Business-Class Router Solutions – All Telecommunication Services in One Access
OneOs Voice Configuration
Dial-Peer Concept
4.11 H323
4.15 SIP
4.21 MGCP
www.oneaccess-net.com
contact@oneaccess-net.com
v0.11 2007
Copyright © OneAccess Networks – All rights reserved
VoIP Configuration Diagram (H.323 / SIP)
ONE
CLASSICAL
TELEPHONE NETWORK
ISDN PBX
2
Voice Routing
Interface
Interface
BRI/PRI
BRI/PRI
BRI
PRI
BRI
5
VoIP Coder
Profile
4
Digital
1
Physical
Digital
Voice-port
Physical
Voice-port
Dial-Peer
3 Voice
POTS
Dial-Peer Voice
POTS
POTS Group 1
ISDN Phone
POTS
IP NETWORK
6
Analogue
1
Physical
Analogue
Voice-port
Physical
Voice-port
Dial-Peer
3 Voice
POTS
Dial-Peer Voice
POTS
Dial-Peer Voice
VoIP
Dial-Peer Voice
VoIP
7
H.323 or SIP
Gateway
8
SIP Server
POTS Group 2
SIP
Phones
Copyright © OneAccess Networks – All rights reserved
2
VoIP Configuration Diagram (MGCP)
ONE
CLASSICAL
TELEPHONE NETWORK
IP NETWORK
5
VoIP Coder
Profile
4
Dial-Peer Voice
VoIP
Analogue
1
Physical
Analogue
Voice-port
Physical
Voice-port
7
MGCP
Gateway
Dial-Peer
3 Voice
POTS
Dial-Peer Voice
POTS
POTS Group 2
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3
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 – Gateway (SIP or H323)
Copyright © OneAccess Networks – All rights reserved
4
Voice Port Configuration
1 - Physical voice ports (1/3)
BRI (2 or 4 or 8 ports) or FXS (4 or 8 ports)
CLI# configure terminal
CLI(configure)# voice-port 5/0
CLI(voice-port)# exit
CLI(configure)# voice-port 5/1
CLI(voice-port)# exit
. . .
CLI(configure)# voice-port 5/7
CLI(voice-port)# exit
PRI
one200>conf t
one200(configure)>voice-port 5/0
one200(voice-port)>exit
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5
VoIP Configuration
1 - Physical voice ports (2/3)
one200(voice-port)> ?
analog-aoc-type
aoc-d-service
aoc-e-service
call-hold
call-waiting
caller-id
cas-conf
clock-source
coder-law
dialing-timer
echo-cancellation
echo-cancellation-le
echo-disable
end-of-dialing-timer
exit
force-clir
initial-ring
...
-
-
Analog AOC type (FXS port only)
method for AOC-D behaviour
method for AOC-E behaviour
Set VOIP call hold
Set VOIP call waiting
Set VOIP caller id
Configuration of cas signal analysis
Synchro source options (global over voice ports)
Set coder law
Set maximum time-out for receiving 1st digit (in sec)
Set echo cancellation
Set echo-cancellation-length
For echo cancellation
Modem:remove echo on 2100Hz phase reversal detection
Voicemodem: modem + reactivate echo
when voice is back again
Digit timeout (in sec) to consider a call as complete
Exit intermediate mode
Set caller line identity request
Initial ring tone in ms for caller-id
Copyright © OneAccess Networks – All rights reserved
6
VoIP Configuration
1 - Physical voice ports (3/3)
...
input-gain
inter-digit
isdn-release-tone
isdn-ringback-tone
max-ringing
metering
no
output-gain
power-source-one
pulse-dial
ring
shutdown
sig-conf
signal-analysis
sntp-time
tone
tone-level
user-metering
user-ring
user-tone
without-loss-signal
<cr>
-
Set input gain
Set VOIP DTMF inter-digit duration (in sec)
set isdn-release-tone localy and force PI
set isdn-ringback-tone localy and force PI
Maximum time for ringing before off_hook detection
metering choice
no
Set output gain
Set Power source 1 for all BRI voice-ports
Select country to validate pulse dial
Select country to define current ring
Shutdown for voice-port
Configuration of signal analysis
Set signal transparency
SNTP date/time inserted when ie is absent
Select a country to validate tone
Set level tone
User metering pulse profile
Modify the userdefined ring
Select the type of userdefined tone to modify:
dial, network-failure, congestion, busy, callback
- Set without loss signal
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7
VoIP Configuration
2 - Interfaces (1/4)
FXS: No configuration
BRI
one200(configure)>interface bri 5/0
one200(config-if)> ?
