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  1. #1
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    Join Date
    Jun 2008
    Posts
    9

    Nokia SIP Stack Problem (RE-INVITE)

    Hello,

    I'm trying to integrate Nokia phones with a VOIP Architecture.

    There are no problems with outbounds calls.

    We are experimenting problems with inbounds calls :

    When the sip signalization begin the phone receive a first INVITE. There is no SDP information in this INVITE.
    Then the SIP Proxy send a RE-INVITE with SDP information but the phone do not seems to catch these packet. We think that the Nokia SIP Stack does not recognize RE-INVITE packets. In our Ethereal/Wireshark capture file the RE-INVITE looks to be interpreted as an new INVITE packet by the phone.

    We have seen this problem with different Nokia phone :
    - E65.
    - E51.
    - E90.

    This problem is not present with the E65 with 1.0633.18.01 firmware. Unfortunately we upgraded the firmware on this phone.

    Is that a known problem with the SIP Stack of Nokia phones ? Or have you any information on this subject ?

    Here is my SIP/RTP call flow if you want see the problem in details.

    ================BEGIN CAPTURE===========================
    |Time | XXX.XX.XXX.XXX | XX.X.X.X1 | XXX.XX.XXX.XXX |
    |0,000 | INVITE | | |SIP From: siphone_number@domain.fr To:siphone_number2@XXX.XX.XXX.XXX
    | |(5060) ------------------> (5060) | |
    |0,100 | 100 Trying| | |SIP Status
    | |(5060) <------------------ (5060) | |
    |4,858 | 200 OK SDP ( AMR g711U g711A iLBC g729 telepho...event CN) | |SIP Status
    | |(5060) <------------------ (5060) | |
    |4,859 | ACK SDP ( g729 g711A telephone-event) | |SIP Request
    | |(5060) ------------------> (5060) | |
    |4,879 | INVITE SDP ( g729 g711A telephone-event) | |SIP From: siphone_number@domaine To:siphone_number2@XXX.XX.XXX.XXX
    | |(5060) ------------------> (5060) | |
    |4,935 | 100 Trying| | |SIP Status
    | |(5060) <------------------ (5060) | |
    |4,952 | | RTP (g729) |RTP Num packets:728 Duration:24.959s SSRC:0x206F
    | | |(49152) <------------------ (34582) |
    |32,348 | BYE | | |SIP Request
    | |(5060) <------------------ (5060) | |
    |32,348 | 200 OK | | |SIP Status
    | |(5060) ------------------> (5060) | |
    ================END CAPTURE==============================

    Thanks if you can answer or if you have any idea concerning this problem.

    Best regards,
    Last edited by paul_nougat; 2008-06-24 at 16:40.

  2. #2
    Regular Contributor
    Join Date
    Aug 2007
    Posts
    74

    Re: Nokia SIP Stack Problem (RE-INVITE)

    Hi,

    The 100 response that is sent to the second INVITE, is an indication that the phone does receive the second INVITE.
    But like you said it likely treats it as an out-of-dialog INVITE, not a re-INVITE.
    Could you copy the SIP message headers here?

    -jp40

  3. #3
    Registered User
    Join Date
    Jun 2008
    Posts
    9

    Re: Nokia SIP Stack Problem (RE-INVITE)

    Hello,

    Thanks for your quick answer.

    Here are the sip messages Header.

    I just replace ip by xx.x.x.xx and yyy.yy.yyy.yyy and the phones numbers by called_number, caller_number.

    The Nokia phone is the called number.

    Thanks a lot if you can help me.

    Best regards.

    -------->
    INVITE sip:called_number@xx.x.x.xx:5060;transport=udp SIP/2.0
    Allow: UPDATE,REFER,INFO
    Call-ID: 22708-EH-006ef423-58c6ed611@domain.com
    Contact: <sip:yyy.yy.yyy.yyy:5060>
    CSeq: 6180794 INVITE
    From: " " <sip:caller_number@domain.com;user=phone>;tag=22708-AN-006ef424-711b880c6
    Max-Forwards: 31
    P-Asserted-Identity: <sip:caller_number@yyy.yy.yyy.yyy;user=phone>
    P-Preferred-Identity: <sip:caller_number@yyy.yy.yyy.yyy;user=phone>
    To: <sip:called_number@yyy.yy.yyy.yyy;user=phone>
    User-Agent: SIPProxy/v4.41f (gw_sip)
    Via: SIP/2.0/UDP yyy.yy.yyy.yyy:5060;branch=z9hG4bK-5FB8-3D480B
    Content-Length: 0

