Repeat of packets - sr-users

16 May 2017

Thanks for the help. 
I have reverted to the default config file
(https://github.com/sipwise/kamailio/blob/master/etc/kamailio.cfg),
and trying to place a call between two ua (linphone & zoiper).  I
am testing totally on a LAN, clients & kamailo on the same subnet,
no nat.

Register is fine, Invite is fine, I receive the 200OK from called,
then I get the ACK from the calling, and while processing it, I
get the following errors in the log: 

May 17 14:12:32 kamailio : ERROR: <core> [resolve.c:1694]:
sip_hostport2su(): could not resolve hostname: "(null)"
May 17 14:12:32 kamailio : ERROR: <core> [forward.c:495]:
forward_request(): bad host name (null), dropping packet
May 17 14:12:32 kamailio : ERROR: sl [sl_funcs.c:363]:
sl_reply_error(): ERROR: sl_reply_error used: Unresolvable
destination (478/SL)

Digging a bit more, I've noticed that the calling party, using
Linphone, has the Contact field a bit weird: 
Contact: <sip:user@(null)>

Changing to another softphone that populates correctly this field
works ok. Is there a way to mitigate this external issue ? The
softphone works ok directly connected to asterisk for instance

Rgds
J.

On Tue, May 16, 2017 at 6:15 PM, David Villasmil
<david.villasmil.work@gmail.com
<mailto:david.villasmil.work@gmail.com>> wrote:

    There isn't an ACK received, check in kamailio side to make
    sure it is received. This is most probably a nat issue.
    On Tue, May 16, 2017 at 11:20 PM Jean Cérien
    <cerien.jean@gmail.com <mailto:cerien.jean@gmail.com>> wrote:

        Hello

        I am getting started with Kamailio (or restarted, used it
        briefly years ago), with the final objective to do load
        balancing.

        For the time being, I am just trying to have one asterisk
        and one kamailio, on the same box. I have setup a box with
        an asterisk 11.3, and kamailio 4.4. I've taken the config
        file
        from http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb. 

        My idea is that asterisk runs on port 5080, while kamailio
        on port 5060. Client interacts with Kamailio on port 5060. 

        It almost works... Registration is fine, but when I send
        an invite, it is properly acknowledged (by asterisk 100
        trying then 200 OK) - but the OK message gets repeated
        multiple times and asterisk issues its infamous
        'Retransmission timeout reached ...' - as if Kamailio
        wasnt processing it. See below ngrep traces between
        asterisk and kamailio

        Any ideas where to look ?

        Thanks
        J.

        #
        U +18.289105 192.168.2.228:5060
        <http://192.168.2.228:5060> -> 192.168.2.228:5080
        <http://192.168.2.228:5080>
          INVITE sip:102@192.168.2.228
        <mailto:sip%3A102@192.168.2.228> SIP/2.0..Record-Route:
        <sip:192.168.2.228;lr=on;ftag=1034946464>..Via:
        SIP/2.0/UDP
        192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b9922
          4198ef593.0..Via: SIP/2.0/UDP
        192.168.2.200:5085;received=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..From:
        <sip:199@192.168.2.228
        <mailto:sip%3A199@192.168.2.228>>;tag=1034946464..To:
        <sip:102@
          192.168.2.228>..Call-ID: 1571382735..CSeq: 21
        INVITE..Contact: <sip:iper@(null)>..Content-Type:
        application/sdp..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,
        REFER, NOTIFY
          , MESSAGE, SUBSCRIBE, INFO..Max-Forwards:
        69..User-Agent: Linphone/3.6.1 (eXosip2/4.1.0)..Subject:
        Phone call..Content-Length:   437....v=0..o=199 2799 2990
        IN IP4 192.
          168.2.200..s=Talk..c=IN IP4 192.168.2.200..t=0
        0..m=audio 7078 RTP/AVP 124 111 110 0 8 101..a=rtpmap:124
        opus/48000..a=fmtp:124 useinbandfec=1;
        usedtx=1..a=rtpmap:111 s
          peex/16000..a=fmtp:111 vbr=on..a=rtpmap:110
        speex/8000..a=fmtp:110 vbr=on..a=rtpmap:101
        telephone-event/8000..a=fmtp:101 0-11..m=video 9078
        RTP/AVP 103 99..a=rtpmap:103
           VP8/90000..a=rtpmap:99 MP4V-ES/90000..a=fmtp:99
        profile-level-id=3..                                      

