set maximum time-out for receiving 1st digit (in sec)
echo-cancellation - Set echo cancellation
echo-cancellation-le - Set echo-cancellation-length
echo-disable For echo cancellation
Modem:remove echo on 2100Hz phase reversal detection
Voicemodem: modem + reactivate echo when voice is back again
end-of-dialing-timer - Digit timeout (in sec) to consider a call as complete
exit Exit intermediate mode
force-clir Set caller line identity request
initial-ring Initial ring tone in ms for caller-id
...

17. 1 - Physical voice ports (3/3)
VoIP Configuration
1 - Physical voice ports (3/3)
...
input-gain Set input gain
inter-digit Set VOIP DTMF inter-digit duration (in sec)
isdn-release-tone - set isdn-release-tone localy and force PI
isdn-ringback-tone - set isdn-ringback-tone localy and force PI
max-ringing Maximum time for ringing before off_hook detection
metering metering choice
no no
output-gain Set output gain
power-source-one Set Power source 1 for all BRI voice-ports
pulse-dial Select country to validate pulse dial
ring Select country to define current ring
shutdown Shutdown for voice-port
sig-conf Configuration of signal analysis
signal-analysis Set signal transparency
sntp-time SNTP date/time inserted when ie is absent
tone Select a country to validate tone
tone-level Set level tone
user-metering User metering pulse profile
user-ring Modify the userdefined ring
user-tone Select the type of userdefined tone to modify:
dial, network-failure, congestion, busy, callback
without-loss-signal - Set without loss signal
<cr>

18. 2 - Interfaces (1/4) FXS: No configuration BRI PRI VoIP Configuration
one200(configure)>interface bri 5/0
one200(config-if)> ?
exit exit
isdn Set isdn level
no no
shutdown shutdown
one200(configure)>interface pri 5/0
one200(config-if)> ?
exit Exit intermediate mode
framing Set type of frames
isdn Set isdn level
linecode Select line physical code
no no
physical-interface - Select the type of interface : E1 or T1
shutdown Shutdown for the PRI interface

19. 2 - Interfaces (2/4) BRI VoIP Configuration one200(isdn)> ?
application-interfac - Set the application interface name
exit Exit to root node
facility facility message is transmit
k-window Set the value of k window
layer1-emulation Set the layer 1 emulation type
layer2-emulation Set the layer 2 emulation type
life-line-hold Life line hold for line 0 on ISDN Voice Board
max max
modulo-window Set the modulo window value
n200-counter Set the value of N200 counter
n202-counter Set the value of N202 counter
no no
operator Set the operator name
protocol-emulation - Set the type of protocol emulation
static-tei Set the value of static tei
t200-timer Set the value of T200 timer
...
t310-timer Set the value of T310 timer --> can be set to 100 for GSM calls
tei-negotiation Set the tei negociation mode
<cr>
one200(isdn)>

20. BRI: Example with ISDN phone
VoIP Configuration
2 - Interfaces (3/4)
BRI: Example with ISDN phone
BRI: Example with PBX
one200(configure)>interface bri 5/0
one200(conf-if)> isdn
one200(isdn)> protocol-emulation isdn-nt
one200(isdn)> exit
one200(conf-if)> no shutdown
one200(conf-if)> execute
one200(conf-if)> exit
one200(configure)> interface bri 5/0
one200(conf-if)> isdn
one200(isdn)> tei-negotiation static
one200(isdn)> protocol-emulation isdn-nt
one200(isdn)> exit
one200(conf-if)> no shutdown
one200(conf-if)> execute
one200(conf-if)> exit

21. 2 - Interfaces (4/4) PRI VoIP Configuration one200(isdn)> ?
application-interfac - Set the application interface name
exit Exit to root node
facility message facility is transmit
k-window Set the value of k window
layer2-emulation Set the layer 2 emulation type
max max
n200-counter Set the value of N200 counter
no no
operator Set the operator name
protocol-emulation - Set the type of protocol emulation
t200-timer Set the value of T200 timer
t203-timer Set the value of T203 timer
t301-timer Set the value of T301 timer
t302-timer Set the value of T302 timer
t303-timer Set the value of T303 timer
t304-timer Set the value of T304 timer
t305-timer Set the value of T305 timer
t306-timer Set the value of T306 timer
t308-timer Set the value of T308 timer
t309-timer Set the value of T309 timer
t310-timer Set the value of T310 timer
t313-timer Set the value of T313 timer

