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Jerry777

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Posts posted by Jerry777

  1. IPv6 SDP Support

    in VoIP Phones

    Posted

    Responses are sent from the socket which received the message. That part does not use the operating system's routing table.

     

    So what is the output from "route -A inet6"?

     

    Here is the output from the "route -A inet6" command:

    Kernel IPv6 routing table

    Destination Next Hop Flags Metric Ref Use Iface

    ::1/128 * U 0 10 2 lo

    fe80::204:23ff:fea7:d60a/128 * U 0 34856 2 lo

    fe80::204:8dff:feff:fffe/128 fe80::204:8dff:feff:fffe UC 0 1 1 eth1

    fe80::/64 * U 256 0 0 eth1

    ff00::/8 * U 256 0 0 eth1

     

    Best Regards

  2. IPv6 SDP Support

    in VoIP Phones

    Posted

    We are using Red Hat 3.4.6-3. I did find a way to get it to set the SDP address correctly, via the SystemAdministration|Ports|Rtp:*|BindtospecificIPaddress(IPv6) menu. I set this field to the workstation's IPv6 IP address and now that address is appearing in the SDP.

     

    However, in the SIP header portion of the INVITE I am still seeing 0.0.0.0 in the Contact and Via SIP headers. So I have not found a way to cause this to be populated with the valid IPv4 address.

     

    I presume the routing table is correct, because messages are sent/received correctly from/to the workstation's IPv6 address. It's only the SIP header content that has the issue.

     

    Best Regards

     

    Correction: Paragraph two of my last reply should have said "IPv6" instead of "IPv4", as follows:

     

    However, in the SIP header portion of the INVITE I am still seeing 0.0.0.0 in the Contact and Via SIP headers. So I have not found a way to cause this to be populated with the valid IPv6 address.

     

    Thanks

  3. IPv6 SDP Support

    in VoIP Phones

    Posted

    What does your routing table look like? The PBX typically insers a "0.0.0.0" in desperation because it cannot find a routing table entry. What OS are you on?

     

    We are using Red Hat 3.4.6-3. I did find a way to get it to set the SDP address correctly, via the SystemAdministration|Ports|Rtp:*|BindtospecificIPaddress(IPv6) menu. I set this field to the workstation's IPv6 IP address and now that address is appearing in the SDP.

     

    However, in the SIP header portion of the INVITE I am still seeing 0.0.0.0 in the Contact and Via SIP headers. So I have not found a way to cause this to be populated with the valid IPv4 address.

     

    I presume the routing table is correct, because messages are sent/received correctly from/to the workstation's IPv6 address. It's only the SIP header content that has the issue.

     

    Best Regards

  4. IPv6 SDP Support

    in VoIP Phones

    Posted

    Here is the pbxnsip log when I make a call from line 44 to 45 (both registered with IPv6 addresses). Notice that the pbxnsip server is sending SDP to the far-end with an IP6 address of 0.0.0.0. I can provide follow-on messages, if needed. Any idea why this is happening?

     

    Logfile

    Clear or Reload the log.

     

    [9] 2009/06/30 09:08:48: Remote site closed the connection

    [9] 2009/06/30 09:08:56: SIP Rx udp:[fe80::204:8dff:feff:fffe]:5060:

    INVITE sip:45@[fe80::204:23ff:fea7:d60a]:5060;user=phone SIP/2.0

    Via: SIP/2.0/UDP [fe80::204:8dff:feff:fffe];rport;branch=z9hG4bKN24Qa3ert8NSB

    Max-Forwards: 70

    From: Jerry's 44 <sip:44@[fe80::204:23ff:fea7:d60a]:5060;user=phone>;tag=jm9pcvga1BDyD

    To: <sip:45@[fe80::204:23ff:fea7:d60a]:5060>;user=phone

    Call-ID: 902f30db-dfe8-122c-a4a6-09ac0212a3be

    CSeq: 117054713 INVITE

    Contact: Jerry's 44 <sip:44@[fe80::204:8dff:feff:fffe]:5060;transport=udp>

    User-Agent: TONE COMMANDER 7810/fffffe

    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER

    Supported: replaces, callerid, 100rel, resource-priority, sdp-anat

    Allow-Events: talk, hold, refer, call-info, dialog, sla, include-session-description

