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Voice

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... appconfig application=SIP RTP_predict_for_term_SIP_ALG=enabled Voice Ports FXS = Analogue phone FXO = PSTN-line DECT = DECT handsets associated to ST7x7 ... – PowerPoint PPT presentation

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Title: Voice


1
(No Transcript)
2
Voice
3
Overview
  • Introduction
  • SIP Basic Configuration
  • SIP Advanced Configuration
  • SIP Services
  • Codec Use
  • Country Specific Settings
  • Advanced Voice Routing
  • Debug VoIP

4
Introduction
5
SIP Network Elements
Proxy
SIP phone
IP Network
Registrar
IP / PSTN Gateway
PSTN phone
SIP phone
SIP phone
6
SIP Session Setup Example
SIP User Agent Client
SIP Proxy
INVITE sippicard_at_uunet.com
200 OK
ACK
Media Stream
BYE
200 OK
host.wcom.com
sip.uunet.com
7
SIP Protocol Main Actors
  • Client board, SoftPhone, SIP phone
  • Board / Voice ports
  • FXS1 / FXS2
  • DECT
  • SIP phone ST2020, ST2030, ST288
  • Registrar
  • Proxy
  • Gateways

8
VoIP on Board
  • Inbound SIP client
  • Voice Codecs
  • Voice Services
  • Others

9
SIP ALG
  • Handles connections
  • Does NAT mapping
  • Replaces parts of SIP message containing address
    / port of Local Network to corresponding ones of
    NAPT mapping

10
Inbound SIP Client Illustrated
11
SIP Basic Configuration
12
Basic Configuration GUI Access
  • Factory defaults on LAN
  • DHCP-Server enabled on PC
  • Thomson Gateway IP-address 192.168.1.254
  • URL http//192.168.1.254
  • Telnet/GUI Username Administrator
  • No password
  • Thomson Gateway Set-Up Wizard
  • This launches pop-up window for easy configuration

13
Toolbox Telephony
14
Toolbox Telephony Expert Configure
  • Registrar used as domain-name
  • FQDN or IP
  • Proxy SIP destination Server IP-address
  • FQDN or IP

15
Toolbox Telephony Configure
  • SIP URI telephone number
  • Username and Password SIP authentication
  • Display name CLIP
  • Abbr. Number internal number between local
    phones
  • Port assign number to

16
SIP Registration
  • VoIP activation
  • Enable Telephony
  • Register successful
  • Register not successful

17
SIP Advanced Configuration
18
service System
  • Service manager on Thomson Gateway
  • Enable / disable service
  • Change local port (source port)
  • Access list on Interface/IP-address
  • Labels for routing and QoS
  • Service manager automatically generates
  • NAT entries
  • Firewall rules

19
service System Continued
  • Manually enable VoIP serviceservice system
    modify nameVOIP_SIP stateenableservice system
    list expandenablede.g. Change voice source
    port, default5060service system modify
    nameVOIP_SIP port5091

20
6.2 SIP-ALG
  • Since 6.2, SIP-ALG has to be used for local voice
    application
  • gtconnection bindlist
  • Application Proto Portrange Flags
  • SIP udp 5060 SIP_ALGE
    RTP_predict_for_term_SIP_ALGE
  • if not in list
  • connection bind applicationSIP port5060
  • if FLAG not set
  • automatically learning RTP flow from SIP SDP
    negotiation
  • connection appconfig applicationSIP
    RTP_predict_for_term_SIP_ALGenabled

21
Voice Ports
  • FXS Analogue phone
  • FXO PSTN-line
  • DECT DECT handsets associated to ST7x7
  • Common ports are groups of voice ports
  • All / COMMON all available ports
  • All DECT

22
Common Number
  • Multiple common numbers supported
  • Outgoing call from local voice port uses COMMON
    number as CID (Caller ID)
  • When no local number specifically configured for
    that port
  • Automatically activated when COMMON number used
  • Cannot be withdrawn
  • Incoming call on COMMON number rings all
    available ports

23
Voice HW Regulatory and FXO
  • Analogue Outgoing Telephone LinePOTS Back-Up
  • Level 0 No FXO nor PSTN Backup
  • Level 1 ONLY support for power failure to dial
    out
  • Level 2 Power Failed VoIP Service Failed
    Prefix dialing first then PSTN Incoming FXO
    Calls
  • Reduced FXO
  • Level 3 Power Failed VoIP Service Failed
    Prefix dialing first then PSTN Incoming FXO
    Calls Emergence call without prefix dialing
    needs (ex 110, 911... Specific No. can be
    configurable by customers...etc.)
  • Full FXO

706 / 716
780 / 785
24
Voice FXO
  • gtvoice fxoport config
  • Incoming fxo enabled
  • FXO disconnect timer 1000
  • incfxo ltenableddisabledgt
  • Enable or disable incoming FXO calls ?
    disable FXO relay
  • fxodisconnect ltnumber500-5000gt
  • The FXO disconnect timer (in ms)

