www.oneaccess-net.com [email protected] business-class router solutions – all...

www.oneaccess-net.com [email protected]

Business-Class Router Solutions – All Telecommunication Services in One Access

Copyright © OneAccess Networks – All rights reserved

OneOs Voice Configuration Dial-Peer Concept

4.11 H3234.15 SIP4.21 MGCP

v0.11 2007

Copyright © OneAccess Networks – All rights reserved 2

ONE

VoIP Configuration Diagram (H.323 / SIP)

Analogue Physical

Voice-port

Digital Physical

Voice-port

ISD

N P

BX

Interface BRI/PRI

Dial-Peer VoiceVoIP

4

Voice Routing6

VoIP Coder Profile

5

H.323 or SIP Gateway

7BRIPRI

CLASSICAL TELEPHONE NETWORK

IP NETWORK

Analogue Physical

Voice-port

1

Digital Physical

Voice-port

1

POTS Group 1

Dial-Peer VoicePOTSDial-Peer VoicePOTS

3

POTS Group 2


3

Interface BRI/PRI

2

SIP Server 8

Dial-Peer VoiceVoIP

SIP Phones

POTS

ISDN Phone

BRI


ONE

VoIP Configuration Diagram (MGCP)

Analogue Physical

Voice-port

Dial-Peer VoiceVoIP

4

VoIP Coder Profile

5

MGCP Gateway

7

CLASSICAL TELEPHONE NETWORK

IP NETWORK

Analogue Physical

Voice-port

1

POTS Group 2


3


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)

VoIP Configuration


Voice Port Configuration

one200>conf t

one200(configure)>voice-port 5/0

one200(voice-port)>exit

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



. . .





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...

VoIP Configuration



... 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>

VoIP Configuration


2 - Interfaces (1/4) FXS: No configuration BRI

PRI

one200(configure)>interface bri 5/0one200(config-if)> ? exit - exit isdn - Set isdn level no - no shutdown - shutdown

one200(configure)>interface pri 5/0one200(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

VoIP Configuration


2 - Interfaces (2/4) BRI

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)>

VoIP Configuration


2 - Interfaces (3/4) BRI: Example with ISDN phone

BRI: Example with PBX

one200(configure)>interface bri 5/0one200(conf-if)> isdnone200(isdn)> protocol-emulation isdn-ntone200(isdn)> exitone200(conf-if)> no shutdownone200(conf-if)> executeone200(conf-if)> exit

one200(configure)> interface bri 5/0one200(conf-if)> isdnone200(isdn)> tei-negotiation staticone200(isdn)> protocol-emulation isdn-ntone200(isdn)> exitone200(conf-if)> no shutdownone200(conf-if)> executeone200(conf-if)> exit

VoIP Configuration


2 - Interfaces (4/4) PRI

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

VoIP Configuration


CLI(configure)# dial-peer voice pots <id>CLI(pots)# pots-group <port> CLI(pots)# port 5/<port>CLI(pots)# no shutdownCLI(pots)# exit

Logical Local Voice Port

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.

3 - Internal Local Voice Port (POTS)


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 shutdownCLI(pots)# exit


- 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.


3 – Local Voice Port optional parameters

one200(configure)> dial-peer voice pots 0one200(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)>



4 - Dial peer VoIP (SIP)

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 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.

Dial-peer VoIP (1/2)


CLI(configure)# dial-peer voice voip 0CLI(voip)# sig-protocol sipCLI(voip)# gw-ip-address <ip-address[:port]> <hostname>CLI(voip)# force-prackCLI(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

Dial-peer VoIP (2/2)

force-prack # 100rel is added in supported header field. FromThere, 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 bodypassthrough-mode reinvite # Require if voice call is establish for G729 and Fax-relay passthrough and/or modem-passthrough is validatedDtmf-relay # Transmission of dtmf digit to voip. OneOs doesn’tSupport incoming SIP INFO message (not useful)


4 - Dial-peer VoIP H323 (1/2)

one200(configure)> dial-peer voice voip 0one200(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