        #
        U +0.000683 192.168.2.228:5080 <http://192.168.2.228:5080>
        -> 192.168.2.228:5060 <http://192.168.2.228:5060>
          OPTIONS sip:199@192.168.2.228:5060
        <http://sip:199@192.168.2.228:5060> SIP/2.0..Via:
        SIP/2.0/UDP
        192.168.2.228:5080;branch=z9hG4bK7c2d71fb..Max-Forwards:
        70..From: "asterisk" <sip:199@192.168.2.228:5080
        <http://sip:199@192.168.2.228:5080>>;
          tag=as57c98c4b..To: <sip:199@192.168.2.228:5060
        <http://sip:199@192.168.2.228:5060>>..Contact:
        <sip:199@192.168.2.228:5080
        <http://sip:199@192.168.2.228:5080>>..Call-ID:
        52fa034372ce18ca2b93fc1817ad38a5@192.168.2.228:5080..CSeq:
        102 OPTIONS
          ..User-Agent: Asterisk PBX 11.3.0..Date: Tue, 16 May
        2017 21:08:29 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS,
        BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH..Supported: re
          places, timer..Content-Length: 0....                    

        #
        U +0.001035 192.168.2.228:5080 <http://192.168.2.228:5080>
        -> 192.168.2.228:5060 <http://192.168.2.228:5060>
          OPTIONS sip:199@192.168.2.228:5060
        <http://sip:199@192.168.2.228:5060> SIP/2.0..Via:
        SIP/2.0/UDP
        192.168.2.228:5080;branch=z9hG4bK4248158e..Max-Forwards:
        70..From: "asterisk" <sip:199@192.168.2.228:5080
        <http://sip:199@192.168.2.228:5080>>;
          tag=as30d9cfb4..To: <sip:199@192.168.2.228:5060
        <http://sip:199@192.168.2.228:5060>>..Contact:
        <sip:199@192.168.2.228:5080
        <http://sip:199@192.168.2.228:5080>>..Call-ID:
        4331da391bca02965b2af65254717a18@192.168.2.228:5080..CSeq:
        102 OPTIONS
          ..User-Agent: Asterisk PBX 11.3.0..Date: Tue, 16 May
        2017 21:08:29 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS,
        BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH..Supported: re
          places, timer..Content-Length: 0....                    

        #
        U +0.000407 192.168.2.228:5080 <http://192.168.2.228:5080>
        -> 192.168.2.228:5060 <http://192.168.2.228:5060>
          SIP/2.0 100 Trying..Via: SIP/2.0/UDP
        192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via:
        SIP/2.0/UDP 192.168.2.200:5085;rec

        eived=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route:
        <sip:192.168.2.228;lr=on;ftag=1034946464>..From:
        <sip:199@192.168.2.228
        <mailto:sip%3A199@192.168.2.228>>;tag=1034946464..To: <sip
          :102@192.168.2.228 <mailto:102@192.168.2.228>>..Call-ID:
        1571382735..CSeq: 21 INVITE..Server: Asterisk PBX
        11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
        SUBSCRIBE, NOTIFY, INFO, PUBLIS
          H..Supported: replaces, timer..Contact:
        <sip:102@192.168.2.228:5080
        <http://sip:102@192.168.2.228:5080>>..Content-Length:
        0....                                                    

