Voice
Overview <ul><li>Introduction </li></ul><ul><li>SIP Basic Configuration </li></ul><ul><li>SIP Advanced Configuration </li>...
Introduction
SIP – Network Elements IP Network Registrar Proxy SIP phone SIP phone SIP phone IP / PSTN Gateway PSTN phone Signalling Si...
SIP Session Setup Example 200 OK ACK INVITE sip:picard@uunet.com host.wcom.com sip.uunet.com SIP User Agent Client SIP Pro...
SIP Protocol: Main Actors <ul><li>Client: board, SoftPhone, SIP phone </li></ul><ul><ul><li>Board / Voice ports </li></ul>...
VoIP on Board <ul><li>Inbound SIP client </li></ul><ul><li>Voice Codecs </li></ul><ul><li>Voice Services </li></ul><ul><li...
SIP ALG <ul><li>Handles connections  </li></ul><ul><li>Does NAT mapping </li></ul><ul><li>Replaces parts of SIP message co...
Inbound SIP Client – Illustrated SIP SERVER SIP SERVER
SIP Basic Configuration
Basic Configuration – GUI Access <ul><li>Factory defaults on LAN </li></ul><ul><ul><li>DHCP-Server enabled on PC </li></ul...
Toolbox: Telephony
Toolbox: Telephony – Expert Configure <ul><li>Registrar: used as domain-name </li></ul><ul><ul><li>FQDN or IP </li></ul></...
Toolbox: Telephony – Configure <ul><li>SIP URI: telephone number </li></ul><ul><li>Username and Password: SIP authenticati...
SIP Registration <ul><li>VoIP activation </li></ul><ul><li>Enable Telephony </li></ul><ul><ul><li>Register successful </li...
SIP Advanced Configuration
:service System <ul><li>Service manager on Thomson Gateway </li></ul><ul><ul><li>Enable / disable service </li></ul></ul><...
:service System – Continued <ul><li>Manually enable VoIP service: :service system modify name=VOIP_SIP state=enable :servi...
6.2 SIP-ALG <ul><li>Since 6.2, SIP-ALG has to be used for local voice application </li></ul><ul><li>=>connection bindlist ...
Voice Ports <ul><li>FXS = Analogue phone </li></ul><ul><li>FXO = PSTN-line </li></ul><ul><li>DECT = DECT handsets associat...
Common Number <ul><li>Multiple common numbers supported  </li></ul><ul><li>Outgoing call from local voice port uses COMMON...
Voice HW: Regulatory and FXO  <ul><li>Analogue Outgoing Telephone LinePOTS Back-Up </li></ul><ul><ul><li>Level 0 :  No FXO...
Voice FXO <ul><li>=>voice fxoport config  </li></ul><ul><li>Incoming fxo  : enabled </li></ul><ul><li>FXO disconnect timer...
:voice Configuration <ul><li>CLI: =>:voice help config   </li></ul><ul><li>[autofxo = <{disabled|enabled}>]:  automaticall...
:voice Configuration – Continued <ul><li>CLI: =>:voice help config   </li></ul><ul><li>[secondintf = <{loop|Internet|Local...
SIP Configuration: Overview <ul><li>:voice sip config   </li></ul><ul><li>UserAgent domain  : thomsongateway.sip  </li></u...
SIP Configuration: Overview – Continued <ul><li>:voice sip config   </li></ul><ul><li>Call Waiting reply  : 182 </li></ul>...
:voice Profile – Create VoIP User <ul><li>Voice profile add </li></ul><ul><ul><li>SIP_URI = <string>    telephone number ...
:voice Country <ul><li>=>voice country config country=belgium </li></ul><ul><ul><li>Pre-loaded country settings </li></ul>...
:voice Cac <ul><li>=>:voice cac help config </li></ul><ul><ul><li>[max#portsperprofile = <{one|all}>]  </li></ul></ul><ul>...
SIP Services
Local Services: 3 Port Call <ul><li>What is needed for 3-way conference call using VoIP? </li></ul>Voice Network hold R+2 ...
Supplementary Services SpeedTouch 6.1 <ul><li>Transfer:  Call Transfer between local ports </li></ul><ul><li>Hold:  put ac...