exit
- exit
isdn
- Set isdn level
no
- no
shutdown
- shutdown
PRI
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
Copyright © OneAccess Networks – All rights reserved
8
VoIP Configuration
2 - Interfaces (2/4)
BRI
one200(isdn)> ?
application-interfac
exit
facility
k-window
layer1-emulation
layer2-emulation
life-line-hold
max
modulo-window
n200-counter
n202-counter
no
operator
protocol-emulation
static-tei
t200-timer
...
t310-timer
tei-negotiation
<cr>
one200(isdn)>
-
Set the application interface name
Exit to root node
facility message is transmit
Set the value of k window
Set the layer 1 emulation type
Set the layer 2 emulation type
Life line hold for line 0 on ISDN Voice Board
max
Set the modulo window value
Set the value of N200 counter
Set the value of N202 counter
no
Set the operator name
Set the type of protocol emulation
Set the value of static tei
Set the value of T200 timer
- Set the value of T310 timer --> can be set to 100 for GSM calls
- Set the tei negociation mode
Copyright © OneAccess Networks – All rights reserved
9
VoIP Configuration
2 - Interfaces (3/4)
BRI: Example with ISDN phone
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
BRI: Example with PBX
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
Copyright © OneAccess Networks – All rights reserved
10
VoIP Configuration
2 - Interfaces (4/4)
PRI
one200(isdn)> ?
application-interfac
exit
facility
k-window
layer2-emulation
max
n200-counter
no
operator
protocol-emulation
t200-timer
t203-timer
t301-timer
t302-timer
t303-timer
t304-timer
t305-timer
t306-timer
t308-timer
t309-timer
t310-timer
t313-timer
-
Set the
Exit to
message
Set the
Set the
max
Set the
no
Set the
Set the
Set the
Set the
Set the
Set the
Set the
Set the
Set the
Set the
Set the
Set the
Set the
Set the
application interface name
root node
facility is transmit
value of k window
layer 2 emulation type
value of N200 counter
operator name
type of protocol emulation
value of T200 timer
value of T203 timer
value of T301 timer
value of T302 timer
value of T303 timer
value of T304 timer
value of T305 timer
value of T306 timer
value of T308 timer
value of T309 timer
value of T310 timer
value of T313 timer
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11
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 rather than to a port.
Copyright © OneAccess Networks – All rights reserved
12
Logical Local Voice Port
- For a 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
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13
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)>
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14
Dial-peer VoIP (1/2)
4 - Dial peer VoIP
Determine the SIP destination (User agent) for an
outgoing call:
Sig-protocol sip: Determines the signalling protocol
Gw-ip-address: Determines the remote end point of the voice
call (SIP transaction messages and RTP/RTCP packets
(SIP)
SIP end point may be another SIP gateway (UA)
SIP end point may be a SIP phone or Softphone (UA)
SIP end point may be a SIP proxy server (uses when different proxy
server have to be reached, voice-routing)
This parameter is not required if prox-dns-add exists in sip-gateway.
Fax, DTMF handling, ealy media capability.
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15
Dial-peer VoIP (2/2)
CLI(configure)# dial-peer voice voip 0
CLI(voip)# sig-protocol sip
CLI(voip)# gw-ip-address <ip-address[:port]> <hostname>
CLI(voip)# force-prack
CLI(voip)# sip-sdp-on-alert {receive-only|send-receive|send-only}
CLI(voip)# fax-relay {passthrough|t38|t38orpassthrough|t38nse}
for t38orpassthrough priority {t38|passthrough}
CLI(voip)# passthrough-mode {reinvite}
CLI(voip)# dtmf-relay {in-band (rfc2833)|sip-info}
CLI(voip)# no shutdown
force-prack # 100rel is added in supported header field. From
There, proxy may require PRACK to ack a 1xx message.
