Loadbalancing SIP and RTP

Hi
I want to loadbalance two asterisk boxes with a CSM-S. I'm using Direct Server return (no nat server), and it works fine for any tcp based service, but not for UDP, such as SIP or RTP. Can anybody help me? Is this kind of configuration not possible with Direct Server Return?
Best regards
Simon

Hi Gilles
Tanks for your response, i did the sniff and I found the fault immediately, I made a typo with an IP address!!!!!!! Now it works perfectly fine!
Thanks to ethereal!
Simon

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    Hi Aok
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    From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
    To: ;tag=125E62C-1354
    Date: Tue, 05 Mar 2013 13:52:31 GMT
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    Expires: 180
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    Supported: Geolocation
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    Content-Length: 244
    v=0
    o=CiscoSystemsCCM-SIP 38874 2 IN IP4 192.168.1.241
    s=SIP Call
    c=IN IP4 0.0.0.0
    b=TIAS:64000
    b=AS:64
    t=0 0
    m=audio 24784 RTP/AVP 8 101
    a=rtpmap:8 PCMA/8000
    a=ptime:20
    a=inactive
    a=rtpmap:101 telephone-event/8000
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    Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK355253C
    From: [email protected]>;tag=125E594-5C7
    To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
    Date: Tue, 05 Mar 2013 14:19:57 GMT
    Call-ID: [email protected]
    Route:
    Supported: 100rel,timer,resource-priority,replaces,sdp-anat
    Min-SE:  1800
    Cisco-Guid: 3698592896-0000065536-0000000107-4043417792
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 102 INVITE
    Max-Forwards: 70
    Timestamp: 1362493197
    Contact:
    Expires: 60
    Allow-Events: telephone-event
    Content-Type: application/sdp
    Content-Length: 262
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 6506 3807 IN IP4 10.249.13.130
    s=SIP Call
    c=IN IP4 10.249.13.130
    t=0 0
    m=audio 19234 RTP/AVP 8 101
    c=IN IP4 10.249.13.130
    a=inactive
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:20
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    Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK79816bbd4196
    From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
    To: ;tag=125E62C-1354
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    Session-Expires: 1800;refresher=uac
    Content-Type: application/sdp
    Content-Length: 213
    Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE, INFO
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    t=0 0
    m=audio 54932 RTP/AVP 8 101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:20
    a=inactive
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    SIP/2.0 200 OK
    Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK79816bbd4196
    From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
    To: ;tag=125E62C-1354
    Date: Tue, 05 Mar 2013 14:19:57 GMT
    Call-ID: [email protected]
    CSeq: 102 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    Allow-Events: telephone-event
    Contact:
    Supported: replaces
    Supported: sdp-anat
    Server: Cisco-SIPGateway/IOS-12.x
    Session-Expires:  1800;refresher=uac
    Require: timer
    Supported: timer
    Content-Type: application/sdp
    Content-Length: 259
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 9410 5774 IN IP4 192.168.1.10
    s=SIP Call
    c=IN IP4 192.168.1.10
    t=0 0
    m=audio 19314 RTP/AVP 8 101
    c=IN IP4 192.168.1.10
    a=inactive
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:20
    *Mar  5 14:19:57.946: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
    Sent:
    ACK sip:[email protected]:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK3562A4
    From: [email protected]>;tag=125E594-5C7
    To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
    Date: Tue, 05 Mar 2013 14:19:57 GMT
    Call-ID: [email protected]
    Route:
    Max-Forwards: 70
    CSeq: 102 ACK
    Allow-Events: telephone-event
    Content-Length: 0
    *Mar  5 14:19:57.946: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    ACK sip:[email protected]:5060;transport=tcp SIP/2.0
    Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK798246ab3597
    From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
    To: ;tag=125E62C-1354
    Date: Tue, 05 Mar 2013 13:52:31 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 102 ACK
    Allow-Events: presence
    Content-Length: 0
    *Mar  5 14:19:58.146: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    INVITE sip:[email protected]:5060;transport=tcp SIP/2.0
    Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7983739137ab
    From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
    To: ;tag=125E62C-1354
    Date: Tue, 05 Mar 2013 13:52:31 GMT
    Call-ID: [email protected]
    Supported: timer,resource-priority,replaces
    Min-SE:  1800
    User-Agent: Cisco-CUCM9.0
    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
    CSeq: 103 INVITE
    Max-Forwards: 70
