Call Recovery when using TCP
Hi, FreeSWITCH (Version 1.2.11 git c9790ea 2013-07-16 02:46:21Z) I have two physical boxes active/passive using heartbeat and corosync. Using FS as an SBC with two profiles. Internal and Private. Both profiles are using track-calls for call recovery. Works fine when using UDP as the transport for both legs. Works fine when A leg uses UDP and B leg uses TCP Fails when A leg uses TCP and b leg uses either UDP or TCP. It seems that the A leg always uses UDP as the transport for the INVITE to recover. Even if the initial call used TCP before the failure. Anyone using TCP in this scenario? Cannot find what I'm missing. Thanks A
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.
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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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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Its the same call. Two way audio, sup services etc.. 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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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@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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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 http://pastebin.com/0xe0QyFC
recovered call http://pastebin.com/ByjJ4nhf
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
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. Route: <sip:+35314611947@198.19.255.1:61767;transport=tcp> 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 http://pastebin.com/0xe0QyFC
recovered call http://pastebin.com/ByjJ4nhf
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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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 Route: <sip:1004@1.x.x.x:50938;transport=tcp> 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.
Route: <sip:+35314611947@198.19.255.1:61767;transport=tcp>
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 http://pastebin.com/0xe0QyFC
recovered call http://pastebin.com/ByjJ4nhf
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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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 Route: <sip:1004@1.x.x.x:50938;transport=tcp>
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.
Route: <sip:+35314611947@198.19.255.1:61767;transport=tcp>
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 http://pastebin.com/0xe0QyFC
recovered call http://pastebin.com/ByjJ4nhf
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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maybe its because transport=tcp is not there in the req invite, I'll see if I can force that into the url when applicable. The via from the 200ok is the one that is preserved for the recover invite. This is all a bit of trickery cos the sofia stack has no idea its doing a recovery it thinks its a new outbound uac call. We someday need to change the code to allow us to create a uas instance on demand from the dialog data we have. On Wed, Jul 24, 2013 at 3:49 PM, Anthony McGarry <agtmcgarry@gmail.com>wrote:
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 Route: <sip:1004@1.x.x.x:50938;transport=tcp>
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.
Route: <sip:+35314611947@198.19.255.1:61767;transport=tcp>
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 http://pastebin.com/0xe0QyFC
recovered call http://pastebin.com/ByjJ4nhf
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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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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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
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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I think I would need to see the whole sip trace of the original invite and the recover leg. The only diff I see between mine and yours is mine is using FS on both sides and its including transport=tcp On Wed, Jul 24, 2013 at 4:07 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
maybe its because transport=tcp is not there in the req invite, I'll see if I can force that into the url when applicable. The via from the 200ok is the one that is preserved for the recover invite. This is all a bit of trickery cos the sofia stack has no idea its doing a recovery it thinks its a new outbound uac call. We someday need to change the code to allow us to create a uas instance on demand from the dialog data we have.
On Wed, Jul 24, 2013 at 3:49 PM, Anthony McGarry <agtmcgarry@gmail.com>wrote:
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 Route: <sip:1004@1.x.x.x:50938;transport=tcp>
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.
Route: <sip:+35314611947@198.19.255.1:61767;transport=tcp>
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 http://pastebin.com/0xe0QyFC
recovered call http://pastebin.com/ByjJ4nhf
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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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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ok so I added INVITE sip:+353877857933@voice.plannet21.ie:5060;transport=tcp SIP/2.0 to all outgoing INVITES from my client. Uploaded logs from both servers before crash http://pastebin.freeswitch.org/21233 after crash http://pastebin.freeswitch.org/21234 On 24 Jul 2013, at 22:13, Anthony Minessale <anthony.minessale@gmail.com> wrote:
I think I would need to see the whole sip trace of the original invite and the recover leg. The only diff I see between mine and yours is mine is using FS on both sides and its including transport=tcp
On Wed, Jul 24, 2013 at 4:07 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote: maybe its because transport=tcp is not there in the req invite, I'll see if I can force that into the url when applicable. The via from the 200ok is the one that is preserved for the recover invite. This is all a bit of trickery cos the sofia stack has no idea its doing a recovery it thinks its a new outbound uac call. We someday need to change the code to allow us to create a uas instance on demand from the dialog data we have.
On Wed, Jul 24, 2013 at 3:49 PM, Anthony McGarry <agtmcgarry@gmail.com> wrote: 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 Route: <sip:1004@1.x.x.x:50938;transport=tcp>
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.