Sip-sdp-on-alert # About early media handling, requires to add SDP
Message body at outgoing 180 message, requires to process early media for
Incoming 180 and 183 messages.
Fax-relay # Re-invite including T38 or G711 in SDP message body
passthrough-mode reinvite # Require if voice call is establish for G729
and Fax-relay passthrough and/or modem-passthrough is validated
Dtmf-relay # Transmission of dtmf digit to voip. OneOs doesn’t
Support incoming SIP INFO message (not useful)
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16
VoIP Configuration
4 - Dial-peer VoIP
H323 (1/2)
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
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17
VoIP Configuration
4 - Dial-peer VoIP (2/2)
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
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18
Codec Profiles 1/2
5. VoIP coder profile
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: 10..90 depending on the coder
<cr>
one200(voip-coder)# codec 0 g729ab 30
one200(voip-coder)# codec 1 g711a 20
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19
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
exit
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20
SIP Gateway (1/2)
7. SIP gateway
CLI(configure)# sip-gateway
bandwidth-control
bridge-uri-host
bye-on-refer
bye-on-refer-accept
bye-timer
calling-number-checking
-
callsig-port
clip-privacy-uri
clip-unsubscribe-uri
connect-timer
device-host-name
discard-3XX
exit
gw-interface
-
gw-interface-bw-ctrl
invite-method-timeout
-
invite-response-timer
max-bandwidth
message-waiting-indication
no
outbound-proxy
-
Set SIP gateway - bandwidth control
Set bridge URI host characteristics
Send bye when refer is received
Send bye when Refer Accept is received
duration before bye message
Check origin number is registered to process a call
[default]
Set SIP gateway - SIP listening port
Define clip privacy predefined URI
Set predefined CLIP unsubscribe URI
duration of waiting 200 OK
Set sip gateway host name
Upon receiving a 3XX, the call is cleared
Exit from command node
Output Interface category for SIP GW - default
fastethernet 0
Set SIP gateway - gw interface bandwidth control
Invite methode Timeout. Timeout before receiving a final
response
duration of waiting first 1xx message
Maximum Bandwidth allowed
We should attempt to receive message waiting indication
no
Set outbound proxy for all messages (Noted also SBC)
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21
SIP Gateway (2/2)
CLI(configure)# sip-gateway
payload-64k-unrestricted
presentation-restricted
prox-dns-add
prox-ka
reg-dns-add
reg-failure-timer
reg-interval-timeout
reg-ka
registration-timeout
request-primitive-timer
shutdown
sig-dscp
sip-authentication
sip-called-number
sip-uri-escape
sip-username
softswitch-profile
subscription-duration
subscription-failed
trunking-mode
uri-contact
uri-from
user-agent
voicemail-dns-add
-
Payload for 64k unrestricted
presentation restricted (Anonymous)
Set Proxy characteristics
Proxy keep alive value
Set Registrar characteristics
Start when the UA SIP receives a 4xx, 5xx, 6xx response
Set interval registration timeout
Registrar keep alive value
Set registration timeout
Define Timeout for a Request SIP message
Shutdown SIP
DSCP field value for signalling packets
Set sip gateway username and password
get called number from sip invite message
escape # and * in sip URI
Set sip gateway ident
softswitch type { default | broadworks }
Subscription duration value
Subscription failed duration value
set trunking mode
Set type of URI in contact
Set type of URI in from
Do we include the user-agent header in the SIP INVITE
message { include | exclude }
- Set VoiceMail characteristics
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22
SIP Gateway
SIP gateway not registered, no proxy server
CLI(sipgw)# gw-interface fastethernet 0/0 intrusive
CLI(sipgw)# no shutdown
SIP gateway to proxy server, not registered
CLI(sipgw)# gw-interface fastethernet 0 intrusive
CLI(sipgw)# prox-dns-add 192.168.1.10
CLI(sipgw)# no shutdown
SIP gateway registered (proxy and registrar servers)
CLI(sipgw)#
CLI(sipgw)#
CLI(sipgw)#
CLI(sipgw)#
gw-interface fastethernet 0 intrusive
reg-dns-add 192.168.1.10
prox-dns-add 192.168.1.10
no shutdown
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23
Miscellaneous comments about OneOs (1)
Translation of SIP username to E164 number: This
feature is available for incoming call (dial-peer voip)
A SIP user (SIP phone) may calls remote user using SIP URI
as alice@atlanta.com (called party) instead of its E164 number
Voice-routing entry: sip-username converted to the prefix value
of this entry before to be sent to ISDN stack and populates the
IE 70.