22. 3 - Internal Local Voice Port (POTS)
Logical Local Voice Port
3 - Internal Local Voice Port (POTS)
CLI(configure)# dial-peer voice pots <id>
CLI(pots)# pots-group <port>
CLI(pots)# port 5/<port>
CLI(pots)# no shutdown
CLI(pots)# exit
One Dial-peer voice POTS must be configured for each physical voice port.
It binds a physical port to a pots-group. Several physical ports can be bound to the same pots-group.
Calls are then routed to pots-group.

23. Logical Local Voice Port
- For an outgoing Voip call, user part of From header field is based on 6C IE (calling party number) for BRI interface.
For FXS port, that information must be added at the dial-peer voice pots adding “insert-calling-number”.
That will be used also for 40x challenge on Invite method (to resolve user and its digest username and password.
CLI(configure)# dial-peer voice pots <id>
CLI(pots)# pots-group <id>
CLI(pots)# port 5/<port>
CLI(pots)# insert-calling-number <E164 number>
CLI(pots)# no shutdown
CLI(pots)# exit

24. 3 – Local Voice Port optional parameters
Logical Local Voice Port
3 – Local Voice Port optional parameters
one200(configure)> dial-peer voice pots 0
one200(pots)> ?
bearer-cap Payload category
direct-call Set direct call number
exit Exit intermediate mode
implicit-routing Sets implicit routing to specified
pots group or voip dial peer
insert-calling-numbe - Set VOIP insert calling number
no no
port Links local suscriber and voice port
pots-group Set VOIP pots group
priority Set priority
service to provide a service by the voice pots.
shutdown Shutdown for dial peer POTS
suppress-calling-num - Set VOIP suppresion of the calling number
<cr>
one200(pots)>

25. 4 - Dial-peer VoIP (1/2) VoIP Configuration
one200(configure)> dial-peer voice voip 0
one200(voip)> ?
aoc-format Set VOIP remote AOC coding format
call-media-independa - Set VOIP call media independant
dtmf-relay Set VOIP dtmf relay
exit Exit from command node
fast-connect Set VOIP fast connect
fax-relay Set VOIP fax relay
force-rec-inband Force reception of inband in Alert
gatekeeper Set VOIP gatekeeper
gw-ip-address Set VOIP gateway
h245-tunnel Set VOIP H245 tunnel
implicit-routing Set implicit routing
jitter Set VOIP jitter
jitter-compensation - Set VOIP jitter comp
max-conn Set VOIP maximum call allowed
modem-passthrough - Set VOIP modem passthrough
NdiInsourceAddress - Force NDI in H323 sourceAddress
no no
shutdown Shutdown voip dial peer
silence-detection - Set VOIP silence detection
t38-redundancy Set VOIP T38 redundancy
voip-coder-profile - Set VOIP coder profile

26. 4 - Dial-peer VoIP (2/2) VoIP Configuration
one200(configure)>dial-peer voice voip 0
one200(voip)>fast-connect
one200(voip)>gatekeeper mandatory
one200(voip)>voip-coder-profile 0
one200(voip)>no shutdown
one200(voip)>exit

27. 5 - The following coders are supported:
Codec Profiles 1/2
5 - The following coders are supported:
G.711 A law (64Kbps)
G.711 law (64Kbps)
G.729A (8 Kbps, no silence suppression)
G.729AB (8 kbps, optional silence suppression)
CLI(configure)# voip-coder-profile 0
CLI(voip-coder)# codec ?
<pref-index> - Codec preference index: 0..8
CLI(voip-coder)# codec 0 ?
<coder> - Coder type: g729ab | g711a | g711u
CLI(voip-coder)# codec 0 g729ab ?
<timestamp> - Timestamp value: depending on the coder
<cr>
one200(voip-coder)# codec 0 g729ab 30
one200(voip-coder)# codec 1 g711a 20
one200(voip-coder)# exit

