So this would make more sense. was scratching my head earlier… didn't make sense

Here's what fs gets 

INVITE sip:+353877857933@voice.plannet21.ie:5060 SIP/2.0
Via: SIP/2.0/TCP 198.19.255.1:5060;branch=z9hG4bK42DC2424

Here's whats sent from recovering fs

INVITE sip:+35319032109@198.19.255.1:5060 SIP/2.0
Via: SIP/2.0/UDP 78.158.110.24;rport;branch=z9hG4bKXggpr14eHNNKH
Route: <sip:+35319032109@198.19.255.1:35556;transport=tcp>

Looking for differences the only one I can see is rport. I see on fisheye, for the update, I can see reference to sip_network_port, is it related, I have no experience with C. Does the initial invite need to have rport in the via?
As a test I added rport but still the same behaviour, although 33333 is not real and I just statically set it outgoing from uac, but this causes subsequent packets to update and have the rport present

FS in

INVITE sip:+353877857933@voice.plannet21.ie:5060 SIP/2.0
Via: SIP/2.0/TCP 198.19.255.1:5060;branch=z9hG4bK4306116E;rport=33333

SIP/2.0 100 Trying
Via: SIP/2.0/TCP 198.19.255.1:5060;branch=z9hG4bK4306116E;rport=13129

FS Recover

INVITE sip:+35319032109@198.19.255.1:5060 SIP/2.0
Via: SIP/2.0/UDP 78.158.110.24;rport;branch=z9hG4bKKB0t2By0am6HB
Route: <sip:+35319032109@198.19.255.1:13129;transport=tcp>


On 24 Jul 2013, at 18:18, Anthony Minessale <anthony.minessale@gmail.com> wrote:

Interesting. In the test I did when making the patch, the recover INVITE was tcp, it depends heavily on the VIA header on the original invite having TCP present in it.

send 1281 bytes to tcp/[1.x.x.x]:5060 at 17:14:42.035251:
   ------------------------------------------------------------------------
   INVITE sip:mod_sofia@1.x.x.x:5060 SIP/2.0
   Via: SIP/2.0/TCP 1.x.x.x;branch=z9hG4bKS0Q3SKXvB2Q9r;rport=50938



On Wed, Jul 24, 2013 at 5:34 AM, Anthony McGarry <agtmcgarry@gmail.com> wrote:
Thanks Anthony, yes latest head worked.

A leg still sends the recovery INVITE as UDP however now it has the route header with no loose routing. 


So I'm assuming the UAC uses this info to find session in UAC Table and reestablishes the session, but as UDP, even though UAC specifically is told to only use TCP.

To test I put a firewall in the path and blocked UDP 5060 and the call failed to recover, as expected.

So I think for this to work UDP will still need to be open on the UAC to recover the call.

As you said it won't work for all UACs or all situations but its a workable solution.

For reference this works when UAC is either Cisco or Dialogic.


On 23 Jul 2013, at 19:53, Anthony Minessale <anthony.minessale@gmail.com> wrote:

Try latest head, no promises on every endpoint.
P.S. try Jira next time.



On Tue, Jul 23, 2013 at 11:24 AM, Anthony McGarry <agtmcgarry@gmail.com> wrote:
I have pasted up the logs from a test call A leg UDP, B leg TCP that recovered ok

initial call - fs crashed

recovered call


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

Is the B-leg the same call, or a new call?



On 23 July 2013 16:31, Anthony McGarry <agtmcgarry@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@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@10.1
2013-07-23 16:19:46.439717 [NOTICE] switch_channel.c:1028 Rename Channel sofia/internal/0877857933@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@10.101.23.203:5060 SIP/2.0
   Via: SIP/2.0/UDP 10.101.23.110;rport;branch=z9hG4bK3tjcc8KU636FS
   Route: <sip:0877857933@10.101.23.203:5060;lr>
   Max-Forwards: 70
   From: <sip:ae019032109@10.101.23.110:5060>;tag=B8BUB47FmmerF
   To: <sip:0877857933@10.101.23.203>;tag=95ffcd055e0f78f7d5d397020e89288dba056a96
   Call-ID: 1054-453-6232013151830-IMG2_DEG-2-10.101.23.203
   CSeq: 46960329 INVITE
   Contact: <sip:0877857933@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@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@10.101.23.110:5060>;tag=B8BUB47FmmerF
   To: <sip:0877857933@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@10.101.23.110:5060>;tag=B8BUB47FmmerF
   To: <sip:0877857933@10.101.23.203>;tag=95ffcd055e0f78f7d5d397020e89288dba056a96
   Contact: <sip:0877857933@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@10.101.23.110:5060>;tag=B8BUB47FmmerF
   To: <sip:0877857933@10.101.23.203>;tag=95ffcd055e0f78f7d5d397020e89288dba056a96
   Contact: <sip:0877857933@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@10.101.23.110:5060>;tag=B8BUB47FmmerF
   To: <sip:0877857933@10.101.23.203>;tag=95ffcd055e0f78f7d5d397020e89288dba056a96
   Contact: <sip:0877857933@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@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@78.158.110.24>;tag=BK8QvjBeDHXtN
   To: <sip:+35319032109@198.19.255.1;transport=tcp>;tag=397C7E04-26CA
   Call-ID: f9547f1f-6e4d-1231-028b-001f290685a4
   CSeq: 46960329 INVITE
   Contact: <sip:+35319032109@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@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@78.158.110.24>;tag=BK8QvjBeDHXtN
   To: <sip:+35319032109@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@78.158.110.24>;tag=BK8QvjBeDHXtN
   To: <sip:+35319032109@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@198.19.255.1>;party=called;screen=yes;privacy=off
   Contact: <sip:+35319032109@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@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@78.158.110.24>;tag=BK8QvjBeDHXtN
   To: <sip:+35319032109@198.19.255.1;transport=tcp>;tag=397C7E04-26CA
   Call-ID: f9547f1f-6e4d-1231-028b-001f290685a4
   CSeq: 46960329 ACK
   Contact: <sip:+35319032109@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@10.101.23.203:5060 SIP/2.0
   Via: SIP/2.0/UDP 10.101.23.110;rport;branch=z9hG4bK43B5D34y3cX2m
   Route: <sip:0877857933@10.101.23.203:5060;lr>
   Max-Forwards: 70
   From: <sip:ae019032109@10.101.23.110:5060>;tag=B8BUB47FmmerF
   To: <sip:0877857933@10.101.23.203>;tag=95ffcd055e0f78f7d5d397020e89288dba056a96
   Call-ID: 1054-453-6232013151830-IMG2_DEG-2-10.101.23.203
   CSeq: 46960329 ACK
   Contact: <sip:0877857933@10.101.23.110:5060>
   Content-Length: 0

   ------------------------------------------------------------------------




On 23 Jul 2013, at 14:47, Brian West <brian@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@gmail.com> escreveu:
>
>> Anyone using TCP in this scenario? Cannot find what I'm missing.
>
>
> _________________________________________________________________________
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--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale@hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
googletalk:conf+888@conference.freeswitch.org
pstn:+19193869900
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org