VoIP Configuration


one200(configure)>dial-peer voice voip 0one200(voip)>fast-connectone200(voip)>gatekeeper mandatoryone200(voip)>voip-coder-profile 0one200(voip)>no shutdownone200(voip)>exit

one200(configure)>dial-peer voice voip 0one200(voip)>fast-connectone200(voip)>gatekeeper mandatoryone200(voip)>voip-coder-profile 0one200(voip)>no shutdownone200(voip)>exit

4 - Dial-peer VoIP (2/2)

VoIP Configuration


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 0CLI(voip-coder)# codec ? <pref-index> - Codec preference index: 0..8CLI(voip-coder)# codec 0 ? <coder> - Coder type: g729ab | g711a | g711uCLI(voip-coder)# codec 0 g729ab ? <timestamp> - Timestamp value: 10..90 depending on the coder <cr>one200(voip-coder)# codec 0 g729ab 30one200(voip-coder)# codec 1 g711a 20

Codec Profiles 1/2


one200>show running

...

voip-coder-profile 0 codec 0 g729ab 30 codec 1 g711a 20 exitvoip-coder-profile 1 codec 0 g711a 20 exit

Codec Profiles 2/2


SIP Gateway (1/2)

CLI(configure)# sip-gateway bandwidth-control - Set SIP gateway - bandwidth control bridge-uri-host - Set bridge URI host characteristics bye-on-refer - Send bye when refer is received bye-on-refer-accept - Send bye when Refer Accept is received bye-timer - duration before bye message calling-number-checking - Check origin number is registered to process a call [default] callsig-port - Set SIP gateway - SIP listening port clip-privacy-uri - Define clip privacy predefined URI clip-unsubscribe-uri - Set predefined CLIP unsubscribe URI connect-timer - duration of waiting 200 OK device-host-name - Set sip gateway host name discard-3XX - Upon receiving a 3XX, the call is cleared exit - Exit from command node gw-interface - Output Interface category for SIP GW - default fastethernet 0 gw-interface-bw-ctrl - Set SIP gateway - gw interface bandwidth control invite-method-timeout - Invite methode Timeout. Timeout before receiving a final response invite-response-timer - duration of waiting first 1xx message max-bandwidth - Maximum Bandwidth allowed message-waiting-indication - We should attempt to receive message waiting indication no - no outbound-proxy - Set outbound proxy for all messages (Noted also SBC)

7. SIP gateway


SIP Gateway (2/2)

CLI(configure)# sip-gateway payload-64k-unrestricted - Payload for 64k unrestricted presentation-restricted - presentation restricted (Anonymous) prox-dns-add - Set Proxy characteristics prox-ka - Proxy keep alive value reg-dns-add - Set Registrar characteristics reg-failure-timer - Start when the UA SIP receives a 4xx, 5xx, 6xx response reg-interval-timeout - Set interval registration timeout reg-ka - Registrar keep alive value registration-timeout - Set registration timeout request-primitive-timer - Define Timeout for a Request SIP message shutdown - Shutdown SIP sig-dscp - DSCP field value for signalling packets sip-authentication - Set sip gateway username and password sip-called-number - get called number from sip invite message sip-uri-escape - escape # and * in sip URI sip-username - Set sip gateway ident softswitch-profile - softswitch type { default | broadworks } subscription-duration - Subscription duration value subscription-failed - Subscription failed duration value trunking-mode - set trunking mode uri-contact - Set type of URI in contact uri-from - Set type of URI in from user-agent - Do we include the user-agent header in the SIP INVITE message { include | exclude } voicemail-dns-add - Set VoiceMail characteristics


SIP gateway not registered, no proxy server

SIP gateway to proxy server, not registered

SIP gateway registered (proxy and registrar servers) CLI(sipgw)# gw-interface fastethernet 0/0 intrusiveCLI(sipgw)# no shutdown

CLI(sipgw)# gw-interface fastethernet 0 intrusiveCLI(sipgw)# reg-dns-add 192.168.1.10CLI(sipgw)# prox-dns-add 192.168.1.10CLI(sipgw)# no shutdown

SIP Gateway

CLI(sipgw)# gw-interface fastethernet 0 intrusiveCLI(sipgw)# prox-dns-add 192.168.1.10CLI(sipgw)# no shutdown


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 [email protected] (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.

