2_Voice.ppt
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2_Voice.ppt 2_Voice.ppt Presentation Transcript

  •  
  • Voice
  • Overview
    • Introduction
    • SIP Basic Configuration
    • SIP Advanced Configuration
    • SIP Services
    • Codec Use
    • Country Specific Settings
    • Advanced Voice Routing
    • Debug VoIP
  • Introduction
  • SIP – Network Elements IP Network Registrar Proxy SIP phone SIP phone SIP phone IP / PSTN Gateway PSTN phone Signalling Signalling Data / Voice
  • SIP Session Setup Example 200 OK ACK INVITE sip:picard@uunet.com host.wcom.com sip.uunet.com SIP User Agent Client SIP Proxy BYE 200 OK Media Stream
  • SIP Protocol: Main Actors
    • Client: board, SoftPhone, SIP phone
      • Board / Voice ports
        • FXS1 / FXS2
        • DECT
      • SIP phone ST2020, ST2030, ST288
    • Registrar
    • Proxy
    • Gateways
  • VoIP on Board
    • Inbound SIP client
    • Voice Codecs
    • Voice Services
    • Others
  • SIP ALG
    • Handles connections
    • Does NAT mapping
    • Replaces parts of SIP message containing address / port of Local Network to corresponding ones of NAPT mapping
    SIP SERVER
  • Inbound SIP Client – Illustrated SIP SERVER SIP SERVER
  • SIP Basic Configuration
  • 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
  • Toolbox: Telephony
  • Toolbox: Telephony – Expert Configure
    • Registrar: used as domain-name
      • FQDN or IP
    • Proxy: SIP destination Server IP-address
      • FQDN or IP
  • 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
  • SIP Registration
    • VoIP activation
    • Enable Telephony
      • Register successful
      • Register not successful
  • SIP Advanced Configuration
  • :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
  • :service System – Continued
    • Manually enable VoIP service: :service system modify name=VOIP_SIP state=enable :service system list expand=enabled e.g. Change voice source port, default=5060 :service system modify name=VOIP_SIP port=5091
  • 6.2 SIP-ALG
    • Since 6.2, SIP-ALG has to be used for local voice application
    • =>connection bindlist
    • Application Proto Portrange Flags
    • SIP udp 5060 SIP_ALG:E RTP_predict_for_term_SIP_ALG:E
    • if not in list:
    • :connection bind application=SIP port=5060
    • if FLAG not set
    • = automatically learning RTP flow from SIP SDP negotiation
    • :connection appconfig application=SIP RTP_predict_for_term_SIP_ALG=enabled
  • 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
  • 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
  • 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
    780 / 785 706 / 716
  • Voice FXO
    • =>voice fxoport config
    • Incoming fxo : enabled
    • FXO disconnect timer : 1000
    • [incfxo = <{enabled|disabled}>]
    • Enable or disable incoming FXO calls  disable FXO relay
    • [fxodisconnect = <number{500-5000}>]
    • The FXO disconnect timer (in ms)
  • :voice Configuration
    • CLI: =>:voice help config
    • [autofxo = <{disabled|enabled}>]: automatically make FXO calls when not registered
    • [digitrelay = <{auto|inband|rfc2833|signalling}>]: set digit relay mode
    • [click2dial_ports = <{FXS1|FXS2|all}>]: set click to dial port
    • [rtp_portrange = <port-range>]: RTP port range
    • [sign_internal = <{external|internal}>]: signalling for local calls kept local or external
    • [static_intf = <{disabled|enabled}>]: use static (configured) interface to look for source IP address or not
    • [intf = <{loop|Internet|LocalNetwork}>]: name of IP interface used for VOIP traffic
  • :voice Configuration – Continued
    • CLI: =>:voice help config
    • [secondintf = <{loop|Internet|LocalNetwork}>]: name of backup IP interface used for VOIP traffic
    • [endofnumber = <{#|*}>]: end of number character for dialled number starting with cipher
    • [countrycode = <number{0-999}>]: local country code
    • [delayeddisconnect = <{enabled|disabled}>]: enable or disable delayed disconnect feature
    • [delayeddisconnecttimer = <number{1-600}>]: delayed disconnect timer (in seconds)
    • [ringmuteduration = <number{0-60000}>]: early media mute duration (in minutes)
  • SIP Configuration: Overview
    • :voice sip config
    • UserAgent domain : thomsongateway.sip
    • Primary proxy address : voip.thomson.be:5060
    • Secondary proxy address : 0.0.0.0:5060  Not supported
    • Primary registrar address : voip.Thomson Gateway.be:5060
    • Secondary registrar address : 0.0.0.0:5060  Not supported
    • Listening port : 5060
    • Expire time : 3600
    • Expire time delta : 1
    • Notifier address : 0.0.0.0:5060
    • Subscribe expire time : 3600
  • 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
    • Ping timer : 0
    • Min SE timer : 90
    • Session expires timer : 120
    • Expires timer : 0
  • :voice Profile – Create VoIP User
    • Voice profile add
      • SIP_URI = <string>  telephone number
        • SIP URI related to voice port [username = <string>]
      • Authentication username related to voice port [password = <password>]
      • Authentication password related to voice port [displayname = <string>]  CLIP info
      • Alias name for SIP_URI voiceport = <{FXS1|FXS2|COMMON}>  available ports on TG
      • Analogue line number [abbr = <string>]
    • Abbreviated number mapped to SIP_URI
        • abbr. number only supported when URI has NO LETTERS, only numbers
    • =>voice profile list SIP_URI all
    • Port Uri DisplayName Username Abbr Nbr RegStatus Msg Waiting
    • ---------------------------------------------------------------------------------------------
    • COMMON 2585 2585 2585 85 Registered No
  • :voice Country
    • =>voice country config country=belgium
      • 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
  • :voice Cac
    • =>:voice cac help config
      • [max#portsperprofile = <{one|all}>]
    • 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
  • SIP Services
  • Local Services: 3 Port Call
    • What is needed for 3-way conference call using VoIP?