    Privacy: none

    Content-Type: application/sdp

    Content-Disposition: session

    Content-Length: 266

    P-Early-Media: supported

    P-Preferred-Identity: <sip:44@[fe80::204:23ff:fea7:d60a]>

     

    v=0

    o=TC 722710983 722710983 IN IP6 fe80::204:8dff:feff:fffe

    s=session

    c=IN IP6 fe80::204:8dff:feff:fffe

    t=0 0

    m=audio 1760 RTP/AVP 0 18 4 101

    a=rtpmap:0 PCMU/8000

    a=ptime:20

    a=rtpmap:18 G729/8000

    a=ptime:20

    a=rtpmap:4 G723/8000

    a=rtpmap:101 telephone-event/8000/1

     

    [9] 2009/06/30 09:08:56: UDP: Opening socket on port 52870

    [9] 2009/06/30 09:08:56: UDP: Opening socket on port 52871

    [9] 2009/06/30 09:08:56: UDPv6: Opening socket on port 52870

    [9] 2009/06/30 09:08:56: UDPv6: Opening socket on port 52871

    [8] 2009/06/30 09:08:56: Could not find a trunk (1 trunks)

    [9] 2009/06/30 09:08:56: Resolve 20871: aaaa udp fe80::204:8dff:feff:fffe 5060

    [9] 2009/06/30 09:08:56: Resolve 20871: udp fe80::204:8dff:feff:fffe 5060

    [9] 2009/06/30 09:08:56: SIP Tx udp:[fe80::204:8dff:feff:fffe]:5060:

    SIP/2.0 100 Trying

    Via: SIP/2.0/UDP [fe80::204:8dff:feff:fffe];rport=5060;branch=z9hG4bKN24Qa3ert8NSB

    From: Jerry's 44 <sip:44@[fe80::204:23ff:fea7:d60a]:5060;user=phone>;tag=jm9pcvga1BDyD

    To: <sip:45@[fe80::204:23ff:fea7:d60a]:5060>;tag=1a053bc5c1;user=phone

    Call-ID: 902f30db-dfe8-122c-a4a6-09ac0212a3be

    CSeq: 117054713 INVITE

    Content-Length: 0

     

     

    [7] 2009/06/30 09:08:56: Set packet length to 20

    [7] 2009/06/30 09:08:56: Last message repeated 2 times

    [6] 2009/06/30 09:08:56: Sending RTP for 902f30db-dfe8-122c-a4a6-09ac0212a3be#1a053bc5c1 to [fe80::204:8dff:feff:fffe]:1760

    [7] 2009/06/30 09:08:56: Attendant: Calling extension 45

    [7] 2009/06/30 09:08:56: Cannot convert number 44 into global format

    [8] 2009/06/30 09:08:56: Play audio_moh/noise.wav

    [9] 2009/06/30 09:08:56: UDP: Opening socket on port 61172

    [9] 2009/06/30 09:08:56: UDP: Opening socket on port 61173

    [9] 2009/06/30 09:08:56: UDPv6: Opening socket on port 61172

    [9] 2009/06/30 09:08:56: UDPv6: Opening socket on port 61173

    [9] 2009/06/30 09:08:56: Using outbound proxy sip:[fe80::204:8dff:feff:ffe4]:5060;transport=udp because of flow-label

    [9] 2009/06/30 09:08:56: Resolve 20872: url sip:[fe80::204:8dff:feff:ffe4]:5060;transport=udp

    [9] 2009/06/30 09:08:56: Resolve 20872: a udp [fe80::204:8dff:feff:ffe4] 5060

    [9] 2009/06/30 09:08:56: Resolve 20872: udp [fe80::204:8dff:feff:ffe4] 5060

    [9] 2009/06/30 09:08:56: SIP Tx udp:[fe80::204:8dff:feff:ffe4]:5060:

    INVITE sip:45@[fe80::204:8dff:feff:ffe4]:5060;transport=udp SIP/2.0

    Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK-e1523731b2cb1200ab2fa761491d7109;rport

    From: "Fourty Four" <sip:44@pbx.company.com>;tag=1142770007

    To: "Fourty Five" <sip:45@pbx.company.com>

    Call-ID: c3852a9b@pbx

    CSeq: 9340 INVITE

    Max-Forwards: 70

    Contact: <sip:45@0.0.0.0:5060;transport=udp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

    Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/3.0.1.3023

    Alert-Info: <http://127.0.0.1/Bellcore-dr2>

    Content-Type: application/sdp

    Content-Length: 286

     

    v=0

    o=- 75511103 75511103 IN IP6 0.0.0.0

    s=-

    c=IN IP6 0.0.0.0

    t=0 0

    m=audio 61172 RTP/AVP 0 8 9 2 3 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:8 pcma/8000

    a=rtpmap:9 g722/8000

    a=rtpmap:2 g726-32/8000

    a=rtpmap:3 gsm/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=sendrecv

     

    Also, notice the pbxnsip server's contact address is wrong (0.0.0.0). It should be it's IPv6 address.