    <--------
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP yyy.yy.yyy.yyy:5060;branch=z9hG4bK-5FB8-3D480B
    To: <sip:called_number@yyy.yy.yyy.yyy;user=phone>
    From: " " <sip:caller_number@domain.com;user=phone>;tag=22708-AN-006ef424-711b880c6
    Call-ID: 22708-EH-006ef423-58c6ed611@domain.com
    CSeq: 6180794 INVITE
    Content-Length: 0

    <--------
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP yyy.yy.yyy.yyy:5060;branch=z9hG4bK-5FB8-3D480B
    To: <sip:called_number@yyy.yy.yyy.yyy;user=phone>;tag=eaus790736q7ceueldg5b432
    Contact: <sip:called_number@xx.x.x.xx:5060;transport=UDP>
    From: " " <sip:caller_number@domain.com;user=phone>;tag=22708-AN-006ef424-711b880c6
    Call-ID: 22708-EH-006ef423-58c6ed611@domain.com
    CSeq: 6180794 INVITE
    Allow: INVITE,ACK,BYE,CANCEL,REFER,NOTIFY,OPTIONS,PRACK
    Content-Type: application/sdp
    Accept: application/sdp
    Content-Length: 439

    v=0
    o=Nokia-SIPUA 63382497070157250 63382497070157250 IN IP4 xx.x.x.xx
    s=-
    c=IN IP4 xx.x.x.xx
    t=0 0
    m=audio 49152 RTP/AVP 96 0 8 97 18 98 13
    a=sendrecv
    a=ptime:20
    a=maxptime:200
    a=fmtp:96 mode-change-neighbor=1
    a=fmtp:18 annexb=no
    a=fmtp:98 0-15
    a=rtpmap:96 AMR/8000/1
    a=rtpmap:0 PCMU/8000/1
    a=rtpmap:8 PCMA/8000/1
    a=rtpmap:97 iLBC/8000/1
    a=rtpmap:18 G729/8000/1
    a=rtpmap:98 telephone-event/8000/1
    a=rtpmap:13 CN/8000/1

    -------->
    ACK sip:called_number@xx.x.x.xx:5060;transport=udp SIP/2.0
    Call-ID: 22708-EH-006ef423-58c6ed611@domain.com
    Contact: <sip:yyy.yy.yyy.yyy:5060>
    Content-Type: application/sdp
    CSeq: 6180794 ACK
    From: " " <sip:caller_number@domain.com;user=phone>;tag=22708-AN-006ef424-711b880c6
    Max-Forwards: 31
    To: <sip:called_number@yyy.yy.yyy.yyy;user=phone>;tag=eaus790736q7ceueldg5b432
    User-Agent: SIPProxy/v4.41f (gw_sip)
    Via: SIP/2.0/UDP yyy.yy.yyy.yyy:5060;branch=z9hG4bK-475C-3D4834
    Content-Length: 280

    v=0
    o=cp10 121423391582 121423391582 IN IP4 0.0.0.0
    s=SIP Call
    c=IN IP4 172.0.0.0
    t=0 0
    m=audio 65534 RTP/AVP 18 8 101
    b=AS:64
    a=rtpmap:18 G729/8000/1
    a=fmtp:18 annexb=no
    a=rtpmap:8 PCMA/8000/1
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:20
    a=sendrecv

    -------->
    INVITE sip:called_number@xx.x.x.xx:5060;transport=udp SIP/2.0
    Allow: UPDATE,REFER,INFO
    Call-ID: 22708-EH-006ef423-58c6ed611@domain.com
    Contact: <sip:yyy.yy.yyy.yyy:5060>
    Content-Type: application/sdp
    CSeq: 6180795 INVITE
    From: " " <sip:caller_number@domain.com;user=phone>;tag=22708-AN-006ef424-711b880c6
    Max-Forwards: 31
    P-Asserted-Identity: <sip:caller_number@yyy.yy.yyy.yyy;user=phone>
    P-Preferred-Identity: <sip:caller_number@yyy.yy.yyy.yyy;user=phone>
    To: <sip:called_number@yyy.yy.yyy.yyy;user=phone>;tag=eaus790736q7ceueldg5b432
    User-Agent: SIPProxy/v4.41f (gw_sip)
    Via: SIP/2.0/UDP yyy.yy.yyy.yyy:5060;branch=z9hG4bK-5CDD-3D4836
    Content-Length: 284

    v=0
    o=cp10 121423391582 121423391585 IN IP4 37.5.8.86
    s=SIP Call
    c=IN IP4 212.39.140.113
    t=0 0
    m=audio 34582 RTP/AVP 18 8 98
    b=AS:64
    a=rtpmap:18 G729/8000/1
    a=fmtp:18 annexb=no
    a=rtpmap:8 PCMA/8000/1
    a=rtpmap:98 telephone-event/8000
    a=fmtp:98 0-15
    a=ptime:30
    a=sendrecv