        #
        U +0.003961 192.168.2.228:5080 <http://192.168.2.228:5080>
        -> 192.168.2.228:5060 <http://192.168.2.228:5060>
          SIP/2.0 200 OK..Via: SIP/2.0/UDP
        192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via:
        SIP/2.0/UDP 192.168.2.200:5085;receive

        d=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route:
        <sip:192.168.2.228;lr=on;ftag=1034946464>..From:
        <sip:199@192.168.2.228
        <mailto:sip%3A199@192.168.2.228>>;tag=1034946464..To: <sip:102
          @192.168.2.228
        <http://192.168.2.228>>;tag=as497f35c0..Call-ID:
        1571382735..CSeq: 21 INVITE..Server: Asterisk PBX
        11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
        SUBSCRIBE, NOTIFY, I
          NFO, PUBLISH..Supported: replaces, timer..Contact:
        <sip:102@192.168.2.228:5080
        <http://sip:102@192.168.2.228:5080>>..Content-Type:
        application/sdp..Content-Length: 312....v=0..o=root
        350189084 350189084 I
          N IP4 192.168.2.228..s=Asterisk PBX 11.3.0..c=IN IP4
        192.168.2.228..t=0 0..m=audio 17148 RTP/AVP 0 8
        101..a=rtpmap:0 PCMU/8000..a=rtpmap:8
        PCMA/8000..a=rtpmap:101 telep
          hone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - -
        - -..a=ptime:20..a=sendrecv..m=video 0 RTP/AVP 103 99..  

        #
        U +0.087859 192.168.2.228:5060 <http://192.168.2.228:5060>
        -> 192.168.2.228:5080 <http://192.168.2.228:5080>
          SIP/2.0 200 OK..Via: SIP/2.0/UDP
        192.168.2.228:5080;branch=z9hG4bK7c2d71fb..From:
        "asterisk" <sip:199@192.168.2.228:5080
        <http://sip:199@192.168.2.228:5080>>;tag=as57c98c4b..To:
        <sip:199@192.168.2.228:506 <http://sip:199@192.168.2.228:506>
          0>;tag=524348182..Call-ID:
        52fa034372ce18ca2b93fc1817ad38a5@192.168.2.228:5080..CSeq:
        102 OPTIONS..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS,
        MESSAGE, SUBSCRIBE, NOTIFY,
           INFO..Accept: application/sdp..User-Agent:
        Linphone/3.6.1 (eXosip2/4.1.0)..Content-Length: 0....    

        #
        U +0.000213 192.168.2.228:5060 <http://192.168.2.228:5060>
        -> 192.168.2.228:5080 <http://192.168.2.228:5080>
          SIP/2.0 200 OK..Via: SIP/2.0/UDP
        192.168.2.228:5080;branch=z9hG4bK4248158e..From:
        "asterisk" <sip:199@192.168.2.228:5080
        <http://sip:199@192.168.2.228:5080>>;tag=as30d9cfb4..To:
        <sip:199@192.168.2.228:506 <http://sip:199@192.168.2.228:506>
          0>;tag=939659485..Call-ID:
        4331da391bca02965b2af65254717a18@192.168.2.228:5080..CSeq:
        102 OPTIONS..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS,
        MESSAGE, SUBSCRIBE, NOTIFY,
           INFO..Accept: application/sdp..User-Agent:
        Linphone/3.6.1 (eXosip2/4.1.0)..Content-Length: 0....    

        #
        U +0.011138 192.168.2.228:5080 <http://192.168.2.228:5080>
        -> 192.168.2.228:5060 <http://192.168.2.228:5060>
          SIP/2.0 200 OK..Via: SIP/2.0/UDP
        192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via:
        SIP/2.0/UDP 192.168.2.200:5085;receive