28. Codec Profiles 2/2 one200>show running ... voip-coder-profile 0
codec 0 g729ab 30
codec 1 g711a 20
exit
voip-coder-profile 1
codec 0 g711a 20

29. 6 - VoIP Routing table (1/10)
VoIP Configuration
6 - VoIP Routing table (1/10)
one200(configure)>voice-routing
one200(voice-route)> ?
display-routes Show voice routing table
exit Exit from command node
insert Insert VOIP voice route
move Move VOIP voice route
no no
route Set VOIP voice route
test-route Test voice routing table
<cr>
one200(voice-route)>

30. 6 - VoIP Routing table (2/10)
VoIP Configuration
6 - VoIP Routing table (2/10)
one200(voice-route)>route 1
one200(conf-voice-route)> ?
dial-peer Set route dial peer
exit Exit from command node
force-bearer-cap Set force bearer capability fields
force-clir Set caller line identity request
force-numplan Set origin/destination numplan
insert-calling Set route calling insertion
insert-prefix Set route prefix insertion
insert-suffix Set route suffix insertion
no no
prefix Set route prefix(prefix [number-type][length][timer]
[overlap])
prefix-type Set route direction and type of call
startup-file Set restart equipment with the file name.
suppress-prefix Set route prefix suppression
translate Set route number translate
wildcard Set wildcard value for prefix
<cr>

31. 6 - VoIP Routing table (3/10)
VoIP Configuration
6 - VoIP Routing table (3/10)
one200(configure)>voice-routing
one200(voice-route)>route 1
one200(conf-voice-route)>dial-peer pots-group 0 alias
one200(conf-voice-route)>prefix-type outgoing called last
one200(conf-voice-route)>prefix 110 length 3
one200(conf-voice-route)>exit
one200(voice-route)>route 2
one200(conf-voice-route)>dial-peer voip 0
one200(conf-voice-route)>prefix 11X length 3
one200(voice-route)>exit
one200(configure)>exit
one200(voice-route)>display-routes
all length 3 / pots 0 / outgoing - called - last
2 – 11X all length 3 / voip 0 / outgoing - called - last
one200(voice-route)>

32. 6 - VoIP Routing table (4/10)
VoIP Configuration
6 - VoIP Routing table (4/10)
one200>show running-config
...
voice-routing
route 1
dial-peer pots-group 0
prefix 110 length 3
prefix-type outgoing called last
exit
route 2
dial-peer voip 0
prefix 11X length 3

33. 6 - VoIP Routing table (5/10)
VoIP Configuration
6 - VoIP Routing table (5/10)
1234
Pots-group 0
VoIP 0
route 1
dial-peer pots-group 0
prefix 120A. length 4
suppress-prefix 4 called
wildcard A 0123
exit

34. 6 - VoIP Routing table (6/10)
VoIP Configuration
6 - VoIP Routing table (6/10)
Pots-group 0
Incoming
Table
VoIP 1
VoIP 2
Pots-group 4
route 14
dial-peer voip 1
prefix length 9
prefix-type incoming called
exit
route 15
dial-peer voip 2
prefix . length 0

35. 6 - VoIP Routing table (7/10)
VoIP Configuration
6 - VoIP Routing table (7/10)
ing :
ing :
ed :
Incoming
Table
Outgoing
Table
ed :
ing :
ed :
ing :
ed :
Pots-group 0

36. 6 - VoIP Routing table (8/10)
VoIP Configuration
6 - VoIP Routing table (8/10)
route 2
dial-peer pots-group 0
translate A A calling
prefix . length 0
prefix-type incoming calling next
exit
route 3
dial-peer pots-group 0 alias
insert-prefix 1200 called
prefix length 9
prefix-type incoming calling
alias added to register the number 1200 in the RAS

37. 6 - VoIP Routing table (9/10)
VoIP Configuration
6 - VoIP Routing table (9/10)
ing :
ed :
ing :
ing :
ed :
Incoming
Table
Outgoing
Table
ed :
Voip 1
ing :
ed :
ing :
ed :
Voip 0
Pots-group 0
ing :
ed :

38. 6 - VoIP Routing table (10/10)
VoIP Configuration
6 - VoIP Routing table (10/10)
route 4
dial-peer voip 1
prefix length 4
prefix-type outgoing called last
exit
route 5
dial-peer voip 0
prefix . length 0