    Expires: 180
    Allow-Events: presence
    Supported: X-cisco-srtp-fallback
    Supported: Geolocation
    Session-Expires:  1800;refresher=uac
    P-Asserted-Identity:
    Remote-Party-ID: ;party=calling;screen=yes;privacy=off
    Contact:
    Content-Length: 0
    *Mar  5 14:19:58.158: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
    Sent:
    INVITE sip:[email protected]:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK3571933
    From: [email protected]>;tag=125E594-5C7
    To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
    Date: Tue, 05 Mar 2013 14:19:58 GMT
    Call-ID: [email protected]
    Route:
    Supported: timer,resource-priority,replaces,sdp-anat
    Min-SE:  1800
    Cisco-Guid: 3698592896-0000065536-0000000107-4043417792
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 103 INVITE
    Max-Forwards: 70
    Timestamp: 1362493198
    Contact:
    Expires: 60
    Allow-Events: telephone-event
    Content-Length: 0
    *Mar  5 14:19:58.158: //716/DC740C800000/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 100 Trying
    Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7983739137ab
    From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
    To: ;tag=125E62C-1354
    Date: Tue, 05 Mar 2013 14:19:58 GMT
    Call-ID: [email protected]
    CSeq: 103 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-12.x
    Content-Length: 0
    *Mar  5 14:19:58.218: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
    Received:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK3571933
    To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
    From: [email protected]>;tag=125E594-5C7
    Call-ID: [email protected]
    CSeq: 103 INVITE
    Contact:
    Require: timer
    Session-Expires: 1800;refresher=uac
    Content-Type: application/sdp
    Content-Length: 216
    Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE, INFO
    Accept: application/media_control+xml
    Accept: application/sdp
    Accept: application/x-broadworks-call-center+xml
    v=0
    o=BroadWorks 96335268 3 IN IP4 10.224.42.164
    s=-
    c=IN IP4 10.224.42.72
    t=0 0
    m=audio 54932 RTP/AVP 8 18 96 99
    a=rtpmap:96 AMR/8000
    a=rtpmap:99 telephone-event/8000
    a=fmtp:99 0-15
    a=ptime:20
    a=sendrecv
    *Mar  5 14:19:58.234: //716/DC740C800000/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 200 OK
    Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7983739137ab
    From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
    To: ;tag=125E62C-1354
    Date: Tue, 05 Mar 2013 14:19:58 GMT
    Call-ID: [email protected]
    CSeq: 103 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    Allow-Events: telephone-event
    Contact:
    Supported: replaces
    Supported: sdp-anat
    Server: Cisco-SIPGateway/IOS-12.x
    Session-Expires:  1800;refresher=uac
    Require: timer
    Supported: timer
    Content-Type: application/sdp
    Content-Length: 283
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 9410 5775 IN IP4 192.168.1.10
    s=SIP Call
    c=IN IP4 192.168.1.10
    t=0 0
    m=audio 19314 RTP/AVP 8 18 101
    c=IN IP4 192.168.1.10
    a=rtpmap:8 PCMA/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    *Mar  5 14:19:58.242: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    ACK sip:[email protected]:5060;transport=tcp SIP/2.0
    Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7985648033f2
    From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
    To: ;tag=125E62C-1354
    Date: Tue, 05 Mar 2013 13:52:31 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 103 ACK
    Allow-Events: presence
    Content-Type: application/sdp
    Content-Length: 192
    v=0
    o=CiscoSystemsCCM-SIP 38874 3 IN IP4 192.168.1.241
    s=SIP Call
    c=IN IP4 192.168.1.241
    t=0 0
    m=audio 4000 RTP/AVP 8
    a=X-cisco-media:umoh
    a=rtpmap:8 PCMA/8000
    a=ptime:20
    a=sendonly
    *Mar  5 14:19:58.262: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
    Sent:
    ACK sip:[email protected]:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK358582
    From: [email protected]>;tag=125E594-5C7
    To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
    Date: Tue, 05 Mar 2013 14:19:58 GMT
    Call-ID: [email protected]
    Route:
    Max-Forwards: 70
    CSeq: 103 ACK
    Allow-Events: telephone-event
    Content-Type: application/sdp
    Content-Length: 259
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 6506 3808 IN IP4 10.249.13.130
    s=SIP Call
    c=IN IP4 10.249.13.130
    t=0 0
    m=audio 19234 RTP/AVP 8 99
    c=IN IP4 10.249.13.130
    a=sendonly
    a=rtpmap:8 PCMA/8000
    a=rtpmap:99 telephone-event/8000
    a=fmtp:99 0-15
    a=ptime:20
    SIP-GW#
    SIP-GW#sh voip rtp connections
    VoIP RTP active connections :
    No. CallId     dstCallId  LocalRTP RmtRTP     LocalIP                                RemoteIP
    1     716        717        19314    4000     192.168.1.10                           192.168.1.241
    2     717        716        19234    54932    10.249.13.130                          10.224.42.72
    Found 2 active RTP connections