        d=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route:
        <sip:192.168.2.228;lr=on;ftag=1034946464>..From:
        <sip:199@192.168.2.228
        <mailto:sip%3A199@192.168.2.228>>;tag=1034946464..To: <sip:102
          @192.168.2.228
        <http://192.168.2.228>>;tag=as497f35c0..Call-ID:
        1571382735..CSeq: 21 INVITE..Server: Asterisk PBX
        11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
        SUBSCRIBE, NOTIFY, I
          NFO, PUBLISH..Supported: replaces, timer..Contact:
        <sip:102@192.168.2.228:5080
        <http://sip:102@192.168.2.228:5080>>..Content-Type:
        application/sdp..Content-Length: 312....v=0..o=root
        350189084 350189084 I
          N IP4 192.168.2.228..s=Asterisk PBX 11.3.0..c=IN IP4
        192.168.2.228..t=0 0..m=audio 17148 RTP/AVP 0 8
        101..a=rtpmap:0 PCMU/8000..a=rtpmap:8
        PCMA/8000..a=rtpmap:101 telep
          hone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - -
        - -..a=ptime:20..a=sendrecv..m=video 0 RTP/AVP 103 99..  

        #
        U +0.200291 192.168.2.228:5080 <http://192.168.2.228:5080>
        -> 192.168.2.228:5060 <http://192.168.2.228:5060>
          SIP/2.0 200 OK..Via: SIP/2.0/UDP
        192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via:
        SIP/2.0/UDP 192.168.2.200:5085;receive

        d=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route:
        <sip:192.168.2.228;lr=on;ftag=1034946464>..From:
        <sip:199@192.168.2.228
        <mailto:sip%3A199@192.168.2.228>>;tag=1034946464..To: <sip:102
          @192.168.2.228
        <http://192.168.2.228>>;tag=as497f35c0..Call-ID:
        1571382735..CSeq: 21 INVITE..Server: Asterisk PBX
        11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
        SUBSCRIBE, NOTIFY, I
          NFO, PUBLISH..Supported: replaces, timer..Contact:
        <sip:102@192.168.2.228:5080
        <http://sip:102@192.168.2.228:5080>>..Content-Type:
        application/sdp..Content-Length: 312....v=0..o=root
        350189084 350189084 I
          N IP4 192.168.2.228..s=Asterisk PBX 11.3.0..c=IN IP4
        192.168.2.228..t=0 0..m=audio 17148 RTP/AVP 0 8
        101..a=rtpmap:0 PCMU/8000..a=rtpmap:8
        PCMA/8000..a=rtpmap:101 telep
          hone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - -
        - -..a=ptime:20..a=sendrecv..m=video 0 RTP/AVP 103 99..  

        #
        U +0.400478 192.168.2.228:5080 <http://192.168.2.228:5080>
        -> 192.168.2.228:5060 <http://192.168.2.228:5060>
          SIP/2.0 200 OK..Via: SIP/2.0/UDP
        192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via:
        SIP/2.0/UDP 192.168.2.200:5085;receive

        d=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route:
        <sip:192.168.2.228;lr=on;ftag=1034946464>..From:
        <sip:199@192.168.2.228
        <mailto:sip%3A199@192.168.2.228>>;tag=1034946464..To: <sip:102
          @192.168.2.228
        <http://192.168.2.228>>;tag=as497f35c0..Call-ID:
        1571382735..CSeq: 21 INVITE..Server: Asterisk PBX
        11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
        SUBSCRIBE, NOTIFY, I
          NFO, PUBLISH..Supported: replaces, timer..Contact:
        <sip:102@192.168.2.228:5080
        <http://sip:102@192.168.2.228:5080>>..Content-Type:
        application/sdp..Content-Length: 312....v=0..o=root
        350189084 350189084 I
          N IP4 192.168.2.228..s=Asterisk PBX 11.3.0..c=IN IP4
        192.168.2.228..t=0 0..m=audio 17148 RTP/AVP 0 8
        101..a=rtpmap:0 PCMU/8000..a=rtpmap:8
        PCMA/8000..a=rtpmap:101 telep
          hone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - -
        - -..a=ptime:20..a=sendrecv..m=video 0 RTP/AVP 103 99..  

        #

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