39. Voice routing table testing
One_Training(voice-route)>display-route
1 - . all length 10 / pots 0 / incoming - called - last
/ +prefix-calling 0
all timer / voip 0 / outgoing - called - last
all length 9 / pots 0 / outgoing - called - last
/ -prefix-called 1
One_Training(voice-route)>
One_Training> configure terminal
One_Training(configure)>voice-routing
One_Training(voice-route)>test-route from-voip 0
--> test_route : Calling= Called=
Route match : Calling= Called=
Incoming Routes = Outgoing Routes = 20,
Incoming call from voip id:0 Send towards Isdn on local port: 5/0
One_Training(voice-route)>test-route from-pots 0
--> test_route : Calling= Called=
Route match : Calling= Called=
Incoming Routes = 1, Outgoing Routes = 10,
Outgoing call from local port: 5/0 -> Send towards H323 on voip id:0

40. Must be shutdown to modify parameters
VoIP Configuration
7 - H.323 Gateway (1/6)
Global parameters for H.323 gateway (gatekeeper, RTP ports, timeouts,…)
Must be shutdown to modify parameters
one200(configure)>h323-gateway one200(h323gw)>

41. 7 - H.323 Gateway (2/6) VoIP Configuration one200(h323gw)> ?
alt-gatekeeper Set H323 gateway - alternative gatekeeper
altgk-list Set H323 gateway - alternative gatekeeper list
altgk-mode Set H323 gateway - alternate gatekeeper mode
altgk-timeout The timeout, in seconds, to check Primary gatekeeper status when registered to an alternate gatekeeper.
bandwidth-control - Set H323 gateway - bandwidth control
call-test Set H323 gateway - set call testing when gateway is ready.
callsig-port Set H323 gateway - H225/Q931 listening TCP port
-id Set H323 identifier for the gateway
exit Exit from command node
gatekeeper Set H323 gateway - main gatekeeper parameters
gw-address Set H323 gateway - gateway address mode
gw-interface Output Interface category for H323 GW - default fastethernet 0
gw-interface-bw-ctrl - Set H323 gateway - gw interface bandwidth control
gw-prefix Set H323 gateway – prefix
h235-authentication - Set H235 authentication
h245-response-timeou - Set H323 gateway - timeout used for H245 protocol

42. 7 - H.323 Gateway (3/6) VoIP Configuration
h323-id Set H323 identifier for the gateway
max-bandwidth Set H323 gateway - maximum bandwidth allowed
no no
payload-64k-unrestricted - Set H323 gateway - payload for 64k unrestricted
polling Set H323 gateway
q931-connection-timeout - Set H323 gateway (timeout for receiving CONNECT message)
q931-response-timeout - Set H323 gateway (timeout for the response to a SETUP message)
ras-bandwidth-control - Set H323 gateway - bandwith control by gatekeeper
ras-full-rrq Set H323 gateway - timeout used to send ras full registration
ras-intrusive-voiceport - Set H323 gateway - register/unregister on voice-port condition
ras-keepalive-timeout - Set H323 gateway - keepalive timeout used for RAS
ras-max-retries Set H323 gateway - max retries for RAS protocol
ras-multicast Set H323 gateway (multicast address and port for gatekeeper discover)
ras-port Set H323 gateway - UDP port used for RAS protocol

43. 7 - H.323 Gateway (4/6) VoIP Configuration
ras-response-timeout - Set H323 gateway - timeout used for RAS protocol
ras-timetolive Set H323 gateway - timetolive used for RAS protocol
register Register gateway
resource Set H323 gateway - Set h323 resource parameters
rtp-dscp Set H323 gateway (DSCP field value for transmitted RTP packet)
rtp-port-range Set H323 gateway - UDP port range used for RTP
rtp-uplink-analysis - Enable or disable the rtp jitter analysis
set-portability Set H323 portability
shutdown shutdown
sig-dscp Set H323 gateway (DSCP field value for transmitted signalling packet)
snmp-sysdescr-hw-ident - add hw ident to sysdescr

