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Scaling Asterisk Installations ,[object Object],Copyright Edvina AB, Sollentuna, Sweden 2010. All rights reserved.
Agenda ,[object Object],[object Object],[object Object],[object Object],[object Object],[object Object],[object Object],Many slides copied from the  Asterisk SIP Masterclass ,[object Object],[object Object],[object Object],[object Object],[object Object],[object Object],[object Object],[object Object],[object Object],[object Object],[object Object],[object Object],[object Object],[object Object],[object Object]
Agenda ,[object Object],[object Object],[object Object],[object Object],[object Object],Many slides copied from the  Asterisk SIP Masterclass
Disclaimer ,[object Object],[object Object],[object Object]
The big question What do you want to do today?
[object Object],Audio Video Text White-board 3D Hologr ????
[object Object]
#1 : The Internet. ,[object Object],[object Object],[object Object],[object Object]
#2: Open Standards making it possible making it possible ,[object Object],[object Object],[object Object],[object Object],[object Object]
#3: Open Source ,[object Object],[object Object],[object Object],[object Object]
#4: Voice interfaces ,[object Object],[object Object],[object Object]
#5: Commodity hardware ,[object Object],[object Object]
#6: API’s ,[object Object],[object Object],[object Object],[object Object],[object Object]
#7: Unified, federated identity ,[object Object],LDAP OpenID SAML 2.0 Oauth
In order to follow this ,[object Object]
Conclusion ,[object Object]
Consider If you invest in a unified communication network today and design it for voice/PBX services only,  it won’t work for future services
Without the Internet you still have  an old telephony PBX. an old telephony PBX. an old telephony PBX.
Without the Internet you are on an isolated island. isolated island. isolated island. So how do we get away from it?
Where are we today? ,[object Object],[object Object],[object Object],[object Object]
How do we solve this? ,[object Object],[object Object]
Kamailio & Asterisk Session Border controller Proxy Core proxy Proxy Registrar/Location server Asterisk PSTN Gateway Asterisk Voicemail Server Asterisk Feature/IVR Server Asterisk Proxy
Scaling using SIP ,[object Object],[object Object],[object Object],[object Object],[object Object],[object Object]
The problem with Kamailio ,[object Object],[object Object]
A proxy SIP call and an Asterisk SIP call AUDIO STREAM AUDIO STREAMS One SIP call (dialog) One SIP call (dialog) from Alice to Asterisk One SIP call (dialog) from Asterisk to Bob Alice Bob Two SIP calls, and two media streams SIP Proxy Asterisk
SIP proxy & b2bua working together ,[object Object]
Let’s play together Kamailio Proxy Asterisk PBX Call stateful Transaction stateful *
Asterisk’s view of SIP
Where we should be going... SIP SIP SIP Asterisk interoperating with a SIP infrastructure with full interoperability
SuperQuick Intro to Kamailio
A SIP proxy ,[object Object],[object Object],[object Object],[object Object]
Kamailio ,[object Object],[object Object],[object Object],[object Object],[object Object]
Step 1 ,[object Object]
Simple scenario ,[object Object],[object Object],[object Object],[object Object],User database *
Changes from Asterisk PoV ,[object Object],[object Object],[object Object],[object Object],[object Object]
Possible next step ,[object Object],[object Object],[object Object],* *
Stage 2 ,[object Object]
Simple scenario User database ,[object Object],[object Object],[object Object],*
Changes ,[object Object],[object Object],[object Object],[object Object],[object Object],[object Object],[object Object],[object Object],[object Object],[object Object]
Possible next step User database ,[object Object],[object Object],[object Object],* *
Issues ,[object Object],[object Object],[object Object],[object Object]
Using Realtime ,[object Object]
Realtime in SIP ,[object Object],[object Object],[object Object],[object Object],Proxy with load balancing * * * *
Realtime in SIP with Dundi Dundi cluster SIP Database Server Calls and registrations are distributed with a SIP proxy or DNS or... * * * *
Realtime in SIP with SysName Database Server Calls and registrations are distributed with DNS The DB contains the SystemName of the server that accepted registration for outbound calls (NAT traversal) * * * *
Pro’s and con’s ,[object Object],[object Object]
Realtime hybrid
Sharing data ,[object Object],[object Object],[object Object],User database *
PBX hosting
Requirements ,[object Object],[object Object],[object Object]
PBX hosting ,[object Object],[object Object],[object Object],Pbx customer 1 Pbx customer 2 Pbx customer 3 Voicemail, conferencing services * * * *
What you can do ,[object Object],[object Object],[object Object],Pbx customer 1 Pbx customer 2 Standby Voicemail, conferencing services * * * * *
Configuration
Configuring a peer for the proxy [mypeer] type=peer host=mypeer.mydomain.mytld context=mypeer-in With this configuration, we can place calls and receive calls. Asterisk will match on IP for incoming calls and send outbound calls to the proxy.
