[Freeswitch-users] Call Recovery when using TCP

Anthony McGarry agtmcgarry at gmail.com
Tue Jul 23 20:24:46 MSD 2013 

I have pasted up the logs from a test call A leg UDP, B leg TCP that recovered ok

initial call - fs crashed
http://pastebin.com/0xe0QyFC

recovered call
http://pastebin.com/ByjJ4nhf


On 23 Jul 2013, at 16:39, Steven Ayre <steveayre at gmail.com> wrote:

> Is the B-leg the same call, or a new call?
> 
> 
> 
> On 23 July 2013 16:31, Anthony McGarry <agtmcgarry at gmail.com> wrote:
> Thanks Brian,
> 
> Initially I though the same and looked for something to migrate the tcp session, tcpcp and sockmi, but no joy.
> 
> I was doing some more testing and noticed that if the B leg was TCP and the A leg UDP the call recovered.
> 
> So I though my earlier assumption about TCP connection dropping was wrong as it seems the B leg reestablishes the session on the recovering server.
> The issue just seems to be the recovery of the A leg. FS always sends the A leg recovery INVITE as UDP. Even if original call was TCP. If I could force it to use TCP I believe it would recover the call.
> 
> Below is a call with A leg as UDP and B leg as TCP thats recovers fine.
> 
> 2013-07-23 16:19:46.419717 [NOTICE] switch_channel.c:1030 New Channel sofia/private/+35319032109 at 198
> 2013-07-23 16:19:46.419717 [NOTICE] switch_channel.c:1028 Rename Channel sofia/private/+35319032109@
> 2013-07-23 16:19:46.419717 [NOTICE] switch_core_sqldb.c:2744 Resurrecting fallen channel sofia/priva
> 2013-07-23 16:19:46.439717 [NOTICE] switch_channel.c:1030 New Channel sofia/internal/0877857933 at 10.1
> 2013-07-23 16:19:46.439717 [NOTICE] switch_channel.c:1028 Rename Channel sofia/internal/0877857933 at 1
> 2013-07-23 16:19:46.439717 [NOTICE] switch_core_sqldb.c:2744 Resurrecting fallen channel sofia/inter
> send 1110 bytes to udp/[10.101.23.203]:5060 at 15:19:46.459562:
>    ------------------------------------------------------------------------
>    INVITE sip:0877857933 at 10.101.23.203:5060 SIP/2.0
>    Via: SIP/2.0/UDP 10.101.23.110;rport;branch=z9hG4bK3tjcc8KU636FS
>    Route: <sip:0877857933 at 10.101.23.203:5060;lr>
>    Max-Forwards: 70
>    From: <sip:ae019032109 at 10.101.23.110:5060>;tag=B8BUB47FmmerF
>    To: <sip:0877857933 at 10.101.23.203>;tag=95ffcd055e0f78f7d5d397020e89288dba056a96
>    Call-ID: 1054-453-6232013151830-IMG2_DEG-2-10.101.23.203
>    CSeq: 46960329 INVITE
>    Contact: <sip:0877857933 at 10.101.23.110:5060>
>    User-Agent: LAB - SBC
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, hold, conference, refer
>    Privacy: none
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 246
>    X-FS-Support: update_display,send_info
>    P-Asserted-Identity: "ae019032109" <sip:ae019032109 at 10.101.23.203>
> 
>    v=0
>    o=FreeSWITCH 1374564092 1374564094 IN IP4 10.101.24.110
>    s=FreeSWITCH
>    c=IN IP4 10.101.24.110
>    t=0 0
>    m=audio 28694 RTP/AVP 8 101 13
>    a=rtpmap:8 PCMA/8000
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=rtpmap:13 CN/8000
>    a=ptime:20
>    ------------------------------------------------------------------------
> recv 398 bytes from udp/[10.101.23.203]:5060 at 15:19:46.463116:
>    ------------------------------------------------------------------------
>    SIP/2.0 100 Trying
>    Via: SIP/2.0/UDP 10.101.23.110;rport;branch=z9hG4bK3tjcc8KU636FS;received=10.101.23.110
