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Page 1: Www.oneaccess-net.com contact@oneaccess-net.com Copyright © OneAccess Networks – All rights reserved United Networking 20 August, 2014 VoIP H323 VoIP H323

www.oneaccess-net.com [email protected]

Copyright © OneAccess Networks – All rights reserved

United Networking

11 April 2023

VoIP H323

VoIP H323 basics , Fax management / H323, statistics

v1.0 2008

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U n i t e d N e t w o r k i n gU n i t e d N e t w o r k i n g

VoIP

VoIP Typical Architecture

PSTN SS7/ISUP

Media Gateway

SoftswitchGatekeeper

ISDN T0/T2

ISDN S0 or FXS

H.323 Voice

Gateway ONE 400

H.323 Voice

Gateway ONE 200

IP

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U n i t e d N e t w o r k i n gU n i t e d N e t w o r k i n g

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

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U n i t e d N e t w o r k i n gU n i t e d N e t w o r k i n g

VoIP Basics

H.323 Protocol Architecture

Q.931

Terminal Control and ManagementData AppAV App

Reliable Transport (TCP)Unreliable Transport (UDP)

G.7XX H.26XH.245

RTP

T.124

RTCP

H.225.0 Call Signaling

Terminal to Gatekeeper Signaling

(RAS)

T.123

T.125

TPKT

Physical Layer

Link Layer

Network Layer (IP)

Conference ManagerConference Manager

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U n i t e d N e t w o r k i n gU n i t e d N e t w o r k i n g

VoIP Basics

(UDP)

SETUP

CONNECT

(H245 Address)

Q.931

(TCP)

H.245 Messages

Open Logical Channels

(RTCP address)

(RTCP address)

(RTCP & RTP addresses)

H.245(TCP)

Media

H.323 Terminal

H.323 Terminal

Gatekeeper Registration

Admission

SETUP

CONNECT

RegistrationRAS

(UDP)

(RTCP & RTP addresses)

RTP streamRTP stream

RTCP stream

H.323 Protocol Overview

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U n i t e d N e t w o r k i n gU n i t e d N e t w o r k i n g

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

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U n i t e d N e t w o r k i n gU n i t e d N e t w o r k i n g

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)

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VoIP: Dialing Modes

ISDN terminal Remote EndGatekeeperONE200/400

SETUPSETUP ACK

INFO Digit 3INFO Digit 0INFO Digit 2

CALL PROCEEDING

CALL PROCEEDINGSETUP ACK

ALERTCONNECT

ALERTALERTCONNECTCONNECT

ARQ N° 302ACF

SETUP N° 302 SETUP N° 302

Dialling: In-bloc

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VoIP: Dialing modes

ISDN terminal Remote EndGatekeeperONE200/400

SETUPSETUP ACK

INFO Digit 3INFO Digit 0INFO Digit 2

CALL PROCEEDING

SETUP ACK

ALERTCONNECT

ALERTALERTCONNECTCONNECT

ARQ N° 302ACF

SETUP N° 302 SETUP N° 302

CALL PROCEED.CALL PROCEED.INFO Digit 9 INFO Digit 9 INFO Digit 9INFO Digit 7 INFO Digit 7 INFO Digit 7INFO Digit 1 INFO Digit 1 INFO Digit 1

Dialing: Overlap

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U n i t e d N e t w o r k i n gU n i t e d N e t w o r k i n g

VoIP: FAX/Modem processing

Fax / Modem processing Three types of devices are supported:

G3 FAX (up to V17 – 14400 bps)Super G3 FAX (V34): processed as a ModemModem (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

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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)

Echo canceller automatically disabled in case of Modem NLP automatically disabled in case of G3 Fax

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

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U n i t e d N e t w o r k i n gU n i t e d N e t w o r k i n g

VoIP: DTMF Processing

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

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Configuration diagram (1)

Voice-port

Physical Port

Interface BRI/PRI

Dial-peerVoice POTS

Voice Routing

H.323 Gateway

Dial-peerVoice Voip

VoIP coder profile

VoIP Configuration

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U n i t e d N e t w o r k i n gU n i t e d N e t w o r k i n g

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

VoIP Configuration

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Voice Port Configuration

1 - Physical voice ports (1/3)

BRI (2 or 4 or 8 ports) or FXS (4 or 8 ports)

PRIone200>conf t

one200(configure)>voice-port 5/0

one200(voice-port)>exit

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

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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...