     

    Best Regards

  5. IPv6 SDP Support

    in VoIP Phones

    Posted

    That should be working beautifully. Do you see that the PBX sends the INVITE to a IPv6 address? Are you using DNS? Make sure that you have proper AAAA records set up... On log level 9 you can see how the PBX determines the destination address.

     

    Otherwise, make sure that the outbound proxy looks like this: "sip:[2001:db8::1234]:5060".

     

    Here is the pbxnsip log when I make a call from line 44 to 45 (both registered with IPv6 addresses). Notice that the pbxnsip server is sending SDP to the far-end with an IP6 address of 0.0.0.0. I can provide follow-on messages, if needed. Any idea why this is happening?

     

    Logfile

    Clear or Reload the log.

     

    [9] 2009/06/30 09:08:48: Remote site closed the connection

    [9] 2009/06/30 09:08:56: SIP Rx udp:[fe80::204:8dff:feff:fffe]:5060:

    INVITE sip:45@[fe80::204:23ff:fea7:d60a]:5060;user=phone SIP/2.0

    Via: SIP/2.0/UDP [fe80::204:8dff:feff:fffe];rport;branch=z9hG4bKN24Qa3ert8NSB

    Max-Forwards: 70

    From: Jerry's 44 <sip:44@[fe80::204:23ff:fea7:d60a]:5060;user=phone>;tag=jm9pcvga1BDyD

    To: <sip:45@[fe80::204:23ff:fea7:d60a]:5060>;user=phone

    Call-ID: 902f30db-dfe8-122c-a4a6-09ac0212a3be

    CSeq: 117054713 INVITE

    Contact: Jerry's 44 <sip:44@[fe80::204:8dff:feff:fffe]:5060;transport=udp>

    User-Agent: TONE COMMANDER 7810/fffffe

    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER

    Supported: replaces, callerid, 100rel, resource-priority, sdp-anat

    Allow-Events: talk, hold, refer, call-info, dialog, sla, include-session-description

    Privacy: none

    Content-Type: application/sdp

    Content-Disposition: session

    Content-Length: 266

    P-Early-Media: supported

    P-Preferred-Identity: <sip:44@[fe80::204:23ff:fea7:d60a]>

     

    v=0

    o=TC 722710983 722710983 IN IP6 fe80::204:8dff:feff:fffe

    s=session

    c=IN IP6 fe80::204:8dff:feff:fffe

    t=0 0

    m=audio 1760 RTP/AVP 0 18 4 101

    a=rtpmap:0 PCMU/8000

    a=ptime:20

    a=rtpmap:18 G729/8000

    a=ptime:20

    a=rtpmap:4 G723/8000

    a=rtpmap:101 telephone-event/8000/1

     

    [9] 2009/06/30 09:08:56: UDP: Opening socket on port 52870

    [9] 2009/06/30 09:08:56: UDP: Opening socket on port 52871

    [9] 2009/06/30 09:08:56: UDPv6: Opening socket on port 52870

    [9] 2009/06/30 09:08:56: UDPv6: Opening socket on port 52871

    [8] 2009/06/30 09:08:56: Could not find a trunk (1 trunks)

    [9] 2009/06/30 09:08:56: Resolve 20871: aaaa udp fe80::204:8dff:feff:fffe 5060

    [9] 2009/06/30 09:08:56: Resolve 20871: udp fe80::204:8dff:feff:fffe 5060

    [9] 2009/06/30 09:08:56: SIP Tx udp:[fe80::204:8dff:feff:fffe]:5060:

    SIP/2.0 100 Trying

    Via: SIP/2.0/UDP [fe80::204:8dff:feff:fffe];rport=5060;branch=z9hG4bKN24Qa3ert8NSB