    <--------
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP yyy.yy.yyy.yyy:5060;branch=z9hG4bK-5CDD-3D4836
    To: <sip:called_number@yyy.yy.yyy.yyy;user=phone>;tag=eaus790736q7ceueldg5b432
    From: " " <sip:caller_number@domain.com;user=phone>;tag=22708-AN-006ef424-711b880c6
    Call-ID: 22708-EH-006ef423-58c6ed611@domain.com
    CSeq: 6180795 INVITE
    Content-Length: 0

    ====================
    UNDIRECTIONNAL RTP
    WHITE COMMUNICATION (NO VOICE FLOW)
    ====================

    <--------
    BYE sip:yyy.yy.yyy.yyy:5060;transport=UDP SIP/2.0
    Via: SIP/2.0/UDP xx.x.x.xx:5060;branch=z9hG4bK4rmqmo491dhc7pa399t7kqm;rport
    To: " " <sip:caller_number@domain.com;user=phone>;tag=22708-AN-006ef424-711b880c6
    From: <sip:called_number@yyy.yy.yyy.yyy;user=phone>;tag=eaus790736q7ceueldg5b432
    Call-ID: 22708-EH-006ef423-58c6ed611@domain.com
    CSeq: 1 BYE
    Max-Forwards: 70
    Content-Length: 0

    -------->
    SIP/2.0 200 OK
    Call-ID: 22708-EH-006ef423-58c6ed611@domain.com
    CSeq: 1 BYE
    From: <sip:called_number@yyy.yy.yyy.yyy;user=phone>;tag=eaus790736q7ceueldg5b432
    Server: SIPProxy/v4.41f (gw_sip)
    To: " " <sip:caller_number@domain.com;user=phone>;tag=22708-AN-006ef424-711b880c6
    Via: SIP/2.0/UDP xx.x.x.xx:5060;received=xx.x.x.xx;rport=5060;branch=z9hG4bK4rmqmo491dhc7pa399t7kqm
    Content-Length: 0

  4. #4
    Registered User
    Join Date
    Jun 2008
    Posts
    9

    Re: Nokia SIP Stack Problem (RE-INVITE)

    Hello : some news about my problem.

    I've seen that someone have a similar problem and describe in it this post : http://discussion.forum.nokia.com/fo...ear#post304822

    Unfortunatly I do not find some help in this thread.

    We had a chance to have a Nokia E61i phone with firmware 1.0633.22.5

    With this firmware the sip captures are the same, but we do not have the problem describe below.

    1 - All calls from an MGCP phone are working.

    2 - Calls from mobile phone or from analogic phone are not working everytime.

    In almost 25% of calls we are experimtings some problems. There is no problem in signalisation (all the same)but calls duration is around 1 second and the caller phone hang-up automaticly.The mobile phone notice the user with a message like "Service unavailable". .

    We do not notice any problem in SIP flow.

    Thanks a lot if you can help me or if you have any idea about thi s problem.

    Best regards.

  5. #5
    Registered User
    Join Date
    Jun 2008
    Posts
    9

    Re: Nokia SIP Stack Problem (RE-INVITE)

    Here again some news about my problem.

    We have analyzed the SIP Flow, and noticed a strange comportement.

    While the codec negociation, the Nokia Phone do not replay with any codecs and the SIP Proxy answer by a bye.