>    Call-ID: 1054-453-6232013151830-IMG2_DEG-2-10.101.23.203
>    From: <sip:ae019032109 at 10.101.23.110:5060>;tag=B8BUB47FmmerF
>    To: <sip:0877857933 at 10.101.23.203>;tag=95ffcd055e0f78f7d5d397020e89288dba056a96
>    CSeq: 46960329 INVITE
>    Server: Dialogic-SIP/10.5.3.372 IMG2_DEG 2
>    Content-Length: 0
> 
>    ------------------------------------------------------------------------
> recv 961 bytes from udp/[10.101.23.203]:5060 at 15:19:46.463966:
>    ------------------------------------------------------------------------
>    SIP/2.0 200 OK
>    Via: SIP/2.0/UDP 10.101.23.110;rport;branch=z9hG4bK3tjcc8KU636FS;received=10.101.23.110
>    Call-ID: 1054-453-6232013151830-IMG2_DEG-2-10.101.23.203
>    From: <sip:ae019032109 at 10.101.23.110:5060>;tag=B8BUB47FmmerF
>    To: <sip:0877857933 at 10.101.23.203>;tag=95ffcd055e0f78f7d5d397020e89288dba056a96
>    Contact: <sip:0877857933 at 10.101.23.203:5060>
>    CSeq: 46960329 INVITE
>    Server: Dialogic-SIP/10.5.3.372 IMG2_DEG 2
>    Allow: INVITE, BYE, REGISTER, ACK, OPTIONS, CANCEL, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
>    Supported: path, replaces, timer, tdialog
>    Require: timer
>    Session-Expires: 1800;refresher=uas
>    Accept: application/sdp, application/dtmf-relay, text/plain
>    Content-Type: application/sdp
>    Content-Length: 239
> 
>    v=0
>    o=Dialogic_SDP 963778 1 IN IP4 10.101.23.203
>    s=Dialogic-SIP
>    c=IN IP4 10.101.24.203
>    t=0 0
>    m=audio 8332 RTP/AVP 8 101
>    a=rtpmap:8 PCMA/8000
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-15
>    a=silenceSupp:off - - - -
>    a=ptime:20
>    ------------------------------------------------------------------------
> recv 961 bytes from udp/[10.101.23.203]:5060 at 15:19:47.101507:
>    ------------------------------------------------------------------------
>    SIP/2.0 200 OK
>    Via: SIP/2.0/UDP 10.101.23.110;rport;branch=z9hG4bK3tjcc8KU636FS;received=10.101.23.110
>    Call-ID: 1054-453-6232013151830-IMG2_DEG-2-10.101.23.203
>    From: <sip:ae019032109 at 10.101.23.110:5060>;tag=B8BUB47FmmerF
>    To: <sip:0877857933 at 10.101.23.203>;tag=95ffcd055e0f78f7d5d397020e89288dba056a96
>    Contact: <sip:0877857933 at 10.101.23.203:5060>
>    CSeq: 46960329 INVITE
>    Server: Dialogic-SIP/10.5.3.372 IMG2_DEG 2
>    Allow: INVITE, BYE, REGISTER, ACK, OPTIONS, CANCEL, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
>    Supported: path, replaces, timer, tdialog
>    Require: timer
>    Session-Expires: 1800;refresher=uas
>    Accept: application/sdp, application/dtmf-relay, text/plain
>    Content-Type: application/sdp
>    Content-Length: 239
> 
>    v=0
>    o=Dialogic_SDP 963778 1 IN IP4 10.101.23.203
>    s=Dialogic-SIP
>    c=IN IP4 10.101.24.203
>    t=0 0
>    m=audio 8332 RTP/AVP 8 101
>    a=rtpmap:8 PCMA/8000
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-15
>    a=silenceSupp:off - - - -
>    a=ptime:20
>    ------------------------------------------------------------------------
> recv 961 bytes from udp/[10.101.23.203]:5060 at 15:19:48.070746:
>    ------------------------------------------------------------------------
>    SIP/2.0 200 OK
>    Via: SIP/2.0/UDP 10.101.23.110;rport;branch=z9hG4bK3tjcc8KU636FS;received=10.101.23.110
>    Call-ID: 1054-453-6232013151830-IMG2_DEG-2-10.101.23.203