44. 7 - H.323 Gateway (5/6) VoIP Configuration ...
start-h245-discarded - Set H323 gateway - facility start h245 is discarded
tcp-keepalive Set H323 gateway - tcp keepalive option (default)
unregister Unregister gateway
<cr>

45. 7 - H.323 Gateway (6/6) VoIP Configuration
one200(config)> h323-gateway
one200(h323gw)> gw-interface fastethernet 0/0
one200(h323gw)> gatekeeper id training address
one200(h323gw)> h323-id GW1
one200(h323gw)> max-bandwidth
one200(h323gw)> no shutdown

46. BRI statistics VoIP Statistics CLI# show voice voice-port bri index 0
protocol descriptor BRI_NT
current state activated
config state up
layer 1 status activated
attached vmoabri dial peer
number of voice communication 0
bri Tx frames on D channel
bri Rx frames on D channel
Outgoing calls : 102
Outgoing calls failures : 5
Physical Interface down : 0
Cause Class 0 (normal event) : 0
Cause Class 1 (normal event) : 5
Normal Cause (16) : 2
User busy (17) : 3
No answer (18) : 0
Cause Class 2 (unavailable ressources) : 0
Cause Class 3 (unavailable service) : 0
Cause Class 4 (service not provided) : 0
Cause Class 5 (invalid message) : 0
Cause Class 6 (protocol error) : 0
Cause Class 7 (interworking) : 0
Incoming calls : 54
Incoming calls failures : 7
Remote failure : 0
Unknown number : 5
DSP unavailable : 0
Not specified : 2

47. PRI statistics VoIP Statistics CLI# show voice voice-port pri index 0
physical type E1
protocol descriptor E1_PRI
current state activated
config state up
layer 1 status deactivated
number of voice communications 0
pri AIS occurence
pri RDI occurence
Outgoing calls : 67
Outgoing calls failures : 3
Physical Interface down : 0
Cause Class 0 (normal event) : 0
Cause Class 1 (normal event) : 3
Normal Cause (16) : 0
User busy (17) : 3
No answer (18) : 0
Cause Class 2 (unavailable ressources) : 0
Cause Class 3 (unavailable service) : 0
Cause Class 4 (service not provided) : 0
Cause Class 5 (invalid message) : 0
Cause Class 6 (protocol error) : 0
Cause Class 7 (interworking) : 0
Incoming calls : 23
Incoming calls failures : 2
Remote failure : 2
Unknown number : 0
DSP unavailable : 0
Not specified : 0

48. FXS statistics VoIP Statistics CLI# show voice voice-port fxs index 0
current state on hook
config state up
attached vmoa fxs dial peer 0
voice communication no
Outgoing calls : 32
Outgoing calls failures : 3
User busy : 2
No answer : 1
Incoming calls : 6
Incoming calls failures : 0
Remote failure : 0
Unknown number : 0
DSP unavailable : 0
Not specified : 0

49. Dial-peer VoIP Statistics (1)
one200> show voice dial-peer voice voip type index <port id> [reset] or
one200> show voice dial-peer voice voip type global[reset]
type may be :
current : statistics on current calls
outgoing : outgoing calls only
incoming : incoming calls only
user-plan : voice & fax only
all (default) : all the statistics are provided

50. Dial-peer VoIP Statistics (2): Outgoing Calls
Current Calls
Outgoing Calls
Outgoing calls failures
RAS Call Failures
Gatekeeper Unavailable
Admission Rejects
H225/Q931 Call failures
Cause Class 0 (normal event)
Cause Class 1 (normal event)
Normal Cause (16)
User busy (17)
No answer (18)
Cause Class 2 (unavailable ressources) 0
Cause Class 3 (unavailable service) 0
Cause Class 4 (service not provided) 0
Cause Class 5 (invalid message)
Cause Class 6 (protocol error)
Cause Class 7 (interworking)
H245 Call failures
Incompatible capabilities
Protocol errors
Internal call failures
DSP unavailable
Max-bandwidth exceeded
Max-connection exceeded
Not specified

51. Dial-peer VoIP Statistics (3): Incoming Calls
Incoming calls failures
RAS Call failures
Gatekeeper Unavailable
Admission Rejects
Local Port Call failures
H245 Call failures
Incompatible capabilities
Protocol errors
Internal call failures
DSP unavailable
Unknown number
Channel / port unavailable
Max-bandwidth exceeded
Max-connection exceeded
Not specified

