OneOs Voice Configuration Dial-Peer Concept H SIP 4

OneOs Voice Configuration Dial-Peer Concept 4. 11 H323 4. 15 SIP 4
OneOs Voice Configuration Dial-Peer Concept H SIP 4.21 MGCP v

Copyright © OneAccess Networks – All rights reserved
VoIP Configuration Diagram (H.323 / SIP) ONE CLASSICAL TELEPHONE NETWORK IP NETWORK Voice Routing 6 VoIP Coder Profile 5 Interface BRI/PRI 2 Interface BRI/PRI H.323 or SIP Gateway 7 ISDN PBX Dial-Peer Voice VoIP 4 BRI PRI POTS Group 1 Dial-Peer Voice POTS 3 Digital Physical Voice-port Digital Physical Voice-port 1 BRI Dial-Peer Voice VoIP ISDN Phone SIP Server 8 POTS Group 2 Dial-Peer Voice POTS 3 Analogue Physical Voice-port Analogue Physical Voice-port 1 POTS SIP Phones Copyright © OneAccess Networks – All rights reserved

VoIP Configuration Diagram (MGCP) ONE CLASSICAL TELEPHONE NETWORK IP NETWORK VoIP Coder Profile 5 MGCP Gateway 7 4 Dial-Peer Voice VoIP POTS Group 2 Dial-Peer Voice POTS 3 Analogue Physical Voice-port Analogue Physical Voice-port 1 Copyright © OneAccess Networks – All rights reserved

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

Voice Port Configuration 1 - Physical voice ports (1/3) BRI (2 or 4 or 8 ports) or FXS (4 or 8 ports) PRI CLI# configure terminal CLI(configure)# voice-port 5/0 CLI(voice-port)# exit CLI(configure)# voice-port 5/1 . . . CLI(configure)# voice-port 5/7 one200>conf t one200(configure)>voice-port 5/0 one200(voice-port)>exit Copyright © OneAccess Networks – All rights reserved

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

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

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

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)> Copyright © OneAccess Networks – All rights reserved

VoIP Configuration 2 - Interfaces (3/4) BRI: Example with ISDN phone BRI: Example with PBX one200(configure)>interface bri 5/0 one200(conf-if)> isdn one200(isdn)> protocol-emulation isdn-nt one200(isdn)> exit one200(conf-if)> no shutdown one200(conf-if)> execute one200(conf-if)> exit one200(configure)> interface bri 5/0 one200(conf-if)> isdn one200(isdn)> tei-negotiation static one200(isdn)> protocol-emulation isdn-nt one200(isdn)> exit one200(conf-if)> no shutdown one200(conf-if)> execute one200(conf-if)> exit Copyright © OneAccess Networks – All rights reserved

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 Copyright © OneAccess Networks – All rights reserved

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

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 Copyright © OneAccess Networks – All rights reserved

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)> Copyright © OneAccess Networks – All rights reserved

Dial-peer VoIP (1/2) 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. Copyright © OneAccess Networks – All rights reserved

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) Copyright © OneAccess Networks – All rights reserved

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 Copyright © OneAccess Networks – All rights reserved

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 Copyright © OneAccess Networks – All rights reserved

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: depending on the coder <cr> one200(voip-coder)# codec 0 g729ab 30 one200(voip-coder)# codec 1 g711a 20 Copyright © OneAccess Networks – All rights reserved

SIP Gateway (1/2) 7. SIP gateway 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) Copyright © OneAccess Networks – All rights reserved

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 } voic -dns-add Set Voic characteristics Copyright © OneAccess Networks – All rights reserved

SIP Gateway 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 intrusive CLI(sipgw)# no shutdown CLI(sipgw)# gw-interface fastethernet 0 intrusive CLI(sipgw)# prox-dns-add CLI(sipgw)# no shutdown CLI(sipgw)# gw-interface fastethernet 0 intrusive CLI(sipgw)# reg-dns-add CLI(sipgw)# prox-dns-add CLI(sipgw)# no shutdown Copyright © OneAccess Networks – All rights reserved

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 (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. Copyright © OneAccess Networks – All rights reserved

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 Copyright © OneAccess Networks – All rights reserved

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)> Copyright © OneAccess Networks – All rights reserved

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 -id Set H323 identifier for the gateway exit Exit from command node gatekeeper Set H323 gateway - main gatekeeper parameters gw-address Set H323 gateway - gateway address mode gw-interface Output Interface category for H323 GW - default fastethernet 0 gw-interface-bw-ctrl - Set H323 gateway - gw interface bandwidth control gw-prefix Set H323 gateway – prefix h235-authentication - Set H235 authentication h245-response-timeou - Set H323 gateway - timeout used for H245 protocol ... Copyright © OneAccess Networks – All rights reserved

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 Copyright © OneAccess Networks – All rights reserved

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 Copyright © OneAccess Networks – All rights reserved

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> Copyright © OneAccess Networks – All rights reserved

VoIP Configuration 7 - H.323 Gateway (6/6) one200(config)> h323-gateway one200(h323gw)> gw-interface fastethernet 0/0 one200(h323gw)> gatekeeper id training address one200(h323gw)> h323-id GW1 one200(h323gw)> max-bandwidth one200(h323gw)> no shutdown Copyright © OneAccess Networks – All rights reserved

Voice Toubleshooting & Statistics

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

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

VoIP Statistics FXS statistics CLI# show voice voice-port fxs index 0 voice port /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 Copyright © OneAccess Networks – All rights reserved

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

VoIP Statistics Dial-peer VoIP Statistics (2): Outgoing Calls Dial Peer Current Calls 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) 0 Cause Class 3 (unavailable service) 0 Cause Class 4 (service not provided) 0 Cause Class 5 (invalid message) Cause Class 6 (protocol error) Cause Class 7 (interworking) H245 Call failures Incompatible capabilities Protocol errors Internal call failures DSP unavailable Max-bandwidth exceeded Max-connection exceeded Not specified Copyright © OneAccess Networks – All rights reserved

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

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 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 Copyright © OneAccess Networks – All rights reserved

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

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

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 :16384 – :16386 Play time (voice) : 00h00m46s Tx Coder : G729 / 20 ms ; Rx Coder : G729 VAD enabled local / remote : no / no ERL : 15 dB ACOM : 32 dB RTP Packets received (DSP / Uplink) : 2337 / 2337 lost : 0 out of sequence : 0 invalid : 0 RTP Packets transmitted (DSP / Uplink) : 2338 / 2338 lost (RTCP reported) : 0 Jitter parameter : 100 ms Number of Excessive Jitter events : 1 Copyright © OneAccess Networks – All rights reserved

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

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

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

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

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

OneOs Voice Configuration Dial-Peer Concept H SIP 4

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