Supplementary Services SpeedTouch 6.1 – Continued <ul><li>forcedFXO:  switch to FXO (PSTN) </li></ul><ul><li>Cfu:  Call Fo...
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: ADP...
Advanced Voice Routing
Advanced Routed Scenario <ul><li>Multiple routed interfaces </li></ul><ul><li>How VoIP interface binding? </li></ul><ul><l...
Internal Overview dhcp  voice_ip_intf default bridge video group id=2 bridge pvc 8/35 ETH ports Flexiport move  Eth-port t...
VoIP Interface Specific <ul><li>Source IP address selection (on interface) </li></ul><ul><ul><li>:voip config static_intf=...
Add Label Routing <ul><li>default QoS rule </li></ul><ul><li>:label rule add chain=qos_default_labels index=3 serv=sip log...
Label Routing / Forwarding <ul><li>Static IP-address  :ip rtadd dst 0.0.0.0/0 label=Voip_only gateway=123.123.123.123 </li...
Debug VoIP
Tips and Tricks <ul><li>Services not working </li></ul><ul><ul><li>No hookflash detection etc.? </li></ul></ul><ul><ul><ul...
Debug VoIP: Standard Traces <ul><li>Ctrl-q  ->  start debug </li></ul><ul><li>Ctrl-s  ->  stop debug </li></ul><ul><li>Ctr...
Hands on - debugging <ul><li>Install and configure x-lite (SoftPhone) </li></ul>
Debugging
Debugging
Debugging <ul><li>200 OK is received by the CPE but not forwarded to the Computer running x-lite </li></ul>
Debugging <ul><li>[IN]LocalNet->  : 192.168.1.64  192.168.0.101  0583 UDP 62748->5080 </li></ul><ul><li>[UT]LocalNet->ip_v...
Solving <ul><li>The data sent to the SIPserver is wrong obviously because of NAT. </li></ul><ul><li>The ALG should modify ...
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2_Voice.ppt

  1. 2. Voice
  2. 3. Overview <ul><li>Introduction </li></ul><ul><li>SIP Basic Configuration </li></ul><ul><li>SIP Advanced Configuration </li></ul><ul><li>SIP Services </li></ul><ul><li>Codec Use </li></ul><ul><li>Country Specific Settings </li></ul><ul><li>Advanced Voice Routing </li></ul><ul><li>Debug VoIP </li></ul>
  3. 4. Introduction
  4. 5. SIP – Network Elements IP Network Registrar Proxy SIP phone SIP phone SIP phone IP / PSTN Gateway PSTN phone Signalling Signalling Data / Voice
  5. 6. 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
  6. 7. SIP Protocol: Main Actors <ul><li>Client: board, SoftPhone, SIP phone </li></ul><ul><ul><li>Board / Voice ports </li></ul></ul><ul><ul><ul><li>FXS1 / FXS2 </li></ul></ul></ul><ul><ul><ul><li>DECT </li></ul></ul></ul><ul><ul><li>SIP phone ST2020, ST2030, ST288 </li></ul></ul><ul><li>Registrar </li></ul><ul><li>Proxy </li></ul><ul><li>Gateways </li></ul>
  7. 8. VoIP on Board <ul><li>Inbound SIP client </li></ul><ul><li>Voice Codecs </li></ul><ul><li>Voice Services </li></ul><ul><li>Others </li></ul>
  8. 9. SIP ALG <ul><li>Handles connections </li></ul><ul><li>Does NAT mapping </li></ul><ul><li>Replaces parts of SIP message containing address / port of Local Network to corresponding ones of NAPT mapping </li></ul>SIP SERVER
  9. 10. Inbound SIP Client – Illustrated SIP SERVER SIP SERVER
  10. 11. SIP Basic Configuration
  11. 12. Basic Configuration – GUI Access <ul><li>Factory defaults on LAN </li></ul><ul><ul><li>DHCP-Server enabled on PC </li></ul></ul><ul><ul><li>Thomson Gateway IP-address: 192.168.1.254 </li></ul></ul><ul><ul><li>URL: http://192.168.1.254 </li></ul></ul><ul><ul><li>Telnet/GUI Username: Administrator </li></ul></ul><ul><ul><li>No password </li></ul></ul><ul><li>Thomson Gateway Set-Up Wizard </li></ul><ul><ul><li>This launches pop-up window for easy configuration </li></ul></ul>
  12. 13. Toolbox: Telephony