Route: <sip:+35314611947@198.19.255.1:61767;transport=tcp>
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 http://pastebin.com/0xe0QyFC
recovered call http://pastebin.com/ByjJ4nhf
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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can you do that again with this extra command right before recover is executed? sofia loglevel all 9 maybe its failing on tcp and reverting to udp On Wed, Jul 24, 2013 at 5:01 PM, Anthony McGarry <agtmcgarry@gmail.com>wrote:
ok so I added INVITE sip:+353877857933@voice.plannet21.ie:5060;transport=tcp SIP/2.0 to all outgoing INVITES from my client.
Uploaded logs from both servers
before crash http://pastebin.freeswitch.org/21233
after crash http://pastebin.freeswitch.org/21234
On 24 Jul 2013, at 22:13, Anthony Minessale <anthony.minessale@gmail.com> wrote:
I think I would need to see the whole sip trace of the original invite and the recover leg. The only diff I see between mine and yours is mine is using FS on both sides and its including transport=tcp
On Wed, Jul 24, 2013 at 4:07 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
maybe its because transport=tcp is not there in the req invite, I'll see if I can force that into the url when applicable. The via from the 200ok is the one that is preserved for the recover invite. This is all a bit of trickery cos the sofia stack has no idea its doing a recovery it thinks its a new outbound uac call. We someday need to change the code to allow us to create a uas instance on demand from the dialog data we have.
On Wed, Jul 24, 2013 at 3:49 PM, Anthony McGarry <agtmcgarry@gmail.com>wrote:
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 Route: <sip:1004@1.x.x.x:50938;transport=tcp>
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.
Route: <sip:+35314611947@198.19.255.1:61767;transport=tcp>
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 http://pastebin.com/0xe0QyFC
recovered call http://pastebin.com/ByjJ4nhf
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. > > > _________________________________________________________________________ > 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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
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_________________________________________________________________________ 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
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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
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
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_________________________________________________________________________ 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
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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
_________________________________________________________________________ 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
-- 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
-- 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
_________________________________________________________________________ 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
-- 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
http://pastebin.freeswitch.org/21235 On 24 Jul 2013, at 23:17, Anthony Minessale <anthony.minessale@gmail.com> wrote:
can you do that again with this extra command right before recover is executed?
sofia loglevel all 9
maybe its failing on tcp and reverting to udp
On Wed, Jul 24, 2013 at 5:01 PM, Anthony McGarry <agtmcgarry@gmail.com> wrote: ok so I added INVITE sip:+353877857933@voice.plannet21.ie:5060;transport=tcp SIP/2.0 to all outgoing INVITES from my client.
Uploaded logs from both servers
before crash http://pastebin.freeswitch.org/21233
after crash http://pastebin.freeswitch.org/21234
On 24 Jul 2013, at 22:13, Anthony Minessale <anthony.minessale@gmail.com> wrote:
I think I would need to see the whole sip trace of the original invite and the recover leg. The only diff I see between mine and yours is mine is using FS on both sides and its including transport=tcp
On Wed, Jul 24, 2013 at 4:07 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote: maybe its because transport=tcp is not there in the req invite, I'll see if I can force that into the url when applicable. The via from the 200ok is the one that is preserved for the recover invite. This is all a bit of trickery cos the sofia stack has no idea its doing a recovery it thinks its a new outbound uac call. We someday need to change the code to allow us to create a uas instance on demand from the dialog data we have.
On Wed, Jul 24, 2013 at 3:49 PM, Anthony McGarry <agtmcgarry@gmail.com> wrote: 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 Route: <sip:1004@1.x.x.x:50938;transport=tcp>
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.
Route: <sip:+35314611947@198.19.255.1:61767;transport=tcp>
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 http://pastebin.com/0xe0QyFC
recovered call http://pastebin.com/ByjJ4nhf
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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is it any different if you turn off auth-calls? There must be something subtle about it that is not obvious. I may need to have it labbed up where i can actually mess with the code to figure it out. On Wed, Jul 24, 2013 at 5:29 PM, Anthony McGarry <agtmcgarry@gmail.com>wrote:
http://pastebin.freeswitch.org/21235
On 24 Jul 2013, at 23:17, Anthony Minessale <anthony.minessale@gmail.com> wrote:
can you do that again with this extra command right before recover is executed?
sofia loglevel all 9
maybe its failing on tcp and reverting to udp
On Wed, Jul 24, 2013 at 5:01 PM, Anthony McGarry <agtmcgarry@gmail.com>wrote:
ok so I added INVITE sip:+353877857933@voice.plannet21.ie:5060;transport=tcp SIP/2.0 to all outgoing INVITES from my client.