VoIP Configuration

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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>

VoIP Configuration

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

PRI

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

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

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U n i t e d N e t w o r k i n gU n i t e d N e t w o r k i n g

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

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U n i t e d N e t w o r k i n gU n i t e d N e t w o r k i n g

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

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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

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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.

3 - Internal Local Voice Port (POTS)

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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

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.

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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)>

Logical Local Voice Port

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4 - Dial-peer VoIP (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

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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

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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 20one200(voip-coder)# exit

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)

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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

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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)>

6 - VoIP Routing table (1/10)

VoIP Configuration

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one200(voice-route)>route 1one200(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>

6 - VoIP Routing table (2/10)

VoIP Configuration

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one200(configure)>voice-routingone200(voice-route)>route 1one200(conf-voice-route)>dial-peer pots-group 0 aliasone200(conf-voice-route)>prefix-type outgoing called lastone200(conf-voice-route)>prefix 110 length 3one200(conf-voice-route)>exitone200(voice-route)>route 2one200(conf-voice-route)>dial-peer voip 0one200(conf-voice-route)>prefix-type outgoing called lastone200(conf-voice-route)>prefix 11X length 3one200(conf-voice-route)>exitone200(voice-route)>exitone200(configure)>exit

one200(voice-route)>display-routes 1 - 110 all length 3 / pots 0 / outgoing - called - last 2 – 11X all length 3 / voip 0 / outgoing - called - lastone200(voice-route)>

6 - VoIP Routing table (3/10)

VoIP Configuration

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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 prefix-type outgoing called last exitexit...

6 - VoIP Routing table (4/10)

VoIP Configuration

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route 1

dial-peer pots-group 0 prefix 120A. length 4 suppress-prefix 4 called wildcard A 0123 exit

Pots-group 0

VoIP 0

12011234

1234

6 - VoIP Routing table (5/10)

VoIP Configuration

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U n i t e d N e t w o r k i n gU n i t e d N e t w o r k i n g

route 14 dial-peer voip 1 prefix 986902739 length 9 prefix-type incoming called exit route 15 dial-peer voip 2 prefix . length 0 prefix-type incoming called exit

Pots-group 0

VoIP 1986902739

VoIP 2986911111Pots-group 4

Incoming Table

6 - VoIP Routing table (6/10)

VoIP Configuration

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ing : 019869027391111 Incoming Table

Outgoing Table

ing : 986902739 ed : 120012345678

ing : 019869027352222

Pots-group 0

ing : 986902735ed : 12345678

ed : 12345678

ed : 12345678

6 - VoIP Routing table (7/10)

VoIP Configuration

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route 2 dial-peer pots-group 0 translate .98690273A. 98690273A calling prefix . length 0 prefix-type incoming calling next exit route 3 dial-peer pots-group 0 alias insert-prefix 1200 called prefix 986902739 length 9 prefix-type incoming calling exit

6 - VoIP Routing table (8/10)

VoIP Configuration

alias added to register the number 1200 in the RAS

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ing : 019869027391111 Incoming Table

Outgoing Table

ing : 986902739 ed : 120012345678

ing : 019869027352222

Pots-group 0

ing : 986902735ed : 12345678

ed : 12345678

ed : 12345678

Voip 0

Voip 1

ing : 986902739 ed : 120012345678

ing : 986902735ed : 12345678

6 - VoIP Routing table (9/10)