  13. 14. Toolbox: Telephony – Expert Configure <ul><li>Registrar: used as domain-name </li></ul><ul><ul><li>FQDN or IP </li></ul></ul><ul><li>Proxy: SIP destination Server IP-address </li></ul><ul><ul><li>FQDN or IP </li></ul></ul>
  14. 15. Toolbox: Telephony – Configure <ul><li>SIP URI: telephone number </li></ul><ul><li>Username and Password: SIP authentication </li></ul><ul><li>Display name: CLIP </li></ul><ul><li>Abbr. Number: internal number between local phones </li></ul><ul><li>Port: assign number to </li></ul>
  15. 16. SIP Registration <ul><li>VoIP activation </li></ul><ul><li>Enable Telephony </li></ul><ul><ul><li>Register successful </li></ul></ul><ul><ul><li>Register not successful </li></ul></ul>
  16. 17. SIP Advanced Configuration
  17. 18. :service System <ul><li>Service manager on Thomson Gateway </li></ul><ul><ul><li>Enable / disable service </li></ul></ul><ul><ul><li>Change local port (source port) </li></ul></ul><ul><ul><li>Access list on Interface/IP-address </li></ul></ul><ul><ul><li>Labels for routing and QoS </li></ul></ul><ul><li>Service manager automatically generates </li></ul><ul><ul><li>NAT entries </li></ul></ul><ul><ul><li>Firewall rules </li></ul></ul>
  18. 19. :service System – Continued <ul><li>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 </li></ul>
  19. 20. 6.2 SIP-ALG <ul><li>Since 6.2, SIP-ALG has to be used for local voice application </li></ul><ul><li>=>connection bindlist </li></ul><ul><li>Application Proto Portrange Flags </li></ul><ul><li>SIP udp 5060 SIP_ALG:E RTP_predict_for_term_SIP_ALG:E </li></ul><ul><li>if not in list: </li></ul><ul><li>:connection bind application=SIP port=5060 </li></ul><ul><li>if FLAG not set </li></ul><ul><li>= automatically learning RTP flow from SIP SDP negotiation </li></ul><ul><li>:connection appconfig application=SIP RTP_predict_for_term_SIP_ALG=enabled </li></ul>
  20. 21. Voice Ports <ul><li>FXS = Analogue phone </li></ul><ul><li>FXO = PSTN-line </li></ul><ul><li>DECT = DECT handsets associated to ST7x7 </li></ul><ul><li>Common ports are groups of voice ports </li></ul><ul><li>All / COMMON = all available ports </li></ul><ul><li>All DECT </li></ul>
  21. 22. Common Number <ul><li>Multiple common numbers supported </li></ul><ul><li>Outgoing call from local voice port uses COMMON number as CID (Caller ID) </li></ul><ul><ul><li>When no local number specifically configured for that port </li></ul></ul><ul><ul><li>Automatically activated when COMMON number used </li></ul></ul><ul><ul><li>Cannot be withdrawn </li></ul></ul><ul><li>Incoming call on COMMON number rings all available ports </li></ul>
  22. 23. Voice HW: Regulatory and FXO <ul><li>Analogue Outgoing Telephone LinePOTS Back-Up </li></ul><ul><ul><li>Level 0 : No FXO nor PSTN Backup </li></ul></ul><ul><ul><li>Level 1 : ONLY support for power failure to dial out </li></ul></ul><ul><ul><li>Level 2 : Power Failed + VoIP Service Failed + Prefix dialing first then PSTN + Incoming FXO Calls </li></ul></ul><ul><ul><li>Reduced FXO </li></ul></ul><ul><ul><li>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.) </li></ul></ul><ul><ul><li>Full FXO </li></ul></ul>780 / 785 706 / 716
  23. 24. Voice FXO <ul><li>=>voice fxoport config </li></ul><ul><li>Incoming fxo : enabled </li></ul><ul><li>FXO disconnect timer : 1000 </li></ul><ul><li>[incfxo = <{enabled|disabled}>] </li></ul><ul><li>Enable or disable incoming FXO calls  disable FXO relay </li></ul><ul><li>[fxodisconnect = <number{500-5000}>] </li></ul><ul><li>The FXO disconnect timer (in ms) </li></ul>