Receiving calls ,[object Object],[object Object],sip: 123456 @ kamailioguru.org ${EXTEN} exten=> 123456,1,noop ${SIPDOMAIN} [kamailio_in] exten =>_X.,1,goto(${SIPDOMAIN},${EXTEN},1) [kamailioguru.org] exten => _X.,1,verbose(hello ${EXTEN})
Sending calls [kamailio] type=peer host=mypeer.mydomain.mytld context=mypeer-in exten => 12345,1,dial(SIP/12345@kamailio) INVITE sip:12345@mypeer.mydomain.mytld To: sip:12345@mypeer.mydomain.mytld
Sending calls (2) [kamailio] type=peer host=mypeer.mydomain.mytld context=mypeer-in exten => 12345,1,dial(SIP/12345@kamailio! 0812346 ) INVITE sip:12345@mypeer.mydomain.mytld To: sip:  0812346 @mypeer.mydomain.mytld From: sip:yyyy@asterisk
Sending calls (2) [kamailio] type=peer host=mypeer.mydomain.mytld context=mypeer-in exten => 12345,1,dial(SIP/12345@kamailio! [email_address] ) INVITE sip:12345@mypeer.mydomain.mytld To: sip:  0812346 @kamailio.org From: sip:yyyy@asterisk
SBC behaviour ,[object Object],[object Object],[object Object],[object Object],[object Object],[object Object],[object Object]
Example ; Get the full FROM Uri ; The username part is already the caller ID exten => _.,1,set(fromuri=${SIPCHANINFO(from)}) ; Get the domain and set it for the outbound call ; Note! This requires a patch. Exist in 1.6.2 and later. exten => _.,n,set(SIPFROMDOMAIN=${CUT(fromuri,@,2}) ; Dial out. The ! part only exist in 1.6.0 and later exten => _.,n,dial(SIP/${EXTEN}@${SIPDOMAIN}!${SIP_HEADER(To)}) The outbound proxy setting will route calls to the proxy
Asterisk as PSTN gateway, not pbx ,[object Object],[object Object],[object Object],[object Object]
Asterisk media server
Playing prompts ; Don’t answer immediately, that confuses people exten => invalidnumber,1,ringing() exten => invalidnumber,n,wait(1) exten => invalidnumber,n,answer ; Let the RTP streams settle exten => invalidnumber,n,wait(.5) exten => invalidnumber,n,playback(invalid-number) ; Let the RTP streams settle exten => invalidnumber,n,wait(.5) exten => invalidnumber,n,hangup
Future and current work
Pinetree ,[object Object],[object Object],[object Object],Pbx customer 1 Pbx customer 2 Standby * * *
Appleraisin ,[object Object],[object Object],[object Object],MySQL MySQL * *
PineFrog ,[object Object],[object Object],[object Object],[object Object],[object Object],[object Object],[object Object]
Pinana ,[object Object],[object Object],[object Object]
Other branches ,[object Object],[object Object],[object Object],[object Object],[object Object],[object Object],[object Object],[object Object]
Summary
Migration is hard ,[object Object],[object Object],[object Object],[object Object]
Building a new platform ,[object Object],[object Object],[object Object],[object Object],[object Object],[object Object],[object Object],[object Object]
One last issue. ,[object Object]
Why are we still building with PSTN focus? PSTN SIP SIP SIP SIP SIP SIP Isolated islands of SIP!
Put the Internet where it belongs @ SIP SIP SIP SIP SIP SIP One Unified Communications Network PSTN PSTN PSTN PSTN PSTN PSTN
IPv6 is the new glue No NAT.  Enough network addresses to build anything you need.  Go and have fun. Discover the new oppurtunities with IPv6!
Questions? ?
© Copyright Edvina AB, Sollentuna, Sweden 2010. All rights reserved. Open Unified Communication Connecting Internet Users in real time.

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Astricon 2010: Scaling Asterisk installations

  • 1.
  • 2.
  • 3.
  • 4.
  • 5. The big question What do you want to do today?
  • 6.