Uploaded logs from both servers
before crash http://pastebin.freeswitch.org/21233
after crash http://pastebin.freeswitch.org/21234
On 24 Jul 2013, at 22:13, Anthony Minessale <anthony.minessale@gmail.com> wrote:
I think I would need to see the whole sip trace of the original invite and the recover leg. The only diff I see between mine and yours is mine is using FS on both sides and its including transport=tcp
On Wed, Jul 24, 2013 at 4:07 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
maybe its because transport=tcp is not there in the req invite, I'll see if I can force that into the url when applicable. The via from the 200ok is the one that is preserved for the recover invite. This is all a bit of trickery cos the sofia stack has no idea its doing a recovery it thinks its a new outbound uac call. We someday need to change the code to allow us to create a uas instance on demand from the dialog data we have.
On Wed, Jul 24, 2013 at 3:49 PM, Anthony McGarry <agtmcgarry@gmail.com>wrote:
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 Route: <sip:1004@1.x.x.x:50938;transport=tcp>
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.
Route: <sip:+35314611947@198.19.255.1:61767;transport=tcp>
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 http://pastebin.com/0xe0QyFC
recovered call http://pastebin.com/ByjJ4nhf
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. > > > > > > > _________________________________________________________________________ > > 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 > > > > _________________________________________________________________________ > 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 >
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
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-- Anthony Minessale II
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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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
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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Still the same with auth-calls off. Appreciate the time you've spent on this. If you have time to put it in the lab then sure, if not no worries. On 24 Jul 2013, at 23:50, Anthony Minessale <anthony.minessale@gmail.com> wrote:
is it any different if you turn off auth-calls?
There must be something subtle about it that is not obvious. I may need to have it labbed up where i can actually mess with the code to figure it out.
On Wed, Jul 24, 2013 at 5:29 PM, Anthony McGarry <agtmcgarry@gmail.com> wrote: http://pastebin.freeswitch.org/21235
On 24 Jul 2013, at 23:17, Anthony Minessale <anthony.minessale@gmail.com> wrote:
can you do that again with this extra command right before recover is executed?
sofia loglevel all 9
maybe its failing on tcp and reverting to udp
On Wed, Jul 24, 2013 at 5:01 PM, Anthony McGarry <agtmcgarry@gmail.com> wrote: ok so I added INVITE sip:+353877857933@voice.plannet21.ie:5060;transport=tcp SIP/2.0 to all outgoing INVITES from my client.
Uploaded logs from both servers
before crash http://pastebin.freeswitch.org/21233
after crash http://pastebin.freeswitch.org/21234
On 24 Jul 2013, at 22:13, Anthony Minessale <anthony.minessale@gmail.com> wrote:
I think I would need to see the whole sip trace of the original invite and the recover leg. The only diff I see between mine and yours is mine is using FS on both sides and its including transport=tcp
On Wed, Jul 24, 2013 at 4:07 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote: maybe its because transport=tcp is not there in the req invite, I'll see if I can force that into the url when applicable. The via from the 200ok is the one that is preserved for the recover invite. This is all a bit of trickery cos the sofia stack has no idea its doing a recovery it thinks its a new outbound uac call. We someday need to change the code to allow us to create a uas instance on demand from the dialog data we have.
On Wed, Jul 24, 2013 at 3:49 PM, Anthony McGarry <agtmcgarry@gmail.com> wrote: 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 Route: <sip:1004@1.x.x.x:50938;transport=tcp>
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.
Route: <sip:+35314611947@198.19.255.1:61767;transport=tcp>
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 http://pastebin.com/0xe0QyFC
recovered call http://pastebin.com/ByjJ4nhf
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. > > > _________________________________________________________________________ > 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
_________________________________________________________________________ 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
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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_________________________________________________________________________ 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
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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
-- 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
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_________________________________________________________________________ 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
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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
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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-- Anthony Minessale II
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If you can setup an open ccm temporarily that passes calls to the same ext @ sidious.freeswitch.org and email me the info direct, I can have a look. On Jul 24, 2013 6:25 PM, "Anthony McGarry" <agtmcgarry@gmail.com> wrote:
Still the same with auth-calls off. Appreciate the time you've spent on this. If you have time to put it in the lab then sure, if not no worries.
On 24 Jul 2013, at 23:50, Anthony Minessale <anthony.minessale@gmail.com> wrote:
is it any different if you turn off auth-calls?
There must be something subtle about it that is not obvious. I may need to have it labbed up where i can actually mess with the code to figure it out.