  24. 25. :voice Configuration <ul><li>CLI: =>:voice help config </li></ul><ul><li>[autofxo = <{disabled|enabled}>]: automatically make FXO calls when not registered </li></ul><ul><li>[digitrelay = <{auto|inband|rfc2833|signalling}>]: set digit relay mode </li></ul><ul><li>[click2dial_ports = <{FXS1|FXS2|all}>]: set click to dial port </li></ul><ul><li>[rtp_portrange = <port-range>]: RTP port range </li></ul><ul><li>[sign_internal = <{external|internal}>]: signalling for local calls kept local or external </li></ul><ul><li>[static_intf = <{disabled|enabled}>]: use static (configured) interface to look for source IP address or not </li></ul><ul><li>[intf = <{loop|Internet|LocalNetwork}>]: name of IP interface used for VOIP traffic </li></ul>
  25. 26. :voice Configuration – Continued <ul><li>CLI: =>:voice help config </li></ul><ul><li>[secondintf = <{loop|Internet|LocalNetwork}>]: name of backup IP interface used for VOIP traffic </li></ul><ul><li>[endofnumber = <{#|*}>]: end of number character for dialled number starting with cipher </li></ul><ul><li>[countrycode = <number{0-999}>]: local country code </li></ul><ul><li>[delayeddisconnect = <{enabled|disabled}>]: enable or disable delayed disconnect feature </li></ul><ul><li>[delayeddisconnecttimer = <number{1-600}>]: delayed disconnect timer (in seconds) </li></ul><ul><li>[ringmuteduration = <number{0-60000}>]: early media mute duration (in minutes) </li></ul>
  26. 27. SIP Configuration: Overview <ul><li>:voice sip config </li></ul><ul><li>UserAgent domain : thomsongateway.sip </li></ul><ul><li>Primary proxy address : voip.thomson.be:5060 </li></ul><ul><li>Secondary proxy address : 0.0.0.0:5060  Not supported </li></ul><ul><li>Primary registrar address : voip.Thomson Gateway.be:5060 </li></ul><ul><li>Secondary registrar address : 0.0.0.0:5060  Not supported </li></ul><ul><li>Listening port : 5060 </li></ul><ul><li>Expire time : 3600 </li></ul><ul><li>Expire time delta : 1 </li></ul><ul><li>Notifier address : 0.0.0.0:5060 </li></ul><ul><li>Subscribe expire time : 3600 </li></ul>
  27. 28. SIP Configuration: Overview – Continued <ul><li>:voice sip config </li></ul><ul><li>Call Waiting reply : 182 </li></ul><ul><li>Transport : UDP </li></ul><ul><li>rtpmapstaticPT : Disabled </li></ul><ul><li>reinvite_stop_audio : Disabled </li></ul><ul><li>PRACK : Disabled </li></ul><ul><li>Clir format : standard </li></ul><ul><li>DTMF */# in INFO method : 1011 </li></ul><ul><li>Clip consider displayname : yes </li></ul><ul><li>SDP packet time : 20 </li></ul><ul><li>Replace # : Enabled </li></ul><ul><li>Symmetric codec : Enabled </li></ul><ul><li>Reinvite at calling fax detect : Disabled </li></ul><ul><li>SIPURI port : Enabled </li></ul><ul><li>rport : Disabled </li></ul><ul><li>SDP username : 780 </li></ul><ul><li>ringtoneat183 : Disabled </li></ul><ul><li>T38 Port increment : 0 </li></ul><ul><li>Ping timer : 0 </li></ul><ul><li>Min SE timer : 90 </li></ul><ul><li>Session expires timer : 120 </li></ul><ul><li>Expires timer : 0 </li></ul>
  28. 29. :voice Profile – Create VoIP User <ul><li>Voice profile add </li></ul><ul><ul><li>SIP_URI = <string>  telephone number </li></ul></ul><ul><ul><ul><li>SIP URI related to voice port [username = <string>] </li></ul></ul></ul><ul><ul><li>Authentication username related to voice port [password = <password>] </li></ul></ul><ul><ul><li>Authentication password related to voice port [displayname = <string>]  CLIP info </li></ul></ul><ul><ul><li>Alias name for SIP_URI voiceport = <{FXS1|FXS2|COMMON}>  available ports on TG </li></ul></ul><ul><ul><li>Analogue line number [abbr = <string>] </li></ul></ul><ul><li>Abbreviated number mapped to SIP_URI </li></ul><ul><ul><ul><li>abbr. number only supported when URI has NO LETTERS, only numbers </li></ul></ul></ul><ul><li>=>voice profile list SIP_URI all </li></ul><ul><li>Port Uri DisplayName Username Abbr Nbr RegStatus Msg Waiting </li></ul><ul><li>--------------------------------------------------------------------------------------------- </li></ul><ul><li>COMMON 2585 2585 2585 85 Registered No </li></ul>