  • N8; How to set up SIP and make (WiFI & 3G)VOIP cal...

    N8 only comes with SIP software and you need to download the the VoIP setting software at this link (use VoIP 3.1):
    http://www.forum.nokia.com/info/sw.nokia.com/id/d476061e-90ca-42e9-b3ea-1a852f3808ec/SIP_VoIP_Settin... 
    You can use the N8 web browser and load directly from the site to the phone.
    then go to:
    Menu>settings>connectivity>Admin.Settings>SIP settings
    option>New SIP Profile>use default profile.
    This window needs to be configured with your VoIP provider setting which may be found on their site or just do a search. I used the settings from my previous phone. I have Gizmo VoIP and will be using the setting for this service as an example to set up Wifi and 3G SIP setting
    Profile Name> Gizmo, (assign the name of your provider e.g.. Gizmo, Google talk, etc)
    Service profile> IETF
    Default Destination> WiFi
    Public username: sip:[email protected] (Input your VoIP provider info)
    Use compression> No
    Registration> When needed
    Use security> No
    Then goto Proxy server:
    Proxy server address> siproxy01.sipphone.com (use your provider proxy address setting)
    Realm> proxy01.sipphone.com (use your provider setting)
    Username> (use your username)
    Password> (enter your password0
    Allow loose routing> Yes
    Transport type> UDP
    Port> 5060
    Go back and then go to Registrar server:
    Registrar server address> siproxy01.sipphone.com (use your provider Registrar setting)
    Real> proxy01.sipphone.com (use your provider setting)
    Username> (use your username)
    Password> (enter password)
    Transport type> UDP
    Port> 5060
    Again folks above is just an example. You must use your provider settings.
    Then when you are done go back to:
     Admin setting> Net setting>Advance VoIP setting>Create new service> choose the SIP profile shown, e.g.  Gizmo 
    This will configure the VoIP setting with the particular SIP profile. It will also add a SIP Tel profile Pane in your Contacts folder, Internet call option to the contacts list and Phone log.
    Now go to the SIP pane in Contacts folder, sign in and then make a test call . To make a SIP call just go to a contact and choose Internet call. If you configured everything correctly, it should work and you are done.
    For 3G connection, follow above direction and create new SIP profile with new name (e.g. Gizmo (3G). All settings are same except:
    Default destination>Internet
    Then set up the VoIP setting with the new SIP profile by going to:
    Admin setting>Net setting>Advance VoIP settings>Create new service> Choose the SIP profile you just configured e.g. Gizmo (3G)
    Then:
    Admin setting>Net setting>Advance VoIP settings>VoIP service>choose the 3G SIP profile, e.g. Gizmo (3G)>Profile setting>>>>>
    Go down to AWCDMA> set it to "on"
    Exit out of everything. Go to SIP (3G) pane in the Contacts folder. Sign in and make test call.
    I have configured many Mobile phones and found the N8 somewhat touchy. But, the sound quality on Gizmo VoIP using both WiFi and 3G was great, no echo or any problems.
    Useful hints:
    If it does not connect or make a call, you need to check the SIP setting for any error. If you find any problem there, you must first remove the SIP profile in the Net setting. Then go to the SIP setting and correct the problem. Double check everyting, go back to Net setting>Advance VoIP setting>Create new service and add the SIP profile. The reason is that when you Configure a VoIP setting with a SIP profile, the SIP info is permanently loaded and will not change even if you make any change in the SIP profile. So any time you change anything in the SIP profile (even the password) you always need to remove, make correction and then reload the corrected SIP profile to the VoIP setting.
    There are also few Mobile service providers who have blocked VoIP on their data network service. So, just check with your service provider. In US both AT&T and TMobile have open network for VoIP.