On Wed, Jul 24, 2013 at 5:29 PM, Anthony McGarry <agtmcgarry@gmail.com>wrote:
http://pastebin.freeswitch.org/21235
On 24 Jul 2013, at 23:17, Anthony Minessale <anthony.minessale@gmail.com> wrote:
can you do that again with this extra command right before recover is executed?
sofia loglevel all 9
maybe its failing on tcp and reverting to udp
On Wed, Jul 24, 2013 at 5:01 PM, Anthony McGarry <agtmcgarry@gmail.com>wrote:
ok so I added INVITE sip:+353877857933@voice.plannet21.ie:5060;transport=tcp SIP/2.0 to all outgoing INVITES from my client.
Uploaded logs from both servers
before crash http://pastebin.freeswitch.org/21233
after crash http://pastebin.freeswitch.org/21234
On 24 Jul 2013, at 22:13, Anthony Minessale <anthony.minessale@gmail.com> wrote:
I think I would need to see the whole sip trace of the original invite and the recover leg. The only diff I see between mine and yours is mine is using FS on both sides and its including transport=tcp
On Wed, Jul 24, 2013 at 4:07 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
maybe its because transport=tcp is not there in the req invite, I'll see if I can force that into the url when applicable. The via from the 200ok is the one that is preserved for the recover invite. This is all a bit of trickery cos the sofia stack has no idea its doing a recovery it thinks its a new outbound uac call. We someday need to change the code to allow us to create a uas instance on demand from the dialog data we have.
On Wed, Jul 24, 2013 at 3:49 PM, Anthony McGarry <agtmcgarry@gmail.com>wrote:
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 Route: <sip:1004@1.x.x.x:50938;transport=tcp>
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.
Route: <sip:+35314611947@198.19.255.1:61767;transport=tcp>
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 > http://pastebin.com/0xe0QyFC > > recovered call > http://pastebin.com/ByjJ4nhf > > > 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. >> > >> > >> > >> _________________________________________________________________________ >> > 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 >> >> >> >> _________________________________________________________________________ >> 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 >> > > > _________________________________________________________________________ > 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 > > > > > _________________________________________________________________________ > 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 > >
-- 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
_________________________________________________________________________ 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
-- 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
_________________________________________________________________________ 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
-- 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
-- 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
_________________________________________________________________________ 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
-- 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
_________________________________________________________________________ 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
-- 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
_________________________________________________________________________ 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
Hi Anthony (McGarry), If possible, could you spend a few minutes summarizing this thread and create a JIRA ticket? That way it won't get forgotten about, and helps us keep track of things better. [1] Also for future ref, try and use the FS pastebin for pasting logs otherwise it makes threads a bit hard to read [2] Thank you! Cal [1] http://wiki.freeswitch.org/wiki/Reporting_Bugs#Reporting_A_Bug_With_JIRA [2] http://wiki.freeswitch.org/wiki/Reporting_Bugs#Using_The_Pastebin On Thu, Jul 25, 2013 at 12:20 AM, Anthony McGarry <agtmcgarry@gmail.com>wrote:
Still the same with auth-calls off. Appreciate the time you've spent on this. If you have time to put it in the lab then sure, if not no worries.
On 24 Jul 2013, at 23:50, Anthony Minessale <anthony.minessale@gmail.com> wrote:
is it any different if you turn off auth-calls?
There must be something subtle about it that is not obvious. I may need to have it labbed up where i can actually mess with the code to figure it out.
On Wed, Jul 24, 2013 at 5:29 PM, Anthony McGarry <agtmcgarry@gmail.com>wrote:
http://pastebin.freeswitch.org/21235
On 24 Jul 2013, at 23:17, Anthony Minessale <anthony.minessale@gmail.com> wrote:
can you do that again with this extra command right before recover is executed?
sofia loglevel all 9
maybe its failing on tcp and reverting to udp
On Wed, Jul 24, 2013 at 5:01 PM, Anthony McGarry <agtmcgarry@gmail.com>wrote:
ok so I added INVITE sip:+353877857933@voice.plannet21.ie:5060;transport=tcp SIP/2.0 to all outgoing INVITES from my client.
Uploaded logs from both servers
before crash http://pastebin.freeswitch.org/21233
after crash http://pastebin.freeswitch.org/21234
On 24 Jul 2013, at 22:13, Anthony Minessale <anthony.minessale@gmail.com> wrote:
I think I would need to see the whole sip trace of the original invite and the recover leg. The only diff I see between mine and yours is mine is using FS on both sides and its including transport=tcp
On Wed, Jul 24, 2013 at 4:07 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
maybe its because transport=tcp is not there in the req invite, I'll see if I can force that into the url when applicable. The via from the 200ok is the one that is preserved for the recover invite. This is all a bit of trickery cos the sofia stack has no idea its doing a recovery it thinks its a new outbound uac call. We someday need to change the code to allow us to create a uas instance on demand from the dialog data we have.