Miscellaneous comments about OneOs (1)


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 dial-

peer 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


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)>

VoIP Configuration


7 - H.323 Gateway (2/6)

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

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

...

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

...

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 - Set H323 gateway (DSCP field value for transmitted signalling packet)

snmp-sysdescr-hw-ident - add hw ident to sysdescr

...

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>

VoIP Configuration


one200(config)> h323-gateway

one200(h323gw)> gw-interface fastethernet 0/0

one200(h323gw)> gatekeeper id training address 192.168.1.1

one200(h323gw)> h323-id GW1

one200(h323gw)> max-bandwidth 600000

one200(h323gw)> no shutdown

7 - H.323 Gateway (6/6)

VoIP Configuration

www.oneaccess-net.com [email protected]

Business-Class Router Solutions – All Telecommunication Services in One Access

Copyright © OneAccess Networks – All rights reserved

Voice Toubleshooting & Statistics

v0.11 2007


CLI# show voice voice-port bri index 0 voice port 5/0 protocol descriptor BRI_NT current state activated config state up layer 1 status activated attached vmoabri dial peer 0 number of voice communication 0 bri Tx frames on D channel 40 bri Rx frames on D channel 41

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

CLI# show voice voice-port bri index 0 voice port 5/0 protocol descriptor BRI_NT current state activated config state up layer 1 status activated attached vmoabri dial peer 0 number of voice communication 0 bri Tx frames on D channel 40 bri Rx frames on D channel 41



BRI statistics

VoIP Statistics


CLI# show voice voice-port pri index 0 voice port 5/0 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 0 pri RDI occurence 0



CLI# show voice voice-port pri index 0 voice port 5/0 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 0 pri RDI occurence 0



PRI statistics

VoIP Statistics


CLI# show voice voice-port fxs index 0 voice port 5/0 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

CLI# show voice voice-port fxs index 0 voice port 5/0 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

FXS statistics

VoIP Statistics


one200> show voice dial-peer voice voip type index <port id> [reset]orone200> 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

one200> show voice dial-peer voice voip type index <port id> [reset]orone200> 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

Dial-peer VoIP Statistics (1)

VoIP Statistics


Dial-peer VoIP Statistics (2): Outgoing Calls Dial Peer 1 Current Calls 0 Outgoing Calls Outgoing Calls 4 Outgoing calls failures 0 RAS Call Failures 0 Gatekeeper Unavailable 0 Admission Rejects 0 H225/Q931 Call failures 0 Cause Class 0 (normal event) 0 Cause Class 1 (normal event) 0 Normal Cause (16) 0 User busy (17) 0 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 H245 Call failures 0 Incompatible capabilities 0 Protocol errors 0 Internal call failures 0 DSP unavailable 0 Max-bandwidth exceeded 0 Max-connection exceeded 0 Not specified 0

VoIP Statistics


Dial-peer VoIP Statistics (3): Incoming Calls

Incoming calls 3 Incoming calls failures 1 RAS Call failures 0 Gatekeeper Unavailable 0 Admission Rejects 0 Local Port Call failures 1 H245 Call failures 0 Incompatible capabilities 0 Protocol errors 0 Internal call failures 0 DSP unavailable 0 Unknown number 0 Channel / port unavailable 0 Max-bandwidth exceeded 0 Max-connection exceeded 0 Not specified 0

VoIP Statistics


Dial-peer VoIP Statistics (4): Voice and Fax

RTP statistics Number of transmitted packets 1237 Number of received packets 1234 Number of transmitted bytes 101484 Number of received bytes 101098 Number of excessive jitter events 3 Number of lost packets 0 Number of invalid packets 0 Number of calls with frame error rate total <0.01<0.1<0.5<1<5>=5 3 0 0 2 1 0 0 Modem passthrough Number of switching to modem mode 0 T38 FAX Calls Number of outgoing fax 0 Number of incoming fax 0 Number of failures 0 Request Mode failure 0 Pre-message procedure failure 0 Page failure 0 Number of transmitted packets 0 Number of received packets 0 Number of transmitted bytes 0 Number of received bytes 0 Number of lost packets 0