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24
Miscellaneous comments about OneOs
OneOs Rules to build the Request URI about Invite
method:
1st: Utilisation of prox-dns-add value in the sip-gateway
2nd: If 1st doesn’t exist, utilisation of gw-ip-address value of
the dial-peer voice voip x of the corresponding voice route
entry.
OneOs Rules to build the To header field for Invite
method:
1st: utilisation of gw-ip-address value of the
dialpeer voice voip x of the corresponding voice route entry.
2nd: If 1st doesn’t exist, utilisation of prox-dns-add value in the
sip-gateway
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25
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)>
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26
VoIP Configuration
7 - H.323 Gateway (2/6)
one200(h323gw)> ?
alt-gatekeeper
altgk-list
altgk-mode
altgk-timeout
- Set H323 gateway - alternative gatekeeper
- Set H323 gateway - alternative gatekeeper list
- Set H323 gateway - alternate gatekeeper mode
- 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
email-id
- Set H323 email 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
...
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27
VoIP Configuration
7 - H.323 Gateway (3/6)
...
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
...
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28
VoIP Configuration
7 - H.323 Gateway (4/6)
...
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
packet)
- Set H323 gateway (DSCP field value for transmitted signalling
snmp-sysdescr-hw-ident - add hw ident to sysdescr
...
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29
VoIP Configuration
7 - H.323 Gateway (5/6)
...
start-h245-discarded - Set H323 gateway - facility start h245 is discarded
tcp-keepalive
- Set H323 gateway - tcp keepalive option (default)
unregister
- Unregister gateway
<cr>
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30
VoIP Configuration
7 - H.323 Gateway (6/6)
one200(config)>
one200(h323gw)>
one200(h323gw)>
one200(h323gw)>
one200(h323gw)>
one200(h323gw)>
h323-gateway
gw-interface fastethernet 0/0
gatekeeper id training address 192.168.1.1
h323-id GW1
max-bandwidth 600000
no shutdown
Copyright © OneAccess Networks – All rights reserved
31
Business-Class Router Solutions – All Telecommunication Services in One Access
Voice Toubleshooting & Statistics
www.oneaccess-net.com
contact@oneaccess-net.com
v0.11 2007
Copyright © OneAccess Networks – All rights reserved
VoIP Statistics
BRI statistics
CLI# show voice voice-port bri index 0
voice port
protocol descriptor
current state
config state
layer 1 status
attached vmoabri dial peer
number of voice communication
bri Tx frames on D channel
bri Rx frames on D channel
5/0
BRI_NT
activated
up
activated
0
0
40
41
Outgoing calls
Outgoing calls failures
Physical Interface down
Cause Class 0 (normal event)
Cause Class 1 (normal event)
Normal Cause (16)
: 2
User busy (17)
: 3
No answer (18)
: 0
Cause Class 2 (unavailable ressources)
Cause Class 3 (unavailable service)
Cause Class 4 (service not provided)
Cause Class 5 (invalid message)
Cause Class 6 (protocol error)
Cause Class 7 (interworking)
:
:
:
:
:
102
5
0
0
5
:
:
:
:
:
:
0
0
0
0
0
0
Incoming calls
Incoming calls failures
Remote failure
Unknown number
DSP unavailable
Not specified
:
:
:
:
:
:
54
7
0
5
0
2
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33
VoIP Statistics
PRI statistics
CLI# show voice voice-port pri index 0
voice port
physical type
protocol descriptor
current state
config state
layer 1 status