On Wed, Jul 24, 2013 at 3:49 PM, Anthony McGarry <agtmcgarry@gmail.com>wrote:
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 Route: <sip:1004@1.x.x.x:50938;transport=tcp>
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.
Route: <sip:+35314611947@198.19.255.1:61767;transport=tcp>
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 > http://pastebin.com/0xe0QyFC > > recovered call > http://pastebin.com/ByjJ4nhf > > > 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. >> > >> > >> > >> _________________________________________________________________________ >> > 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 >> >> >> >> _________________________________________________________________________ >> 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 >> > > > _________________________________________________________________________ > 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 > > > > > _________________________________________________________________________ > 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 > >
-- 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
_________________________________________________________________________ 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
-- 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
_________________________________________________________________________ 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
-- 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
-- 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
_________________________________________________________________________ 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
-- 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
_________________________________________________________________________ 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
-- Anthony Minessale II
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Hi, Opened Jira. http://jira.freeswitch.org/browse/FS-5638 On 25 Jul 2013, at 02:04, Cal Leeming [Simplicity Media Ltd] <cal.leeming@simplicitymedialtd.co.uk> wrote:
Hi Anthony (McGarry),
If possible, could you spend a few minutes summarizing this thread and create a JIRA ticket? That way it won't get forgotten about, and helps us keep track of things better. [1]
Also for future ref, try and use the FS pastebin for pasting logs otherwise it makes threads a bit hard to read [2]
Thank you!
Cal
[1] http://wiki.freeswitch.org/wiki/Reporting_Bugs#Reporting_A_Bug_With_JIRA [2] http://wiki.freeswitch.org/wiki/Reporting_Bugs#Using_The_Pastebin
On Thu, Jul 25, 2013 at 12:20 AM, Anthony McGarry <agtmcgarry@gmail.com> wrote: Still the same with auth-calls off. Appreciate the time you've spent on this. If you have time to put it in the lab then sure, if not no worries.
On 24 Jul 2013, at 23:50, Anthony Minessale <anthony.minessale@gmail.com> wrote:
is it any different if you turn off auth-calls?
There must be something subtle about it that is not obvious. I may need to have it labbed up where i can actually mess with the code to figure it out.
On Wed, Jul 24, 2013 at 5:29 PM, Anthony McGarry <agtmcgarry@gmail.com> wrote: http://pastebin.freeswitch.org/21235
On 24 Jul 2013, at 23:17, Anthony Minessale <anthony.minessale@gmail.com> wrote:
can you do that again with this extra command right before recover is executed?
sofia loglevel all 9
maybe its failing on tcp and reverting to udp
On Wed, Jul 24, 2013 at 5:01 PM, Anthony McGarry <agtmcgarry@gmail.com> wrote: ok so I added INVITE sip:+353877857933@voice.plannet21.ie:5060;transport=tcp SIP/2.0 to all outgoing INVITES from my client.
Uploaded logs from both servers
before crash http://pastebin.freeswitch.org/21233
after crash http://pastebin.freeswitch.org/21234
On 24 Jul 2013, at 22:13, Anthony Minessale <anthony.minessale@gmail.com> wrote:
I think I would need to see the whole sip trace of the original invite and the recover leg. The only diff I see between mine and yours is mine is using FS on both sides and its including transport=tcp
On Wed, Jul 24, 2013 at 4:07 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote: maybe its because transport=tcp is not there in the req invite, I'll see if I can force that into the url when applicable. The via from the 200ok is the one that is preserved for the recover invite. This is all a bit of trickery cos the sofia stack has no idea its doing a recovery it thinks its a new outbound uac call. We someday need to change the code to allow us to create a uas instance on demand from the dialog data we have.
On Wed, Jul 24, 2013 at 3:49 PM, Anthony McGarry <agtmcgarry@gmail.com> wrote: 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 Route: <sip:1004@1.x.x.x:50938;transport=tcp>
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.
Route: <sip:+35314611947@198.19.255.1:61767;transport=tcp>
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 http://pastebin.com/0xe0QyFC
recovered call http://pastebin.com/ByjJ4nhf
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. > > > > > > _________________________________________________________________________ > > 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 > > > _________________________________________________________________________ > 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 > > _________________________________________________________________________ > 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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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-- Anthony Minessale II
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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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
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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-- Anthony Minessale II
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I think its resolved now. I missed part of the patch on the merge to stable branch. On Thu, Jul 25, 2013 at 5:19 AM, Anthony McGarry <agtmcgarry@gmail.com>wrote:
Hi,
Opened Jira. http://jira.freeswitch.org/browse/FS-5638
On 25 Jul 2013, at 02:04, Cal Leeming [Simplicity Media Ltd] < cal.leeming@simplicitymedialtd.co.uk> wrote:
Hi Anthony (McGarry),
If possible, could you spend a few minutes summarizing this thread and create a JIRA ticket? That way it won't get forgotten about, and helps us keep track of things better. [1]
Also for future ref, try and use the FS pastebin for pasting logs otherwise it makes threads a bit hard to read [2]
Thank you!