VoIP Statistics


vxTarget>event filter - Add/remove events filters manager - Add a SNMP manager no - No recover - Recover events from memoryvxTarget>event filter add - Add an event filter remove - Remove a events filter from the tablevxTarget>event filter add vox ALL - All families from vox group GEN - GEN VOATM - VOATM VOIP - VOIPvxTarget>event filter add vox voip <subfam> - <ALL | ControlPlan | UserPlan> <fam2> - <GEN | VOATM>vxTarget>event filter add vox voip all show

vxTarget>event filter - Add/remove events filters manager - Add a SNMP manager no - No recover - Recover events from memoryvxTarget>event filter add - Add an event filter remove - Remove a events filter from the tablevxTarget>event filter add vox ALL - All families from vox group GEN - GEN VOATM - VOATM VOIP - VOIPvxTarget>event filter add vox voip <subfam> - <ALL | ControlPlan | UserPlan> <fam2> - <GEN | VOATM>vxTarget>event filter add vox voip all show

Events

VoIP Statistics


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

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

Voice call history, active calls Gives statistics on the current voice calls and the last 100

calls

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 : 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

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 : 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

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. 2 0 0 0 0 Interarrival max jitter : 9 ms Jitter received (DSP) : Max delay : 93 ms Delays (ms) >50 >100 >150 >200 >300 Nb of occur. 2 0 0 0 0 Interarrival max jitter : 9 ms 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 >200 >300 Nb of occur. 0 0 0 0 0 Interarrival max jitter : 1 ms (RTCP reported) : 2 ms

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. 2 0 0 0 0 Interarrival max jitter : 9 ms Jitter received (DSP) : Max delay : 93 ms Delays (ms) >50 >100 >150 >200 >300 Nb of occur. 2 0 0 0 0 Interarrival max jitter : 9 ms 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 >200 >300 Nb of occur. 0 0 0 0 0 Interarrival max jitter : 1 ms (RTCP reported) : 2 ms

VoIP Statistics


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 0exit

voip-call 1 pots 4 called 0198723000 calling 3000 bearer data duration 180 timeout 10exit

One200> start identifier 1

dial-peer voice pots 4 service bert2047 both 3 pots-group 0exit

voip-call 1 pots 4 called 0198723000 calling 3000 bearer data duration 180 timeout 10exit

One200> start identifier 1


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 tCLI>logging buffered debugCLI>exitCLI>debug isdn all layer 1to3

00:07:21.271 line:5/0 L1 frame sent.00:07:21.271 line:5/0 L2 tx UI P/F=0 NR=4 NS=2 C/R=1.00:07:21.271 hex: 02 ff 0300:07:21.271 line:5/0 L3 tx SETUP callref:8.00:07:21.271 hex1: 08 01 08 05 04 03 80 90 a3 18 01 89 70 04 a1 3100:07:21.272 hex2: 31 31 a100:07:21.343 line:5/0 L1 frame received.00:07:21.343 line:5/0 L2 rx SABME P/F=1 C/R=0.00:07:21.343 hex: 00 01 7f00:07:21.343 line:5/0 L1 frame sent.00:07:21.343 line:5/0 L2 tx UA P/F=1 NR=4 NS=2 C/R=0.00:07:21.343 hex: 00 01 73


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 240888200517:50:17.677 Info vox factory test 1 call-id: 4, ident: auto-call, CALL IN PROGRESS Calling=2408882005 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=[Normal call clearing].


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 7f02:27:34.938 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 7302:27:34.939 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 0002:27:34.940 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 3502:27:34.940 hex2: 38 34 31 a1

02:27:34.940 Called Number : 85841

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