52. Dial-peer VoIP Statistics (4): Voice and Fax
RTP statistics
Number of transmitted packets
Number of received packets
Number of transmitted bytes
Number of received bytes
Number of excessive jitter events
Number of lost packets
Number of invalid packets
Number of calls with frame error rate
total <0.01<0.1<0.5<1<5>=5
Modem passthrough
Number of switching to modem mode
T38 FAX Calls
Number of outgoing fax
Number of incoming fax
Number of failures
Request Mode failure
Pre-message procedure failure
Page failure
Number of transmitted packets
Number of received packets
Number of transmitted bytes
Number of received bytes

53. H.323 Gateway Statistics VoIP Statistics CLI# show voice h323-gateway
Registration state : registered
Gatekeeper identifier : training
Gatekeeper address :
Registration requests : 67
Registration failures : 1
No response : 1
Invalid IP address : 0
Duplicate alias : 0
Invalid terminal type : 0
Ressource unavailable : 0
Invalid alias : 0
Security denial : 0
Undefined reason : 0
Admission requests : 1345
Admission rejects : 4
Called party not registered : 4
Invalid permission : 0
Request denied : 0
Caller not registered : 0
Resource unavailable : 0
Security denial : 0
Invalid Endpoint Ident : 0
Incomplete address : 0
Not specified : 0
Undefined reason : 0

54. Events VoIP Statistics vxTarget>event
filter Add/remove events filters
manager Add a SNMP manager
no No
recover Recover events from memory
vxTarget>event filter
add Add an event filter
remove Remove a events filter from the table
vxTarget>event filter add vox
ALL All families from vox group
GEN GEN
VOATM VOATM
VOIP VOIP
vxTarget>event filter add vox voip
<subfam> <ALL | ControlPlan | UserPlan>
<fam2> <GEN | VOATM>
vxTarget>event filter add vox voip all show

55. Voice call history, active calls
VoIP Statistics
Voice call history, active calls
Gives statistics on the current voice calls and the last 100 calls
vxTarget>show voice voip-call any ind 1
1 - Call from remote voip: 0, to local port: 5/1 call-id: 4 active
calling : 110, called : 111
setup time: 01/02/00 04h58m31s
01/02/00 04h58m31s
RTP Source ip : rtp:16384 /Dest ip : rtp:16386 (active)
Play time (voice) : 00h00m39s
Tx Coder : G729 / 20 ms ; Rx Coder : G729
RTP Packets RX / TX : 1988 / 1989
RTP Packet lost&discarded RX / TX (RTCP reported) : 0 /
Number of Excessive Jitter events : 3

56. RTP sessions history VoIP Statistics
Gives complete statistics about the 200 last RTP sessions
CLI> show voice rtpcall full any ind 2
2 - 01/04/01 00h47m24s RTP :16384 – :16386
Play time (voice) : 00h00m46s
Tx Coder : G729 / 20 ms ; Rx Coder : G729
VAD enabled local / remote : no / no
ERL : 15 dB
ACOM : 32 dB
RTP Packets received (DSP / Uplink) : 2337 / 2337
lost : 0
out of sequence : 0
invalid : 0
RTP Packets transmitted (DSP / Uplink) : 2338 / 2338
lost (RTCP reported) : 0
Jitter parameter : 100 ms
Number of Excessive Jitter events : 1

57. RTP sessions history (continue)
VoIP Statistics
RTP sessions history (continue)
Excessive Jitter events : 2|
1| *
0 30" 1' 2' 4' 8' 12' >16'
Jitter received (uplink) :
Max delay : 93 ms
Delays (ms) >50 >100 >150 >200 >300
Nb of occur
Interarrival max jitter : 9 ms
Jitter received (DSP) :
Frames with a delay >50 ms :
1| * *
Jitter transmitted (uplink) :
Max delay : 6 ms
Nb of occur
Interarrival max jitter : 1 ms
(RTCP reported) : 2 ms 