Cal
[1] http://wiki.freeswitch.org/wiki/Reporting_Bugs#Reporting_A_Bug_With_JIRA [2] http://wiki.freeswitch.org/wiki/Reporting_Bugs#Using_The_Pastebin
On Thu, Jul 25, 2013 at 12:20 AM, Anthony McGarry <agtmcgarry@gmail.com>wrote:
Still the same with auth-calls off. Appreciate the time you've spent on this. If you have time to put it in the lab then sure, if not no worries.
On 24 Jul 2013, at 23:50, Anthony Minessale <anthony.minessale@gmail.com> wrote:
is it any different if you turn off auth-calls?
There must be something subtle about it that is not obvious. I may need to have it labbed up where i can actually mess with the code to figure it out.
On Wed, Jul 24, 2013 at 5:29 PM, Anthony McGarry <agtmcgarry@gmail.com>wrote:
http://pastebin.freeswitch.org/21235
On 24 Jul 2013, at 23:17, Anthony Minessale <anthony.minessale@gmail.com> wrote:
can you do that again with this extra command right before recover is executed?
sofia loglevel all 9
maybe its failing on tcp and reverting to udp
On Wed, Jul 24, 2013 at 5:01 PM, Anthony McGarry <agtmcgarry@gmail.com>wrote:
ok so I added INVITE sip:+353877857933@voice.plannet21.ie:5060;transport=tcp SIP/2.0 to all outgoing INVITES from my client.
Uploaded logs from both servers
before crash http://pastebin.freeswitch.org/21233
after crash http://pastebin.freeswitch.org/21234
On 24 Jul 2013, at 22:13, Anthony Minessale < anthony.minessale@gmail.com> wrote:
I think I would need to see the whole sip trace of the original invite and the recover leg. The only diff I see between mine and yours is mine is using FS on both sides and its including transport=tcp
On Wed, Jul 24, 2013 at 4:07 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
maybe its because transport=tcp is not there in the req invite, I'll see if I can force that into the url when applicable. The via from the 200ok is the one that is preserved for the recover invite. This is all a bit of trickery cos the sofia stack has no idea its doing a recovery it thinks its a new outbound uac call. We someday need to change the code to allow us to create a uas instance on demand from the dialog data we have.
On Wed, Jul 24, 2013 at 3:49 PM, Anthony McGarry <agtmcgarry@gmail.com
wrote:
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 Route: <sip:1004@1.x.x.x:50938;transport=tcp>
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. > > Route: <sip:+35314611947@198.19.255.1:61767;transport=tcp> > > 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 >> http://pastebin.com/0xe0QyFC >> >> recovered call >> http://pastebin.com/ByjJ4nhf >> >> >> 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. >>> > >>> > >>> > >>> _________________________________________________________________________ >>> > 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 >>> >>> >>> >>> _________________________________________________________________________ >>> 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 >>> >> >> >> _________________________________________________________________________ >> 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 >> >> >> >> >> _________________________________________________________________________ >> 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 >> >> > > > -- > 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 > > > > > _________________________________________________________________________ > 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 > >
-- 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
_________________________________________________________________________ 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
-- 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
-- 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
_________________________________________________________________________ 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
-- 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
_________________________________________________________________________ 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
-- 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
_________________________________________________________________________ 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
_________________________________________________________________________ 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
_________________________________________________________________________ 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
-- 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
Hi Anthony (McGarry), Could you update the thread/ticket to let us know if this is now sorted after Anthony's last update? Many thanks again Cal On Thu, Jul 25, 2013 at 8:38 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
I think its resolved now. I missed part of the patch on the merge to stable branch.
On Thu, Jul 25, 2013 at 5:19 AM, Anthony McGarry <agtmcgarry@gmail.com>wrote:
Hi,
Opened Jira. http://jira.freeswitch.org/browse/FS-5638
On 25 Jul 2013, at 02:04, Cal Leeming [Simplicity Media Ltd] < cal.leeming@simplicitymedialtd.co.uk> wrote:
Hi Anthony (McGarry),
If possible, could you spend a few minutes summarizing this thread and create a JIRA ticket? That way it won't get forgotten about, and helps us keep track of things better. [1]
Also for future ref, try and use the FS pastebin for pasting logs otherwise it makes threads a bit hard to read [2]
Thank you!