>    From: <sip:ae019032109 at 10.101.23.110:5060>;tag=B8BUB47FmmerF
>    To: <sip:0877857933 at 10.101.23.203>;tag=95ffcd055e0f78f7d5d397020e89288dba056a96
>    Contact: <sip:0877857933 at 10.101.23.203:5060>
>    CSeq: 46960329 INVITE
>    Server: Dialogic-SIP/10.5.3.372 IMG2_DEG 2
>    Allow: INVITE, BYE, REGISTER, ACK, OPTIONS, CANCEL, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
>    Supported: path, replaces, timer, tdialog
>    Require: timer
>    Session-Expires: 1800;refresher=uas
>    Accept: application/sdp, application/dtmf-relay, text/plain
>    Content-Type: application/sdp
>    Content-Length: 239
> 
>    v=0
>    o=Dialogic_SDP 963778 1 IN IP4 10.101.23.203
>    s=Dialogic-SIP
>    c=IN IP4 10.101.24.203
>    t=0 0
>    m=audio 8332 RTP/AVP 8 101
>    a=rtpmap:8 PCMA/8000
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-15
>    a=silenceSupp:off - - - -
>    a=ptime:20
>    ------------------------------------------------------------------------
> send 1175 bytes to tcp/[198.19.255.1]:5060 at 15:19:49.473346:
>    ------------------------------------------------------------------------
>    INVITE sip:+35319032109 at 198.19.255.1:5060;transport=tcp SIP/2.0
>    Via: SIP/2.0/TCP 78.158.110.24;rport;branch=z9hG4bKBNF06yNpU187Q
>    Via: SIP/2.0/TCP 78.158.110.24;rport;branch=z9hG4bKZ21tSFUQvy38c
>    Max-Forwards: 69
>    From: "+353877857933" <sip:+353877857933 at 78.158.110.24>;tag=BK8QvjBeDHXtN
>    To: <sip:+35319032109 at 198.19.255.1;transport=tcp>;tag=397C7E04-26CA
>    Call-ID: f9547f1f-6e4d-1231-028b-001f290685a4
>    CSeq: 46960329 INVITE
>    Contact: <sip:+35319032109 at 78.158.110.24:5060;transport=tcp>
>    User-Agent: PlanNet21 Communications - SBC
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, hold, conference, refer
>    Privacy: none
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 246
>    X-FS-Support: update_display,send_info
>    P-Asserted-Identity: "+353877857933" <sip:+353877857933 at 78.158.110.24>
> 
>    v=0
>    o=FreeSWITCH 1374569678 1374569680 IN IP4 78.158.110.24
>    s=FreeSWITCH
>    c=IN IP4 78.158.110.24
>    t=0 0
>    m=audio 23108 RTP/AVP 8 101 13
>    a=rtpmap:8 PCMA/8000
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=rtpmap:13 CN/8000
>    a=ptime:20
>    ------------------------------------------------------------------------
> recv 484 bytes from tcp/[198.19.255.1]:5060 at 15:19:49.536636:
>    ------------------------------------------------------------------------
>    SIP/2.0 100 Trying
>    Via: SIP/2.0/TCP 78.158.110.24;rport;branch=z9hG4bKBNF06yNpU187Q,SIP/2.0/TCP 78.158.110.24;rport;
>    From: "+353877857933" <sip:+353877857933 at 78.158.110.24>;tag=BK8QvjBeDHXtN
>    To: <sip:+35319032109 at 198.19.255.1;transport=tcp>;tag=397C7E04-26CA
>    Date: Tue, 23 Jul 2013 15:19:49 GMT
>    Call-ID: f9547f1f-6e4d-1231-028b-001f290685a4
>    CSeq: 46960329 INVITE
>    Allow-Events: telephone-event
>    Server: Cisco-SIPGateway/IOS-12.x
>    Content-Length: 0
> 
>    ------------------------------------------------------------------------
> recv 1073 bytes from tcp/[198.19.255.1]:5060 at 15:19:49.601460:
>    ------------------------------------------------------------------------
>    SIP/2.0 200 OK
>    Via: SIP/2.0/TCP 78.158.110.24;rport;branch=z9hG4bKBNF06yNpU187Q,SIP/2.0/TCP 78.158.110.24;rport;