58. VoIP: Internal Call Generator
Possibility to generate and / or terminate one or several VoIP calls
Two services: RTP loopback or BERT testing
Use of virtual and routable dial-peer pots
dial-peer voice pots 4
service bert2047 both 3
pots-group 0
exit
voip-call 1
pots 4
called
calling 3000
bearer data
duration 180
timeout 10
One200> start identifier 1

59. VoIP: ISDN capture
Possibility to capture the signalling traffic over ISDN BRI & PRI interfaces: layer 1 to 3
For VoIP side: use of the IP capture possibilities
CLI>conf t
CLI>logging buffered debug
CLI>exit
CLI>debug isdn all layer 1to3
00:07: line:5/0 L1 frame sent.
00:07: line:5/0 L2 tx UI P/F=0 NR=4 NS=2 C/R=1.
00:07: hex: 02 ff 03
00:07: line:5/0 L3 tx SETUP callref:8.
00:07: hex1: a a1 31
00:07: hex2: a1
00:07: line:5/0 L1 frame received.
00:07: line:5/0 L2 rx SABME P/F=1 C/R=0.
00:07: hex: f
00:07: line:5/0 L1 frame sent.
00:07: line:5/0 L2 tx UA P/F=1 NR=4 NS=2 C/R=0.
00:07: hex:

60. VoIP H323 signaling messages can be displayed for debug.
VoIP debug
VoIP H323 signaling messages can be displayed for debug.
One_training>debug h323
One_training>
One_training>00:06: H.323 rx TCP-H.225 Msg=Setup IP= :1882 callref=9203.
00:06: Info vox voip controlplan 3 Incoming call on voip id: 0, calling: , called:
, call-id: 1.
00:06: Info vox voip controlplan 3 Outgoing call on local port: 5/0, calling: , called: 7383, call-id: 1.
00:06: Info vox voip controlplan 3 Alert received, call-id: 1.s
00:06: Info vox voip userplan 3 VoIP RTP thransmission started, coder: G729, call- id: 1.
00:06: H.323 tx TCP-H.225 Msg=CallProceeding IP= :1882 callref=41971.
00:06: H.323 tx TCP-H.225 Msg=Alerting IP= :1882 callref=41971.
00:06: H.323 rx TCP-H.225 Msg=Facility IP= :1882 callref=9203.
00:06: H.323 tx TCP-H.225 Msg=Facility IP= :1882 callref=41971.
00:06: H.323 rx TCP-H.225 Msg=Facility IP= :1882 callref=9203.
00:06: H.323 tx TCP-H.225 Msg=Facility IP= :1882 callref=41971.
00:06: H.323 tx TCP-H.225 Msg=Facility IP= :1882 callref=41971.
00:06: H.323 rx TCP-H.225 Msg=Facility IP= :1882 callref=9203.
00:06: Info vox voip controlplan 3 Call connected, call-id: 1.
00:06: H.323 tx TCP-H.225 Msg=Connect IP= :1882 callref=41971.
00:06: Info vox voip userplan 3 VoIP RTP reception started, coder: G729, call-id: 1.

61. For debug, a SETUP can be sent on VoIP.
Call factory over IP
For debug, a SETUP can be sent on VoIP.
One_training>auto-call
<called> called number: up to 21 characters <0..9, #, *>
One_training>auto-call
<calling> calling number: up to 21 characters <0..9, #, *>
<pots-number> pots: 0..29
<bearer> bearer capability < voice | data | voiceband >
overlap units in milliseconds: <0 means no overlap used>
<cr>
One_training>auto-call
17:50: Info vox factory test 1 call-id: 4, ident: auto-call, CALL IN PROGRESS Calling=
005 Called=
one100_interopBW>17:50: Info vox voip controlplan 3 Incoming call on local pots: 0, calling:
, called: , call-id: 4.
17:50: Info vox voip controlplan 3 Outgoing call on voip id: 0, calling: , called:
, call-id: 4.
17:50: Info vox factory test 1 call-id: 4, ident: auto-call, CALL FAILED cause=no codec.
17:50: Info vox factory test 1 call-id: 4, ident: auto-call, CALL FAILED on pots cause=[Normal call clearing].