25
voice Configuration
  • CLI gtvoice help config
  • autofxo ltdisabledenabledgt automatically
    make FXO calls when not registered
  • digitrelay ltautoinbandrfc2833signallinggt
    set digit relay mode
  • click2dial_ports ltFXS1FXS2allgt set click
    to dial port
  • rtp_portrange ltport-rangegt RTP port range
  • sign_internal ltexternalinternalgt
    signalling for local calls kept local or external
  • static_intf ltdisabledenabledgt use static
    (configured) interface to look for source IP
    address or not
  • intf ltloopInternetLocalNetworkgt name of
    IP interface used for VOIP traffic

26
voice Configuration Continued
  • CLI gtvoice help config
  • secondintf ltloopInternetLocalNetworkgt
    name of backup IP interface used for VOIP traffic
  • endofnumber ltgt end of number character
    for dialled number starting with cipher
  • countrycode ltnumber0-999gt local country
    code
  • delayeddisconnect ltenableddisabledgt
    enable or disable delayed disconnect feature
  • delayeddisconnecttimer ltnumber1-600gt
    delayed disconnect timer (in seconds)
  • ringmuteduration ltnumber0-60000gt early
    media mute duration (in minutes)

27
SIP Configuration Overview
  • voice sip config
  • UserAgent domain
    thomsongateway.sip
  • Primary proxy address
    voip.thomson.be5060
  • Secondary proxy address 0.0.0.05060 ?
    Not supported
  • Primary registrar address voip.Thomson
    Gateway.be5060
  • Secondary registrar address 0.0.0.05060 ?
    Not supported
  • Listening port 5060
  • Expire time 3600
  • Expire time delta 1
  • Notifier address 0.0.0.05060
  • Subscribe expire time 3600

28
SIP Configuration Overview Continued
  • voice sip config
  • Call Waiting reply 182
  • Transport UDP
  • rtpmapstaticPT Disabled
  • reinvite_stop_audio Disabled
  • PRACK Disabled
  • Clir format standard
  • DTMF / in INFO method 1011
  • Clip consider displayname yes
  • SDP packet time 20
  • Replace Enabled
  • Symmetric codec Enabled
  • Reinvite at calling fax detect Disabled
  • SIPURI port Enabled
  • rport Disabled
  • SDP username 780
  • ringtoneat183 Disabled
  • T38 Port increment 0

29
voice Profile Create VoIP User
  • Voice profile add
  • SIP_URI ltstringgt ? telephone number
  • SIP URI related to voice port username
    ltstringgt
  • Authentication username related to voice
    portpassword ltpasswordgt
  • Authentication password related to voice
    portdisplayname ltstringgt ? CLIP info
  • Alias name for SIP_URIvoiceport
    ltFXS1FXS2COMMONgt ? available ports on TG
  • Analogue line number abbr ltstringgt
  • Abbreviated number mapped to SIP_URI
  • abbr. number only supported when URI has NO
    LETTERS, only numbers
  • gtvoice profile list SIP_URI all
  • Port Uri DisplayName Username
    Abbr Nbr RegStatus Msg Waiting
  • --------------------------------------------------
    -------------------------------------------
  • COMMON 2585 2585 2585
    85 Registered No

30
voice Country
  • gtvoice country config countrybelgium
  • Pre-loaded country settings
  • Country australia belgium denmark etsi
    france1 france2 france3 germany italy
    netherlands northamerica norway spain
    sweden uk
  • Country specific settings
  • DTMF tones / dial tones / Hook flash timer /
    polarity / etc.
  • Can be changed to specific needs
  • Special file in dl-directory vincfg.bin

31
voice Cac
  • gtvoice cac help config
  • maxportsperprofile ltoneallgt
  • Maximum number of ports that can be used with
    common profile
  • One only 1 call with common number possible at
    same time
  • All multiple calls with common number possible
    at same time

32
SIP Services
33
Local Services 3 Port Call
  • What is needed for 3-way conference call using
    VoIP?
  1. Answer or make call
  2. Put 1st call on hold
  3. Make 2nd call
  4. Switch between calls or put in 3-way conference
    call

Services available?
Voice Network
hold R2
3way R3
switch R2
34
Supplementary Services SpeedTouch 6.1
  • Transfer Call Transfer between local ports
  • Hold put active Call on Hold
  • Waiting incoming call while active call
    indication
  • Mwi Message Waiting Indication
  • Clip Calling Line Identification Presentation
  • Clir Calling Line Identification Restriction
  • 3pty Three Party Call

35
Supplementary Services SpeedTouch 6.1 Continued
  • forcedFXO switch to FXO (PSTN)
  • Cfu Call Forwarding Unconditional
  • Cfnr Call Forwarding on No Reply
  • Cfbs Call Forwarding on Busy
  • Ccbs Call Completion on Busy Subscriber
  • Clironcall CLIR for only one call
  • Waitingoncall Call Waiting active for only one
    call