number of voice communications
pri AIS occurence
pri RDI occurence
5/0
E1
E1_PRI
activated
up
deactivated
0
0
0
Outgoing calls
Outgoing calls failures
Physical Interface down
Cause Class 0 (normal event)
Cause Class 1 (normal event)
Normal Cause (16)
: 0
User busy (17)
: 3
No answer (18)
: 0
Cause Class 2 (unavailable ressources)
Cause Class 3 (unavailable service)
Cause Class 4 (service not provided)
Cause Class 5 (invalid message)
Cause Class 6 (protocol error)
Cause Class 7 (interworking)
:
:
:
:
:
67
3
0
0
3
:
:
:
:
:
:
0
0
0
0
0
0
Incoming calls
Incoming calls failures
Remote failure
Unknown number
DSP unavailable
Not specified
:
:
:
:
:
:
23
2
2
0
0
0
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34
VoIP Statistics
FXS statistics
CLI# show voice voice-port fxs index 0
voice port
current state
config state
attached vmoa fxs dial peer
voice communication
Outgoing calls
Outgoing calls failures
User busy
No answer
Incoming calls
Incoming calls failures
Remote failure
Unknown number
DSP unavailable
Not specified
5/0
on hook
up
0
no
:
:
:
:
:
:
:
:
:
:
32
3
2
1
6
0
0
0
0
0
Copyright © OneAccess Networks – All rights reserved
35
VoIP Statistics
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
Copyright © OneAccess Networks – All rights reserved
36
VoIP Statistics
Dial-peer VoIP Statistics (2): Outgoing Calls
Dial Peer
Current Calls
1
0
Outgoing 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)
Cause Class 3 (unavailable service)
Cause Class 4 (service not provided)
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
Copyright © OneAccess Networks – All rights reserved
4
0
0
0
0
0
0
0
0
0
0
0
0
0
0
0
0
0
0
0
0
0
0
0
0
37
VoIP Statistics
Dial-peer VoIP Statistics (3): Incoming Calls
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
Copyright © OneAccess Networks – All rights reserved
3
1
0
0
0
1
0
0
0
0
0
0
0
0
0
0
38
VoIP Statistics
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
3
0
0
2
1
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
Number of lost packets
Copyright © OneAccess Networks – All rights reserved
1237
1234
101484
101098
3
0
0
0
0
0
0
0
0
0
0
0
0
0
0
0
0
39
VoIP Statistics
Events
vxTarget>event
filter
manager
no
recover
vxTarget>event
add
remove
vxTarget>event
ALL
GEN
VOATM
VOIP
vxTarget>event
<subfam>
<fam2>
vxTarget>event
-
Add/remove events filters
Add a SNMP manager
No
Recover events from memory
filter
- Add an event filter
- Remove a events filter from the table
filter add vox
- All families from vox group
- GEN
- VOATM
- VOIP
filter add vox voip
- <ALL | ControlPlan | UserPlan>
- <GEN | VOATM>
filter add vox voip all show
Copyright © OneAccess Networks – All rights reserved
40
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 :192.168.1.110 rtp:16384 /Dest ip :192.168.1.111 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 / 395307
Number of Excessive Jitter events : 3
Copyright © OneAccess Networks – All rights reserved
41
VoIP Statistics
RTP sessions history
Gives complete statistics about the 200 last RTP sessions
CLI> show voice rtpcall full any ind 2
2 - 01/04/01 00h47m24s RTP 192.168.1.1:16384 – 192.168.1.111:16386
Play time (voice) : 00h00m46s
Tx Coder : G729 / 20 ms ; Rx Coder : G729
VAD enabled local / remote : no / no
ERL
ACOM
: 15 dB
: 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
Copyright © OneAccess Networks – All rights reserved
42
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
Nb of occur.
2
0
0
Interarrival max jitter : 9 ms
Jitter received (DSP) :
Max delay : 93 ms
Delays (ms)
>50 >100 >150
Nb of occur.