  29. 30. :voice Country <ul><li>=>voice country config country=belgium </li></ul><ul><ul><li>Pre-loaded country settings </li></ul></ul><ul><ul><ul><li>Country = australia | belgium | denmark | etsi | france1 | france2 | france3 |germany | italy | netherlands | northamerica | norway | spain | sweden | uk </li></ul></ul></ul><ul><li>Country specific settings: </li></ul><ul><ul><li>DTMF tones / dial tones / Hook flash timer / polarity / etc. </li></ul></ul><ul><ul><li>Can be changed to specific needs </li></ul></ul><ul><ul><li>Special file in dl-directory: vincfg.bin </li></ul></ul>
  30. 31. :voice Cac <ul><li>=>:voice cac help config </li></ul><ul><ul><li>[max#portsperprofile = <{one|all}>] </li></ul></ul><ul><li>Maximum number of ports that can be used with common profile </li></ul><ul><ul><li>One: only 1 call with common number possible at same time </li></ul></ul><ul><ul><li>All: multiple calls with common number possible at same time </li></ul></ul>
  31. 32. SIP Services
  32. 33. Local Services: 3 Port Call <ul><li>What is needed for 3-way conference call using VoIP? </li></ul>Voice Network hold R+2 3way R+3 switch R+2 <ul><li>Answer or make call </li></ul><ul><li>Put 1 st call on hold </li></ul><ul><li>Make 2 nd call </li></ul><ul><li>Switch between calls or put in 3-way conference call </li></ul>Services available?
  33. 34. Supplementary Services SpeedTouch 6.1 <ul><li>Transfer: Call Transfer between local ports </li></ul><ul><li>Hold: put active Call on Hold </li></ul><ul><li>Waiting: incoming call while active call indication </li></ul><ul><li>Mwi: Message Waiting Indication </li></ul><ul><li>Clip: Calling Line Identification Presentation </li></ul><ul><li>Clir: Calling Line Identification Restriction </li></ul><ul><li>3pty: Three Party Call </li></ul>
  34. 35. Supplementary Services SpeedTouch 6.1 – Continued <ul><li>forcedFXO: switch to FXO (PSTN) </li></ul><ul><li>Cfu: Call Forwarding Unconditional </li></ul><ul><li>Cfnr: Call Forwarding on No Reply </li></ul><ul><li>Cfbs: Call Forwarding on Busy </li></ul><ul><li>Ccbs: Call Completion on Busy Subscriber </li></ul><ul><li>Clironcall: CLIR for only one call </li></ul><ul><li>Waitingoncall: Call Waiting active for only one call </li></ul>
  35. 36. Codec Use
  36. 37. 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
  37. 38. Advanced Voice Routing
  38. 39. Advanced Routed Scenario <ul><li>Multiple routed interfaces </li></ul><ul><li>How VoIP interface binding? </li></ul><ul><li>VoIP routing-based </li></ul><ul><ul><li>Best route used </li></ul></ul><ul><ul><li>SIP SoftSwitch is known IP-address </li></ul></ul><ul><ul><li>UA and RTP? </li></ul></ul><ul><ul><li>No IP-address known </li></ul></ul><ul><ul><li>All VoIP GW configured in routing -> complex </li></ul></ul>
  39. 40. 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