Cal
[1] http://wiki.freeswitch.org/wiki/Reporting_Bugs#Reporting_A_Bug_With_JIRA [2] http://wiki.freeswitch.org/wiki/Reporting_Bugs#Using_The_Pastebin
On Thu, Jul 25, 2013 at 12:20 AM, Anthony McGarry <agtmcgarry@gmail.com>wrote:
Still the same with auth-calls off. Appreciate the time you've spent on this. If you have time to put it in the lab then sure, if not no worries.
On 24 Jul 2013, at 23:50, Anthony Minessale <anthony.minessale@gmail.com> wrote:
is it any different if you turn off auth-calls?
There must be something subtle about it that is not obvious. I may need to have it labbed up where i can actually mess with the code to figure it out.
On Wed, Jul 24, 2013 at 5:29 PM, Anthony McGarry <agtmcgarry@gmail.com>wrote:
http://pastebin.freeswitch.org/21235
On 24 Jul 2013, at 23:17, Anthony Minessale < anthony.minessale@gmail.com> wrote:
can you do that again with this extra command right before recover is executed?
sofia loglevel all 9
maybe its failing on tcp and reverting to udp
On Wed, Jul 24, 2013 at 5:01 PM, Anthony McGarry <agtmcgarry@gmail.com>wrote:
ok so I added INVITE sip:+353877857933@voice.plannet21.ie:5060;transport=tcp SIP/2.0 to all outgoing INVITES from my client.
Uploaded logs from both servers
before crash http://pastebin.freeswitch.org/21233
after crash http://pastebin.freeswitch.org/21234
On 24 Jul 2013, at 22:13, Anthony Minessale < anthony.minessale@gmail.com> wrote:
I think I would need to see the whole sip trace of the original invite and the recover leg. The only diff I see between mine and yours is mine is using FS on both sides and its including transport=tcp
On Wed, Jul 24, 2013 at 4:07 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
maybe its because transport=tcp is not there in the req invite, I'll see if I can force that into the url when applicable. The via from the 200ok is the one that is preserved for the recover invite. This is all a bit of trickery cos the sofia stack has no idea its doing a recovery it thinks its a new outbound uac call. We someday need to change the code to allow us to create a uas instance on demand from the dialog data we have.
On Wed, Jul 24, 2013 at 3:49 PM, Anthony McGarry < agtmcgarry@gmail.com> wrote:
> 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 > Route: <sip:1004@1.x.x.x:50938;transport=tcp> > > > > 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. >> >> Route: <sip:+35314611947@198.19.255.1:61767;transport=tcp> >> >> 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 >>> http://pastebin.com/0xe0QyFC >>> >>> recovered call >>> http://pastebin.com/ByjJ4nhf >>> >>> >>> 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=tcpSIP/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. >>>> > >>>> > >>>> > >>>> _________________________________________________________________________ >>>> > 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 >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> 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 >>>> >>> >>> >>> _________________________________________________________________________ >>> 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 >>> >>> >>> >>> >>> _________________________________________________________________________ >>> 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 >>> >>> >> >> >> -- >> 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 >> >> >> >> >> _________________________________________________________________________ >> 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 >> >> > > > -- > 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 > > > > > _________________________________________________________________________ > 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 > >
-- 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
-- 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
_________________________________________________________________________ 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
-- 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
_________________________________________________________________________ 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
-- 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
_________________________________________________________________________ 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
_________________________________________________________________________ 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
_________________________________________________________________________ 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
-- 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
Hey Cal, Yes this is now resolved. We have TCP recovery on both legs. Thanks for all the help Anthony Anthony On 27 Jul 2013, at 04:13, "Cal Leeming [Simplicity Media Ltd]" <cal.leeming@simplicitymedialtd.co.uk> wrote:
Hi Anthony (McGarry),
Could you update the thread/ticket to let us know if this is now sorted after Anthony's last update?
Many thanks again
Cal
On Thu, Jul 25, 2013 at 8:38 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote: I think its resolved now. I missed part of the patch on the merge to stable branch.
On Thu, Jul 25, 2013 at 5:19 AM, Anthony McGarry <agtmcgarry@gmail.com> wrote: Hi,
Opened Jira. http://jira.freeswitch.org/browse/FS-5638
On 25 Jul 2013, at 02:04, Cal Leeming [Simplicity Media Ltd] <cal.leeming@simplicitymedialtd.co.uk> wrote:
Hi Anthony (McGarry),
If possible, could you spend a few minutes summarizing this thread and create a JIRA ticket? That way it won't get forgotten about, and helps us keep track of things better. [1]
Also for future ref, try and use the FS pastebin for pasting logs otherwise it makes threads a bit hard to read [2]
Thank you!