62. For debug, a ‘SETUP’ can be sent on a ISDN local port.
Auto call to ISDN
For debug, a ‘SETUP’ can be sent on a ISDN local port.
One_training>isdn test call ( data call/unrestricted )
02:27: line:5/0 L1 event received PH_AR State:F3.
02:27: line:5/0 L1 event received EV_LOST_FRAMING State:F4.
02:27: line:5/0 L1 event received EV_INFO_2 State:F5.
02:27: line:5/0 L1 event received EV_INFO_4_8(PH_AI) State:F6.
02:27: line:5/0 L1 event received MPH_AI State:F7.
02:27: line:5/0 L1 frame sent.
02:27: line:5/0 L2 tx SABME P/F=1 C/R=0.
02:27: hex: f
02:27: line:5/0 L1 frame received.
02:27: line:5/0 L2 rx UA P/F=1 C/R=0.
02:27: hex:
02:27: line:5/0 L1 frame sent.
02:27: line:5/0 L2 tx INFO P=0 NR=0 NS=0 C/R=0.
02:27: hex:
02:27: line:5/0 L3 tx SETUP callref:4.
02:27: hex1:
02:27: hex2: a1
02:27: Called Number : 85841

63. Fax T38 processing and traces: display of the T30 messages
Fax relay T38 debug
Fax T38 processing and traces: display of the T30 messages
One_training>trace filter add vox up ifp 2 show
01:42: Info vox voip controlplan 3 Outgoing call on voip id: 0, calling: 110,
called: 111, call-id: 29.
01:42: Info vox voip controlplan 3 Alert in band received, call-id: 29.
01:42: Info vox voip userplan 3 VoIP RTP transmission started, coder: G729, call-id: 29.
01:42: Info vox voip userplan 3 VoIP RTP reception started, coder: G729, call-id: 29.
01:42: Info vox voip controlplan 3 Call connected, call-id: 29.
01:42: Info vox voip userplan 3 VoIP RTP transmission stopped, coder: G729, call-id: 29.
01:42: Info vox voip userplan 3 VoIP RTP reception stopped, coder: G729, call-id: 29.
01:42: Info vox voip userplan 3 Fax T38 starting call-id: 29 .
01:43: VOX.up.ifp NSF / CSI / DIS >
01:43: VOX.up.ifp.2 < TSI / DCS
01:43: VOX.up.ifp.2 < TCF (v29_9600)
01:43: Info vox voip userplan 1 T38 Pre-message procedure OK, call-id: 29.
01:43: VOX.up.ifp CFR >
01:43: Info vox voip userplan 1 T38 Transmitting page 1, call-id: 29.
01:43: VOX.up.ifp.2 < PAGE (v29_9600)
01:51: VOX.up.ifp.2 < PPS-EOP
01:51: Info vox voip userplan 1 T38 page 1 OK, call-id: 29.
01:51: VOX.up.ifp MCF >
01:51: VOX.up.ifp.2 < DCN
01:51: Info vox voip controlplan 3 Call Disconnection received on local port: 5/2,
cause: (16)[Normal call clearing], call-id: 29.

64. Modem / fax passthrough event
Example of event in case of modem/fax call.
One_training>event filter add vox all show
00:11: Info vox voip controlplan 3 Incoming call on voip id: 0, calling: 111,
called: 110, call-id: 3.
00:11: Info vox voip controlplan 3 Outgoing call on local port: 5/0, calling: 111,
called: 110, call-id:3.
00:11: Info vox voip controlplan 3 Alert received, call-id: 3.
00:11: Info vox voip controlplan 3 Call connected, call-id: 3.
00:11: Info vox voip userplan 1 Fax/Modem Passthrough starting call-id: 3.
00:11: Info vox voip userplan 3 RTP new transmission coder: G711 A Law, call-id: 3.
00:11: Info vox voip userplan 3 RTP new reception coder: G711 A Law, call-id: 3.
00:12: Info vox voip controlplan 3 Call Disconnection received on local port: 5/1,
cause: (16)[Normal call clearing], call-id: 3.
00:12: Info vox voip userplan 3 VoIP RTP transmission stopped, coder: G711 A Law, call-id: 3.
00:12: Info vox voip userplan 3 VoIP RTP reception stopped, coder: G711 A Law, call-id: 3.