2
0
0
Interarrival max jitter : 9 ms
>200
0
>300
0
>200
0
>300
0
>200
0
>300
0
Frames with a delay >50 ms :
2|
1| *
*
---------------------------------------0 30"
1'
2'
4'
8' 12' >16'
Jitter transmitted (uplink) :
Max delay : 6 ms
Delays (ms)
>50 >100 >150
Nb of occur.
0
0
0
Interarrival max jitter : 1 ms
(RTCP reported) : 2 ms
Copyright © OneAccess Networks – All rights reserved
43
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 0198723000
calling 3000
bearer data
duration 180
timeout 10
exit
One200> start identifier 1
Copyright © OneAccess Networks – All rights reserved
44
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
00:07:21.271
00:07:21.271
00:07:21.271
00:07:21.271
00:07:21.271
00:07:21.272
00:07:21.343
00:07:21.343
00:07:21.343
00:07:21.343
00:07:21.343
00:07:21.343
line:5/0 L1 frame sent.
line:5/0 L2 tx UI P/F=0 NR=4 NS=2 C/R=1.
hex: 02 ff 03
line:5/0 L3 tx SETUP callref:8.
hex1: 08 01 08 05 04 03 80 90 a3 18 01 89 70 04 a1 31
hex2: 31 31 a1
line:5/0 L1 frame received.
line:5/0 L2 rx SABME P/F=1 C/R=0.
hex: 00 01 7f
line:5/0 L1 frame sent.
line:5/0 L2 tx UA P/F=1 NR=4 NS=2 C/R=0.
hex: 00 01 73
Copyright © OneAccess Networks – All rights reserved
45
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 0141877422
<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..2000 <0 means no overlap used>
<cr>
One_training>auto-call 0141877422 2408882005
17:50:17.677 Info vox factory test 1 call-id: 4, ident: auto-call, CALL IN PROGRESS Calling=2408882
005 Called=0141877422.
one100_interopBW>17:50:17.678 Info vox voip controlplan 3 Incoming call on local pots: 0, calling:
2408882005, called: 0141877422, call-id: 4.
17:50:17.710 Info vox voip controlplan 3 Outgoing call on voip id: 0, calling: 2408882005, called:
0141877422, call-id: 4.
17:50:27.660 Info vox factory test 1 call-id: 4, ident: auto-call, CALL FAILED cause=no codec.
17:50:27.661 Info vox factory test 1 call-id: 4, ident: auto-call, CALL FAILED on pots cause=[Norma
l call clearing].
Copyright © OneAccess Networks – All rights reserved
46
Auto call to ISDN
For debug, a ‘SETUP’ can be sent on a ISDN local
port. One_training>isdn test call 85841 ( data call/unrestricted )
02:27:34.914 line:5/0 L1 event received PH_AR State:F3.
02:27:34.923 line:5/0 L1 event received EV_LOST_FRAMING State:F4.
02:27:34.923 line:5/0 L1 event received EV_INFO_2 State:F5.
02:27:34.925 line:5/0 L1 event received EV_INFO_4_8(PH_AI) State:F6.
02:27:34.925 line:5/0 L1 event received MPH_AI State:F7.
02:27:34.925 line:5/0 L1 frame sent.
02:27:34.925 line:5/0 L2 tx SABME P/F=1 C/R=0.
02:27:34.925
hex: 00 83 7f
02:27:34.938 line:5/0 L1 frame received.
02:27:34.938 line:5/0 L2 rx UA P/F=1 C/R=0.
02:27:34.939
hex: 00 83 73
02:27:34.939 line:5/0 L1 frame sent.
02:27:34.940 line:5/0 L2 tx INFO P=0 NR=0 NS=0 C/R=0.
02:27:34.940
hex: 00 83 00 00
02:27:34.940 line:5/0 L3 tx SETUP callref:4.
02:27:34.940
hex1: 08 01 04 05 04 02 88 90 18 01 83 70 06 81 38 35
02:27:34.940
hex2: 38 34 31 a1
02:27:34.940
Called Number : 85841
Copyright © OneAccess Networks – All rights reserved
47