VoIP Configuration

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route 4 dial-peer voip 1 prefix 1200. length 4 prefix-type outgoing called last exit route 5 dial-peer voip 0 prefix . length 0 prefix-type outgoing called last exit

6 - VoIP Routing table (10/10)

VoIP Configuration

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Voice routing table testing

One_Training(voice-route)>display-route 1 - . all length 10 / pots 0 / incoming - called - last / +prefix-calling 0 10 - . all timer / voip 0 / outgoing - called - last 20 - . all length 9 / pots 0 / outgoing - called - last / -prefix-called 1One_Training(voice-route)>

One_Training> configure terminalOne_Training(configure)>voice-routingOne_Training(voice-route)>test-route 0141877422 260402020 from-voip 0--> test_route : Calling=0141877422 Called=260402020Route match : Calling=0141877422 Called=60402020Incoming Routes = Outgoing Routes = 20,Incoming call from voip id:0 Send towards Isdn on local port: 5/0One_Training(voice-route)>One_Training(voice-route)>One_Training(voice-route)>test-route 260402020 0141877422 from-pots 0--> test_route : Calling=260402020 Called=0141877422Route match : Calling=0260402020 Called=0141877422Incoming Routes = 1, Outgoing Routes = 10,Outgoing call from local port: 5/0 -> Send towards H323 on voip id:0One_Training(voice-route)>

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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-gatewayone200(h323gw)>

VoIP Configuration

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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

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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

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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

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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

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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

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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

BRI statistics

VoIP Statistics

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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

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

PRI statistics

VoIP Statistics

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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

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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

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Dial-peer VoIP Statistics (2): Outgoing Calls

Dial Peer 1Current Calls 0Outgoing 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

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Dial-peer VoIP Statistics (3): Incoming Calls

Incoming calls 3Incoming calls failures 1RAS Call failures 0 Gatekeeper Unavailable 0 Admission Rejects 0Local Port Call failures 1H245 Call failures 0 Incompatible capabilities 0 Protocol errors 0Internal 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

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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

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H.323 Gateway Statistics

CLI# show voice h323-gateway

Registration state : registered Gatekeeper identifier : training Gatekeeper address : 192.168.1.1 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

VoIP Statistics

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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

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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

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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

VoIP Statistics

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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 

VoIP Statistics

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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

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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

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VoIP debug

VoIP H323 signaling messages can be displayed for debug.

One_training>debug h323One_training>One_training>00:06:10.516 H.323 rx TCP-H.225 Msg=Setup IP=86.43.21.7:1882 callref=9203.00:06:10.542 Info vox voip controlplan 3 Incoming call on voip id: 0, calling: 90851745010, called: 1000610000, call-id: 1.00:06:10.543 Info vox voip controlplan 3 Outgoing call on local port: 5/0, calling: 90851745010, called: 7383, call-id: 1.00:06:10.717 Info vox voip controlplan 3 Alert received, call-id: 1.s00:06:10.731 Info vox voip userplan 3 VoIP RTP thransmission started, coder: G729, call- id: 1.00:06:10.709 H.323 tx TCP-H.225 Msg=CallProceeding IP=86.43.21.7:1882 callref=41971.00:06:10.717 H.323 tx TCP-H.225 Msg=Alerting IP=86.43.21.7:1882 callref=41971.00:06:10.766 H.323 rx TCP-H.225 Msg=Facility IP=86.43.21.7:1882 callref=9203.00:06:10.783 H.323 tx TCP-H.225 Msg=Facility IP=86.43.21.7:1882 callref=41971.00:06:10.784 H.323 rx TCP-H.225 Msg=Facility IP=86.43.21.7:1882 callref=9203.00:06:10.802 H.323 tx TCP-H.225 Msg=Facility IP=86.43.21.7:1882 callref=41971.00:06:10.808 H.323 tx TCP-H.225 Msg=Facility IP=86.43.21.7:1882 callref=41971.00:06:10.816 H.323 rx TCP-H.225 Msg=Facility IP=86.43.21.7:1882 callref=9203.00:06:17.954 Info vox voip controlplan 3 Call connected, call-id: 1.00:06:17.962 H.323 tx TCP-H.225 Msg=Connect IP=86.43.21.7:1882 callref=41971.00:06:18.010 Info vox voip userplan 3 VoIP RTP reception started, coder: G729, call-id: 1.