36
Codec Use
37
Codec Support
Codec
G.711 audio at 64 Kbit/s, A-law
G.711 audio at 64 Kbit/s, µ-law
G.723.I at either 5.3 or 6.3 Kbit/s
G.723.I at either 5.3 or 6.3 Kbit/s with silence suppression as in AnnexA
G.726 ADPCM at 16 Kbit/s
G.726 ADPCM at 24 Kbit/s
G.726 ADPCM at 32 Kbit/s
G.726 ADPCM at 40 Kbit/s
G.729AnnexA audio at 8 Kbit/s
G.729AnnexA audio at 8 Kbit/s with silence suppression as in AnnexB
38
Advanced Voice Routing
39
Advanced Routed Scenario
  • Multiple routed interfaces
  • How VoIP interface binding?
  • VoIP routing-based
  • Best route used
  • SIP SoftSwitch is known IP-address
  • UA and RTP?
  • No IP-address known
  • All VoIP GW configured in routing ? complex

40
Internal Overview
VoIP - SIP application
Routing
dhcp voice_ip_intf
PPP
ip voice_ip_intf
PPP relay
PPP - default MAC
MER - USB MAC
Eth-mer-if
Eth bridge
default bridge
Eth-mer-if is connected to bridge group video
ETH ports
video group id2
bridge pvc 8/35
Flexiport move Eth-port to video group
41
VoIP Interface Specific
  • Source IP address selection (on interface)
  • voip config static_intfenabled
    intfvoip_ip_intf
  • VoIP label for RTP routing
  • RTP via SIP-ALG and label inheritance
  • (default 7.4)
  • gtconnection bindlist
  • Application Proto Portrange Flags
  • SIP udp 5060 SIP_ALGE
    RTP_predict_for_term_SIP_ALGE

Administratorlabelgtlist Name Class
Def Ack Bidirect Inherit
Tosmark Type Value Use Trace
-------------------------------------------------
--------------------------------------------------
-------- DSCP overwrite dscp
prioritize disabled disabled disabled tos
0 1 disabled Interactive increase 8
6 disabled disabled disabled
tos 0 14 disabled Management
increase 12 12 disabled
disabled disabled tos 0 4
disabled Video increase 10 10
disabled disabled disabled tos 0
2 disabled VoIP-RTP overwrite 14
14 enabled disabled disabled tos
0 1 disabled VoIP-Signal overwrite
12 12 enabled disabled disabled
tos 0 2 disabled default
increase default prioritize disabled
disabled disabled tos 0 1
disabled
42
Add Label Routing
  • default QoS rule
  • label rule add chainqos_default_labels index3
    servsip logdisabled stateenabled labelVoIP
  • to be added for voice routing
  • label add Voip_only inheritanceenabled
  • label rule add chainrt_user_labels index1
    srcintflocal servsip logdisabled stateenabled
    labelVoip_only
  • Label rule instructs that all CPE SIP traffic has
    to use this label

43
Label Routing / Forwarding
  • Static IP-address ip rtadd dst 0.0.0.0/0
    labelVoip_only gateway123.123.123.123
  • PPP VoIP interfaceppp rtadd intfInternet
    dst0.0.0.0/0 labelVoip_only
  • DHCP-Client VoIP interfacedhcp client ifconfig
    intfvoip_ip_intf labelVoip_only gatewayenabled

44
Debug VoIP
45
Tips and Tricks
  • Services not working
  • No hookflash detection etc.?
  • Change country to etsi or northamerica
  • Different analogue phone settings / timers
  • One way voice
  • Codec priority changes
  • Enable / disable codecs
  • Ptime changes
  • Ptime acceptable for Gateway?
  • Connection bindlist
  • SIP-ALG bounded?
  • Symmetrical codec
  • Enabled for 7G?

46
Debug VoIP Standard Traces
  • Ctrl-q ? start debug
  • Ctrl-s ? stop debug
  • Ctrl-t ? clear buffer
  • Ethereal trace has VoIP flow
  • Statistics VoIP Call
  • Additional voice traces
  • Enable ? voice debug exec cmdtrace 1
  • Disable ? voice debug exec cmdtrace 0

47
Hands on - debugging
  • Install and configure x-lite (SoftPhone)

48
Debugging
49
Debugging
50
Debugging
  • 200 OK is received by the CPE but not forwarded
    to the Computer running x-lite

51
Debugging
  • INLocalNet-gt 192.168.1.64
    192.168.0.101 0583 UDP 62748-gt5080
  • UTLocalNet-gtip_voice 192.168.0.104
    192.168.0.101 0583 UDP 49252-gt5080
  • IN ip_voice-gt 192.168.0.101
    192.168.0.104 0443 UDP 5080-gt62748
  • DR ip_voice-gtip_voice 192.168.0.101
    192.168.0.104 0443 UDP 5080-gt62748
  • error caused by NAT-INPUT
  • The 200 OK is dropped because the packet is
    received on the wrong port
  • Why does the SIPServer reply on 62748?

52
Solving
  • The data sent to the SIPserver is wrong obviously
    because of NAT.
  • The ALG should modify the content and generate
    the appropriated NAT entry.
  • The SIP ALG is bound to the wrong Port

Administratorgtconnection bindlist
Application Proto Portrange Flags SIP
udp 5060 SIP_ALGE
RTP_predict_for_term_SIP_ALGE IKE udp
500
Administratorgtconnection bind applicationSIP
port5080
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