>    From: "+353877857933" <sip:+353877857933 at 78.158.110.24>;tag=BK8QvjBeDHXtN
>    To: <sip:+35319032109 at 198.19.255.1;transport=tcp>;tag=397C7E04-26CA
>    Date: Tue, 23 Jul 2013 15:19:49 GMT
>    Call-ID: f9547f1f-6e4d-1231-028b-001f290685a4
>    CSeq: 46960329 INVITE
>    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
>    Allow-Events: telephone-event
>    Remote-Party-ID: "Ian McGrath" <sip:+1947 at 198.19.255.1>;party=called;screen=yes;privacy=off
>    Contact: <sip:+35319032109 at 198.19.255.1:5060;transport=tcp>
>    Supported: replaces
>    Supported: sdp-anat
>    Server: Cisco-SIPGateway/IOS-12.x
>    Supported: timer
>    Content-Type: application/sdp
>    Content-Length: 247
> 
>    v=0
>    o=CiscoSystemsSIP-GW-UserAgent 3902 8614 IN IP4 198.19.255.1
>    s=SIP Call
>    c=IN IP4 198.19.255.1
>    t=0 0
>    m=audio 19050 RTP/AVP 8 101
>    c=IN IP4 198.19.255.1
>    a=rtpmap:8 PCMA/8000
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=ptime:20
>    ------------------------------------------------------------------------
> send 440 bytes to tcp/[198.19.255.1]:5060 at 15:19:49.603288:
>    ------------------------------------------------------------------------
>    ACK sip:+35319032109 at 198.19.255.1:5060;transport=tcp SIP/2.0
>    Via: SIP/2.0/TCP 78.158.110.24;rport;branch=z9hG4bKcy8r8S6SraZtK
>    Max-Forwards: 70
>    From: "+353877857933" <sip:+353877857933 at 78.158.110.24>;tag=BK8QvjBeDHXtN
>    To: <sip:+35319032109 at 198.19.255.1;transport=tcp>;tag=397C7E04-26CA
>    Call-ID: f9547f1f-6e4d-1231-028b-001f290685a4
>    CSeq: 46960329 ACK
>    Contact: <sip:+35319032109 at 78.158.110.24:5060;transport=tcp>
>    Content-Length: 0
> 
>    ------------------------------------------------------------------------
> send 466 bytes to udp/[10.101.23.203]:5060 at 15:19:49.606747:
>    ------------------------------------------------------------------------
>    ACK sip:0877857933 at 10.101.23.203:5060 SIP/2.0
>    Via: SIP/2.0/UDP 10.101.23.110;rport;branch=z9hG4bK43B5D34y3cX2m
>    Route: <sip:0877857933 at 10.101.23.203:5060;lr>
>    Max-Forwards: 70
>    From: <sip:ae019032109 at 10.101.23.110:5060>;tag=B8BUB47FmmerF
>    To: <sip:0877857933 at 10.101.23.203>;tag=95ffcd055e0f78f7d5d397020e89288dba056a96
>    Call-ID: 1054-453-6232013151830-IMG2_DEG-2-10.101.23.203
>    CSeq: 46960329 ACK
>    Contact: <sip:0877857933 at 10.101.23.110:5060>
>    Content-Length: 0
> 
>    ------------------------------------------------------------------------
> 
> 
> 
> 
> On 23 Jul 2013, at 14:47, Brian West <brian at freeswitch.org> wrote:
> 
> > You can't do call recovery on TCP at the moment,  You have no way to re-establish the TCP connections once FreeSWITCH goes down.
> >
> > /b
> >
> > Em Jul 23, 2013, às 8:01 AM, Anthony McGarry <agtmcgarry at gmail.com> escreveu:
> >
> >> Anyone using TCP in this scenario? Cannot find what I'm missing.
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
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> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
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> 
> _________________________________________________________________________
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> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
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