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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].

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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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Fax relay T38 debug

Fax T38 processing and traces: display of the T30 messages

One_training>trace filter add vox up ifp 2 show01:42:38.100 Info vox voip controlplan 3 Outgoing call on voip id: 0, calling: 110, called: 111, call-id: 29.01:42:40.397 Info vox voip controlplan 3 Alert in band received, call-id: 29.01:42:40.917 Info vox voip userplan 3 VoIP RTP transmission started, coder: G729, call-id: 29.01:42:40.997 Info vox voip userplan 3 VoIP RTP reception started, coder: G729, call-id: 29.01:42:52.444 Info vox voip controlplan 3 Call connected, call-id: 29.01:42:57.537 Info vox voip userplan 3 VoIP RTP transmission stopped, coder: G729, call-id: 29.01:42:57.537 Info vox voip userplan 3 VoIP RTP reception stopped, coder: G729, call-id: 29.01:42:57.665 Info vox voip userplan 3 Fax T38 starting call-id: 29 .01:43:04.825 VOX.up.ifp.2 ------ NSF / CSI / DIS ------>01:43:07.717 VOX.up.ifp.2 <-------- TSI / DCS ----------01:43:07.837 VOX.up.ifp.2 <------- TCF (v29_9600) --------01:43:11.814 Info vox voip userplan 1 T38 Pre-message procedure OK, call-id: 29.01:43:11.814 VOX.up.ifp.2 ------------ CFR ------------>01:43:12.437 Info vox voip userplan 1 T38 Transmitting page 1, call-id: 29.01:43:12.437 VOX.up.ifp.2 <------ PAGE (v29_9600) --------01:51:10.502 VOX.up.ifp.2 <--------- PPS-EOP -----------01:51:13.599 Info vox voip userplan 1 T38 page 1 OK, call-id: 29.01:51:13.599 VOX.up.ifp.2 ------------ MCF ------------>01:51:15.282 VOX.up.ifp.2 <----------- DCN -------------01:51:15.846 Info vox voip controlplan 3 Call Disconnection received on local port: 5/2, cause: (16)[Normal call clearing], call-id: 29.

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Modem / fax passthrough event

Example of event in case of modem/fax call.

One_training>event filter add vox all show

00:11:21.485 Info vox voip controlplan 3 Incoming call on voip id: 0, calling: 111, called: 110, call-id: 3.00:11:21.486 Info vox voip controlplan 3 Outgoing call on local port: 5/0, calling: 111, called: 110, call-id:3.00:11:21.550 Info vox voip controlplan 3 Alert received, call-id: 3.00:11:21.631 Info vox voip controlplan 3 Call connected, call-id: 3.00:11:25.142 Info vox voip userplan 1 Fax/Modem Passthrough starting call-id: 3.00:11:25.142 Info vox voip userplan 3 RTP new transmission coder: G711 A Law, call-id: 3.00:11:25.341 Info vox voip userplan 3 RTP new reception coder: G711 A Law, call-id: 3.00:12:23.533 Info vox voip controlplan 3 Call Disconnection received on local port: 5/1, cause: (16)[Normal call clearing], call-id: 3.00:12:23.552 Info vox voip userplan 3 VoIP RTP transmission stopped, coder: G711 A Law, call-id: 3.00:12:23.552 Info vox voip userplan 3 VoIP RTP reception stopped, coder: G711 A Law, call-id: 3.