    From: Jerry's 44 <sip:44@[fe80::204:23ff:fea7:d60a]:5060;user=phone>;tag=jm9pcvga1BDyD

    To: <sip:45@[fe80::204:23ff:fea7:d60a]:5060>;tag=1a053bc5c1;user=phone

    Call-ID: 902f30db-dfe8-122c-a4a6-09ac0212a3be

    CSeq: 117054713 INVITE

    Content-Length: 0

     

     

    [7] 2009/06/30 09:08:56: Set packet length to 20

    [7] 2009/06/30 09:08:56: Last message repeated 2 times

    [6] 2009/06/30 09:08:56: Sending RTP for 902f30db-dfe8-122c-a4a6-09ac0212a3be#1a053bc5c1 to [fe80::204:8dff:feff:fffe]:1760

    [7] 2009/06/30 09:08:56: Attendant: Calling extension 45

    [7] 2009/06/30 09:08:56: Cannot convert number 44 into global format

    [8] 2009/06/30 09:08:56: Play audio_moh/noise.wav

    [9] 2009/06/30 09:08:56: UDP: Opening socket on port 61172

    [9] 2009/06/30 09:08:56: UDP: Opening socket on port 61173

    [9] 2009/06/30 09:08:56: UDPv6: Opening socket on port 61172

    [9] 2009/06/30 09:08:56: UDPv6: Opening socket on port 61173

    [9] 2009/06/30 09:08:56: Using outbound proxy sip:[fe80::204:8dff:feff:ffe4]:5060;transport=udp because of flow-label

    [9] 2009/06/30 09:08:56: Resolve 20872: url sip:[fe80::204:8dff:feff:ffe4]:5060;transport=udp

    [9] 2009/06/30 09:08:56: Resolve 20872: a udp [fe80::204:8dff:feff:ffe4] 5060

    [9] 2009/06/30 09:08:56: Resolve 20872: udp [fe80::204:8dff:feff:ffe4] 5060

    [9] 2009/06/30 09:08:56: SIP Tx udp:[fe80::204:8dff:feff:ffe4]:5060:

    INVITE sip:45@[fe80::204:8dff:feff:ffe4]:5060;transport=udp SIP/2.0

    Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK-e1523731b2cb1200ab2fa761491d7109;rport

    From: "Fourty Four" <sip:44@pbx.company.com>;tag=1142770007

    To: "Fourty Five" <sip:45@pbx.company.com>

    Call-ID: c3852a9b@pbx

    CSeq: 9340 INVITE

    Max-Forwards: 70

    Contact: <sip:45@0.0.0.0:5060;transport=udp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

    Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/3.0.1.3023

    Alert-Info: <http://127.0.0.1/Bellcore-dr2>

    Content-Type: application/sdp

    Content-Length: 286

     

    v=0

    o=- 75511103 75511103 IN IP6 0.0.0.0

    s=-

    c=IN IP6 0.0.0.0

    t=0 0

    m=audio 61172 RTP/AVP 0 8 9 2 3 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:8 pcma/8000

    a=rtpmap:9 g722/8000

    a=rtpmap:2 g726-32/8000

    a=rtpmap:3 gsm/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=sendrecv

  6. IPv6 SDP Support

    in VoIP Phones

    Posted

    No, this should work. Definitevely with version 3.

     

    If the packet is sent on IPv4, the PBX will use a IPv4 identity for that. For example, if a SIP proxy would receive the SIP packet on IPv4 and then send it on IPv6, then you would have a problem. If you send it out on IPv6, then there should be no problem and you should see the IPv6 address on the SDP.

     

    What OS are you using?

     

    We installed pbxnsip on a Red Hat 3.4.6-3 workstation.

  7. IPv6 SDP Support

    in VoIP Phones

    Posted

    Does pbxnsip support IPv6 SDP? I have version 3 of pbxnsip. It allows IPv6 addressing for SIP signaling, but does not appear to support IPv6 addresses inside the SDP body, so calls between two IPv6 VoIP phones have no talk path. I noticed that when forwarding INVITEs from the caller to the callee, pbxnsip indicates the connection address is IP6, however, the address indicated is 0.0.0.0 which is a null IP4 address.

     

    Could this be my pbxnsip configuration? Or is it not supported yet? If it will be supported, is there an estimate of when this will be available?

     

    Best Regards

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