    Here is the SIP Flow if you can help me :

    ----->
    INVITE sip:called_number@xx.x.x.xx:5060;transport=udp SIP/2.0
    Allow: UPDATE,REFER,INFO
    Call-ID: 11282-XG-007fea89-633895e23@domain.com
    Contact: <sip:yyy.yy.yyy.yyy:5060>
    Content-Type: application/sdp
    CSeq: 7159120 INVITE
    From: "caller_number" <sip:caller_number@domain.com;user=phone>;tag=11282-BI-007fea8a-2a619f720
    Max-Forwards: 31
    P-Access-Network-Info: ADSL;dsl_location="NOA=3;APRI=1;ADD=627613000";network-provided
    P-Asserted-Identity: <sip:caller_number@yyy.yy.yyy.yyy;user=phone>
    To: <sip:called_number@yyy.yy.yyy.yyy;user=phone>;tag=062ut1cf22gbikcs2i7uokr2
    User-Agent: SIPProxy/v4.41f (gw_sip)
    Via: SIP/2.0/UDP yyy.yy.yyy.yyy:5060;branch=z9hG4bK-DD8-45B7B4
    Content-Length: 234

    v=0
    o=cp10 121448350346 121448350349 IN IP4 212.39.140.5
    s=SIP Call
    c=IN IP4 212.39.140.101
    t=0 0
    m=audio 32250 RTP/AVP 8 18
    b=AS:64
    a=rtpmap:8 PCMA/8000/1
    a=rtpmap:18 G729/8000/1
    a=fmtp:18 annexb=no
    a=ptime:20
    a=sendrecv

    <--------
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP yyy.yy.yyy.yyy:5060;branch=z9hG4bK-DD8-45B7B4
    To: <sip:called_number@yyy.yy.yyy.yyy;user=phone>;tag=062ut1cf22gbikcs2i7uokr2
    From: "caller_number" <sip:caller_number@domain.com;user=phone>;tag=11282-BI-007fea8a-2a619f720
    Call-ID: 11282-XG-007fea89-633895e23@domain.com
    CSeq: 7159120 INVITE
    Content-Length: 0

    <--------
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP yyy.yy.yyy.yyy:5060;branch=z9hG4bK-DD8-45B7B4
    To: <sip:called_number@yyy.yy.yyy.yyy;user=phone>;tag=062ut1cf22gbikcs2i7uokr2
    Contact: <sip:called_number@xx.x.x.xx:5060;transport=UDP>
    From: "caller_number" <sip:caller_number@domain.com;user=phone>;tag=11282-BI-007fea8a-2a619f720
    Call-ID: 11282-XG-007fea89-633895e23@domain.com
    CSeq: 7159120 INVITE
    Allow: INVITE,ACK,BYE,CANCEL,REFER,NOTIFY,OPTIONS,PRACK
    Content-Type: application/sdp
    Accept: application/sdp
    Content-Length: 182

    v=0
    o=Nokia-SIPUA 63335907022772750 63335907022772751 IN IP4 xx.x.x.xx
    s=-
    c=IN IP4 xx.x.x.xx
    t=0 0
    m=audio 0 RTP/AVP 101
    a=fmtp:101 0-15
    a=rtpmap:101 telephone-event/8000/1


    ------>
    ACK sip:called_number@xx.x.x.xx:5060;transport=udp SIP/2.0
    Call-ID: 11282-XG-007fea89-633895e23@domain.com
    Contact: <sip:yyy.yy.yyy.yyy:5060>
    CSeq: 7159120 ACK
    From: "caller_number" <sip:caller_number@domain.com;user=phone>;tag=11282-BI-007fea8a-2a619f720
    Max-Forwards: 31
    To: <sip:called_number@yyy.yy.yyy.yyy;user=phone>;tag=062ut1cf22gbikcs2i7uokr2
    User-Agent: SIPProxy/v4.41f (gw_sip)
    Via: SIP/2.0/UDP yyy.yy.yyy.yyy:5060;branch=z9hG4bK-2EEF-45B7B6
    Content-Length: 0

    ------->
    BYE sip:called_number@xx.x.x.xx:5060;transport=udp SIP/2.0
    Call-ID: 11282-XG-007fea89-633895e23@domain.com
    CSeq: 7159121 BYE
    From: "caller_number" <sip:caller_number@domain.com;user=phone>;tag=11282-BI-007fea8a-2a619f720
    Max-Forwards: 31
    Reason: q.850;cause=65
    To: <sip:called_number@yyy.yy.yyy.yyy;user=phone>;tag=062ut1cf22gbikcs2i7uokr2
    User-Agent: SIPProxy/v4.41f (gw_sip)
    Via: SIP/2.0/UDP yyy.yy.yyy.yyy:5060;branch=z9hG4bK-1E3C-45B7B7
    Content-Length: 0

    Thanks if you can help.

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