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  • 17. Consider If you invest in a unified communication network today and design it for voice/PBX services only, it won’t work for future services
  • 18. Without the Internet you still have an old telephony PBX. an old telephony PBX. an old telephony PBX.
  • 19. Without the Internet you are on an isolated island. isolated island. isolated island. So how do we get away from it?
  • 20.
  • 21.
  • 22. Kamailio & Asterisk Session Border controller Proxy Core proxy Proxy Registrar/Location server Asterisk PSTN Gateway Asterisk Voicemail Server Asterisk Feature/IVR Server Asterisk Proxy
  • 23.
  • 24.
  • 25. A proxy SIP call and an Asterisk SIP call AUDIO STREAM AUDIO STREAMS One SIP call (dialog) One SIP call (dialog) from Alice to Asterisk One SIP call (dialog) from Asterisk to Bob Alice Bob Two SIP calls, and two media streams SIP Proxy Asterisk
  • 26.
  • 27. Let’s play together Kamailio Proxy Asterisk PBX Call stateful Transaction stateful *
  • 29. Where we should be going... SIP SIP SIP Asterisk interoperating with a SIP infrastructure with full interoperability
  • 31.
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  • 44. Realtime in SIP with Dundi Dundi cluster SIP Database Server Calls and registrations are distributed with a SIP proxy or DNS or... * * * *
  • 45. Realtime in SIP with SysName Database Server Calls and registrations are distributed with DNS The DB contains the SystemName of the server that accepted registration for outbound calls (NAT traversal) * * * *
  • 46.
  • 48.
  • 50.
  • 51.
  • 52.
  • 54. Configuring a peer for the proxy [mypeer] type=peer host=mypeer.mydomain.mytld context=mypeer-in With this configuration, we can place calls and receive calls. Asterisk will match on IP for incoming calls and send outbound calls to the proxy.
  • 55.
  • 56. Sending calls [kamailio] type=peer host=mypeer.mydomain.mytld context=mypeer-in exten => 12345,1,dial(SIP/12345@kamailio) INVITE sip:12345@mypeer.mydomain.mytld To: sip:12345@mypeer.mydomain.mytld
  • 57. Sending calls (2) [kamailio] type=peer host=mypeer.mydomain.mytld context=mypeer-in exten => 12345,1,dial(SIP/12345@kamailio! 0812346 ) INVITE sip:12345@mypeer.mydomain.mytld To: sip: 0812346 @mypeer.mydomain.mytld From: sip:yyyy@asterisk
  • 58. Sending calls (2) [kamailio] type=peer host=mypeer.mydomain.mytld context=mypeer-in exten => 12345,1,dial(SIP/12345@kamailio! [email_address] ) INVITE sip:12345@mypeer.mydomain.mytld To: sip: 0812346 @kamailio.org From: sip:yyyy@asterisk
  • 59.
  • 60. Example ; Get the full FROM Uri ; The username part is already the caller ID exten => _.,1,set(fromuri=${SIPCHANINFO(from)}) ; Get the domain and set it for the outbound call ; Note! This requires a patch. Exist in 1.6.2 and later. exten => _.,n,set(SIPFROMDOMAIN=${CUT(fromuri,@,2}) ; Dial out. The ! part only exist in 1.6.0 and later exten => _.,n,dial(SIP/${EXTEN}@${SIPDOMAIN}!${SIP_HEADER(To)}) The outbound proxy setting will route calls to the proxy
  • 61.
  • 63. Playing prompts ; Don’t answer immediately, that confuses people exten => invalidnumber,1,ringing() exten => invalidnumber,n,wait(1) exten => invalidnumber,n,answer ; Let the RTP streams settle exten => invalidnumber,n,wait(.5) exten => invalidnumber,n,playback(invalid-number) ; Let the RTP streams settle exten => invalidnumber,n,wait(.5) exten => invalidnumber,n,hangup
  • 65.
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  • 73.
  • 74. Why are we still building with PSTN focus? PSTN SIP SIP SIP SIP SIP SIP Isolated islands of SIP!
  • 75. Put the Internet where it belongs @ SIP SIP SIP SIP SIP SIP One Unified Communications Network PSTN PSTN PSTN PSTN PSTN PSTN
  • 76. IPv6 is the new glue No NAT. Enough network addresses to build anything you need. Go and have fun. Discover the new oppurtunities with IPv6!
  • 78. © Copyright Edvina AB, Sollentuna, Sweden 2010. All rights reserved. Open Unified Communication Connecting Internet Users in real time.