  40. 41. VoIP Interface Specific <ul><li>Source IP address selection (on interface) </li></ul><ul><ul><li>:voip config static_intf=enabled intf=voip_ip_intf </li></ul></ul><ul><li>VoIP label for RTP routing </li></ul><ul><ul><li>RTP via SIP-ALG and label inheritance </li></ul></ul><ul><ul><li>(default 7.4) </li></ul></ul><ul><li>=>connection bindlist </li></ul><ul><li>Application Proto Portrange Flags </li></ul><ul><li>SIP udp 5060 SIP_ALG:E RTP_predict_for_term_SIP_ALG:E </li></ul>{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
  41. 42. Add Label Routing <ul><li>default QoS rule </li></ul><ul><li>:label rule add chain=qos_default_labels index=3 serv=sip log=disabled state=enabled label=VoIP </li></ul><ul><li>to be added for voice routing: </li></ul><ul><li>:label add Voip_only inheritance=enabled </li></ul><ul><li>:label rule add chain=rt_user_labels index=1 srcintf=local serv=sip log=disabled state=enabled label=Voip_only </li></ul><ul><li>Label rule instructs that all CPE SIP traffic has to use this label </li></ul>
  42. 43. Label Routing / Forwarding <ul><li>Static IP-address :ip rtadd dst 0.0.0.0/0 label=Voip_only gateway=123.123.123.123 </li></ul><ul><li>PPP VoIP interface :ppp rtadd intf=Internet dst=0.0.0.0/0 label=Voip_only </li></ul><ul><li>DHCP-Client VoIP interface :dhcp client ifconfig intf=voip_ip_intf label=Voip_only gateway=enabled </li></ul>
  43. 44. Debug VoIP
  44. 45. Tips and Tricks <ul><li>Services not working </li></ul><ul><ul><li>No hookflash detection etc.? </li></ul></ul><ul><ul><ul><li>Change country to etsi or northamerica </li></ul></ul></ul><ul><ul><ul><li>Different analogue phone settings / timers </li></ul></ul></ul><ul><li>One way voice </li></ul><ul><ul><li>Codec priority changes </li></ul></ul><ul><ul><ul><li>Enable / disable codecs </li></ul></ul></ul><ul><ul><li>Ptime changes </li></ul></ul><ul><ul><ul><li>Ptime acceptable for Gateway? </li></ul></ul></ul><ul><ul><li>Connection bindlist </li></ul></ul><ul><ul><ul><li>SIP-ALG bounded? </li></ul></ul></ul><ul><ul><li>Symmetrical codec </li></ul></ul><ul><ul><ul><li>Enabled for 7G? </li></ul></ul></ul>
  45. 46. Debug VoIP: Standard Traces <ul><li>Ctrl-q -> start debug </li></ul><ul><li>Ctrl-s -> stop debug </li></ul><ul><li>Ctrl-t -> clear buffer </li></ul><ul><li>Ethereal trace has VoIP flow </li></ul><ul><li>Statistics : VoIP Call </li></ul><ul><li>Additional voice traces </li></ul><ul><li>Enable -> :voice debug exec cmd=“trace 1” </li></ul><ul><li>Disable -> :voice debug exec cmd=“trace 0” </li></ul>
  46. 47. Hands on - debugging <ul><li>Install and configure x-lite (SoftPhone) </li></ul>
  47. 48. Debugging
  48. 49. Debugging
  49. 50. Debugging <ul><li>200 OK is received by the CPE but not forwarded to the Computer running x-lite </li></ul>
  50. 51. Debugging <ul><li>[IN]LocalNet-> : 192.168.1.64 192.168.0.101 0583 UDP 62748->5080 </li></ul><ul><li>[UT]LocalNet->ip_voice : 192.168.0.104 192.168.0.101 0583 UDP 49252->5080 </li></ul><ul><li>[IN] ip_voice-> : 192.168.0.101 192.168.0.104 0443 UDP 5080->62748 </li></ul><ul><li>[DR] ip_voice->ip_voice : 192.168.0.101 192.168.0.104 0443 UDP 5080->62748 </li></ul><ul><li>: error caused by NAT-INPUT </li></ul><ul><li>The 200 OK is dropped because the packet is received on the wrong port </li></ul><ul><ul><li>Why does the SIPServer reply on 62748? </li></ul></ul>
  51. 52. Solving <ul><li>The data sent to the SIPserver is wrong obviously because of NAT. </li></ul><ul><li>The ALG should modify the content and generate the appropriated NAT entry. </li></ul><ul><li>The SIP ALG is bound to the wrong Port </li></ul>{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

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