    Hi
    I have an unbranded N8 and am trying to get VoipCheap to work.
    I have tried to set up VOIP but still facing a problem or 2 so please assist me.
    I have an active account via VoipCheap.com with a profile name and password and it is verified.
    1/ When I look in Admin Settings / Sip Settings it shows VoipCheap with "not registered" under it..why?
    My settings when I click on VoipCheap are:
    profile name: VoipCheap
    Service profile: IETF
    Default destination: Internet
    Public username: sip:[email protected]
    Use compression: No
    Registration: When needed
    Use security: No
    Proxy server address: sip:sip.VoipCheap.com
    Realm: VoipCheap.com
    Username: [email protected]
    Password:  ****** (I have entered my password)
    Allow loose routing: Yes
    Transport type: UDP
    Port: 5060
    Registrar server settings same as Proxy server settings.
    I have also gone to Net settings / VoipCheap settings and see:
    Username:  sip:[email protected]
    Password: ***** (I have entered my password)
    Default service: No
    Service connectivity: Internet
    In Contacts I have a VoipCheap tab with the icon for internet calling and it says
    Sign in (example)
    but when I click on this it shows connecting (via my internet connection) but hangs there - it does not seem to connect............
    what do you think I am doing wrong, please ????????????????

  • Capture and RTP on a SIGNED applet.

    I am trying to build an applet that allows users to chat both ways. I want the user to install nothing (other than java) and i provide the libraries they need (including jmf). I have signed the applet to ease my troubles, but it seems they are still there. The basic goal: capture audio from two microphones, send the audio via RTP, hear audio on both sides.
    Even though I have a signed applet with all permissions, I lack permission to capture from applets? Aside from the fact that this seems absurd, it's frustrating. I am looking to find a way around this. I don't want the users to have to manually change the JMFRegistry or download a program to it.
    The better options :
    1) Is there a way to take advantage of the signed applet to alter the permissions in the program? (Seems very unlikely, but it would be the best).
    2) Is there a way to provide the permissions in the jmf.jar (including jmf.properties?) that I include? (Seems more unlikely than 1)
    3) What makes this more interesting (frustrating), I can use the Java Sound API to access the microphone and open an AudioInputStream. Can I plug this stream, or some other object from javasound, into the JMF somehow to send it via RTP?
    4) Other options?

    I found a way to use javasound instead of the jmf for the source audio (that's number 3 in my original post). So, i'm good for my problem.
    If you want to do mixing, javasound should have more support for mixing (though I've not done it). Check out this thread, it was very useful for me.
    http://forum.java.sun.com/thread.jspa?threadID=680525&start=0&tstart=0
    I might have spoken too soon... I am just getting static - but that might be related to audio formats.....
    Message was edited by:
    mortserg

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