    Voice Network hold R+2 3way R+3 switch R+2
    • Answer or make call
    • Put 1 st call on hold
    • Make 2 nd call
    • Switch between calls or put in 3-way conference call
    Services available?
  • 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
  • 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
  • Codec Use
  • Codec Support G.729AnnexA audio at 8 Kbit/s with silence suppression as in AnnexB G.729AnnexA audio at 8 Kbit/s G.726: ADPCM at 40 Kbit/s G.726: ADPCM at 32 Kbit/s G.726: ADPCM at 24 Kbit/s G.726: ADPCM at 16 Kbit/s G.723.I at either 5.3 or 6.3 Kbit/s with silence suppression as in AnnexA G.723.I at either 5.3 or 6.3 Kbit/s G.711 audio at 64 Kbit/s, µ-law G.711 audio at 64 Kbit/s, A-law Codec
  • Advanced Voice Routing
  • 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
  • Internal Overview dhcp voice_ip_intf default bridge video group id=2 bridge pvc 8/35 ETH ports Flexiport move Eth-port to video group Eth bridge Eth-mer-if ip voice_ip_intf PPP relay PPP Routing PPP - default MAC MER - USB MAC Eth-mer-if is connected to bridge group video VoIP - SIP application
  • VoIP Interface Specific
    • Source IP address selection (on interface)
      • :voip config static_intf=enabled intf=voip_ip_intf
    • VoIP label for RTP routing
      • RTP via SIP-ALG and label inheritance
      • (default 7.4)
    • =>connection bindlist
    • Application Proto Portrange Flags
    • SIP udp 5060 SIP_ALG:E RTP_predict_for_term_SIP_ALG:E
    {Administrator}[label]=>list 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
  • Add Label Routing
    • default QoS rule
    • :label rule add chain=qos_default_labels index=3 serv=sip log=disabled state=enabled label=VoIP
    • to be added for voice routing:
    • :label add Voip_only inheritance=enabled
    • :label rule add chain=rt_user_labels index=1 srcintf=local serv=sip log=disabled state=enabled label=Voip_only
    • Label rule instructs that all CPE SIP traffic has to use this label
  • Label Routing / Forwarding
    • Static IP-address :ip rtadd dst 0.0.0.0/0 label=Voip_only gateway=123.123.123.123
    • PPP VoIP interface :ppp rtadd intf=Internet dst=0.0.0.0/0 label=Voip_only
    • DHCP-Client VoIP interface :dhcp client ifconfig intf=voip_ip_intf label=Voip_only gateway=enabled
  • Debug VoIP
  • 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?
  • 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 cmd=“trace 1”
    • Disable -> :voice debug exec cmd=“trace 0”
  • Hands on - debugging
    • Install and configure x-lite (SoftPhone)
  • Debugging
  • Debugging
  • Debugging
    • 200 OK is received by the CPE but not forwarded to the Computer running x-lite
  • Debugging
    • [IN]LocalNet-> : 192.168.1.64 192.168.0.101 0583 UDP 62748->5080
    • [UT]LocalNet->ip_voice : 192.168.0.104 192.168.0.101 0583 UDP 49252->5080
    • [IN] ip_voice-> : 192.168.0.101 192.168.0.104 0443 UDP 5080->62748
    • [DR] ip_voice->ip_voice : 192.168.0.101 192.168.0.104 0443 UDP 5080->62748
    • : 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?
  • 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
    {Administrator}=>connection bindlist Application Proto Portrange Flags SIP udp 5060 SIP_ALG:E RTP_predict_for_term_SIP_ALG:E IKE udp 500 {Administrator}=>:connection bind application=SIP port=5080