Cal
[1] http://wiki.freeswitch.org/wiki/Reporting_Bugs#Reporting_A_Bug_With_JIRA [2] http://wiki.freeswitch.org/wiki/Reporting_Bugs#Using_The_Pastebin
On Thu, Jul 25, 2013 at 12:20 AM, Anthony McGarry <agtmcgarry@gmail.com> wrote: Still the same with auth-calls off. Appreciate the time you've spent on this. If you have time to put it in the lab then sure, if not no worries.
On 24 Jul 2013, at 23:50, Anthony Minessale <anthony.minessale@gmail.com> wrote:
is it any different if you turn off auth-calls?
There must be something subtle about it that is not obvious. I may need to have it labbed up where i can actually mess with the code to figure it out.
On Wed, Jul 24, 2013 at 5:29 PM, Anthony McGarry <agtmcgarry@gmail.com> wrote: http://pastebin.freeswitch.org/21235
On 24 Jul 2013, at 23:17, Anthony Minessale <anthony.minessale@gmail.com> wrote:
can you do that again with this extra command right before recover is executed?
sofia loglevel all 9
maybe its failing on tcp and reverting to udp
On Wed, Jul 24, 2013 at 5:01 PM, Anthony McGarry <agtmcgarry@gmail.com> wrote: ok so I added INVITE sip:+353877857933@voice.plannet21.ie:5060;transport=tcp SIP/2.0 to all outgoing INVITES from my client.
Uploaded logs from both servers
before crash http://pastebin.freeswitch.org/21233
after crash http://pastebin.freeswitch.org/21234
On 24 Jul 2013, at 22:13, Anthony Minessale <anthony.minessale@gmail.com> wrote:
I think I would need to see the whole sip trace of the original invite and the recover leg. The only diff I see between mine and yours is mine is using FS on both sides and its including transport=tcp
On Wed, Jul 24, 2013 at 4:07 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote: maybe its because transport=tcp is not there in the req invite, I'll see if I can force that into the url when applicable. The via from the 200ok is the one that is preserved for the recover invite. This is all a bit of trickery cos the sofia stack has no idea its doing a recovery it thinks its a new outbound uac call. We someday need to change the code to allow us to create a uas instance on demand from the dialog data we have.
On Wed, Jul 24, 2013 at 3:49 PM, Anthony McGarry <agtmcgarry@gmail.com> wrote: 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 Route: <sip:1004@1.x.x.x:50938;transport=tcp>
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.
Route: <sip:+35314611947@198.19.255.1:61767;transport=tcp>
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 > http://pastebin.com/0xe0QyFC > > recovered call > http://pastebin.com/ByjJ4nhf > > > 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. >> > >> > >> > _________________________________________________________________________ >> > 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 >> >> >> _________________________________________________________________________ >> 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 >> >> _________________________________________________________________________ >> 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 > > > _________________________________________________________________________ > 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 > > > > > -- > 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
_________________________________________________________________________ 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
-- 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
_________________________________________________________________________ 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
-- 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
-- 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
_________________________________________________________________________ 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
-- 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
_________________________________________________________________________ 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
-- 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
_________________________________________________________________________ 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
_________________________________________________________________________ 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
_________________________________________________________________________ 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
-- 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
_________________________________________________________________________ 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
Hi Anthony, call recovery is only possible with UDP. -- On Tuesday, July 23, 2013 at 9:01 AM, Anthony McGarry wrote:
Hi,
FreeSWITCH (Version 1.2.11 git c9790ea 2013-07-16 02:46:21Z)
I have two physical boxes active/passive using heartbeat and corosync. Using FS as an SBC with two profiles. Internal and Private. Both profiles are using track-calls for call recovery.
Works fine when using UDP as the transport for both legs. Works fine when A leg uses UDP and B leg uses TCP Fails when A leg uses TCP and b leg uses either UDP or TCP.
It seems that the A leg always uses UDP as the transport for the INVITE to recover. Even if the initial call used TCP before the failure.
Anyone using TCP in this scenario? Cannot find what I'm missing.
Thanks A
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org (mailto:consulting@freeswitch.org) http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
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participants (6)
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Anthony McGarry
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Anthony Minessale
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Brian West
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Cal Leeming [Simplicity Media Ltd]
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Gilad Abada
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Steven Ayre