<html><head><meta http-equiv="Content-Type" content="text/html charset=iso-8859-1"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">Its the same call. Two way audio, sup services etc..<div><br><div><div>On 23 Jul 2013, at 16:39, Steven Ayre <<a href="mailto:steveayre@gmail.com">steveayre@gmail.com</a>> wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite">Is the B-leg the same call, or a new call?<div><br></div><div><br><br><div class="gmail_quote">On 23 July 2013 16:31, Anthony McGarry <span dir="ltr"><<a href="mailto:agtmcgarry@gmail.com" target="_blank">agtmcgarry@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Thanks Brian,<br>
<br>
Initially I though the same and looked for something to migrate the tcp session, tcpcp and sockmi, but no joy.<br>
<br>
I was doing some more testing and noticed that if the B leg was TCP and the A leg UDP the call recovered.<br>
<br>
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.<br>
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.<br>
<br>
Below is a call with A leg as UDP and B leg as TCP thats recovers fine.<br>
<br>
2013-07-23 16:19:46.419717 [NOTICE] switch_channel.c:1030 New Channel sofia/private/<a href="tel:%2B35319032109" value="+35319032109">+35319032109</a>@198<br>
2013-07-23 16:19:46.419717 [NOTICE] switch_channel.c:1028 Rename Channel sofia/private/<a href="tel:%2B35319032109" value="+35319032109">+35319032109</a>@<br>
2013-07-23 16:19:46.419717 [NOTICE] switch_core_sqldb.c:2744 Resurrecting fallen channel sofia/priva<br>
2013-07-23 16:19:46.439717 [NOTICE] switch_channel.c:1030 New Channel <a href="mailto:sofia/internal/0877857933@10.1">sofia/internal/0877857933@10.1</a><br>
2013-07-23 16:19:46.439717 [NOTICE] switch_channel.c:1028 Rename Channel sofia/internal/0877857933@1<br>
2013-07-23 16:19:46.439717 [NOTICE] switch_core_sqldb.c:2744 Resurrecting fallen channel sofia/inter<br>
send 1110 bytes to udp/[10.101.23.203]:5060 at 15:19:46.459562:<br>
------------------------------------------------------------------------<br>
INVITE <a href="http://sip:0877857933@10.101.23.203:5060/" target="_blank">sip:0877857933@10.101.23.203:5060</a> SIP/2.0<br>
Via: SIP/2.0/UDP 10.101.23.110;rport;branch=z9hG4bK3tjcc8KU636FS<br>
Route: <<a href="sip:0877857933@10.101.23.203:5060;lr">sip:0877857933@10.101.23.203:5060;lr</a>><br>
Max-Forwards: 70<br>
From: <<a href="http://sip:ae019032109@10.101.23.110:5060/" target="_blank">sip:ae019032109@10.101.23.110:5060</a>>;tag=B8BUB47FmmerF<br>
To: <<a href="mailto:sip%3A0877857933@10.101.23.203">sip:0877857933@10.101.23.203</a>>;tag=95ffcd055e0f78f7d5d397020e89288dba056a96<br>
Call-ID: 1054-453-6232013151830-IMG2_DEG-2-10.101.23.203<br>
CSeq: 46960329 INVITE<br>
Contact: <<a href="http://sip:0877857933@10.101.23.110:5060/" target="_blank">sip:0877857933@10.101.23.110:5060</a>><br>
User-Agent: LAB - SBC<br>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY<br>
Supported: timer, precondition, path, replaces<br>
Allow-Events: talk, hold, conference, refer<br>
Privacy: none<br>
Content-Type: application/sdp<br>
Content-Disposition: session<br>
Content-Length: 246<br>
X-FS-Support: update_display,send_info<br>
P-Asserted-Identity: "ae019032109" <<a href="mailto:sip%3Aae019032109@10.101.23.203">sip:ae019032109@10.101.23.203</a>><br>
<br>
v=0<br>
o=FreeSWITCH 1374564092 1374564094 IN IP4 10.101.24.110<br>
s=FreeSWITCH<br>
c=IN IP4 10.101.24.110<br>
t=0 0<br>
m=audio 28694 RTP/AVP 8 101 13<br>
a=rtpmap:8 PCMA/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=rtpmap:13 CN/8000<br>
a=ptime:20<br>
------------------------------------------------------------------------<br>
recv 398 bytes from udp/[10.101.23.203]:5060 at 15:19:46.463116:<br>
------------------------------------------------------------------------<br>
SIP/2.0 100 Trying<br>
Via: SIP/2.0/UDP 10.101.23.110;rport;branch=z9hG4bK3tjcc8KU636FS;received=10.101.23.110<br>
Call-ID: 1054-453-6232013151830-IMG2_DEG-2-10.101.23.203<br>
From: <<a href="http://sip:ae019032109@10.101.23.110:5060/" target="_blank">sip:ae019032109@10.101.23.110:5060</a>>;tag=B8BUB47FmmerF<br>
To: <<a href="mailto:sip%3A0877857933@10.101.23.203">sip:0877857933@10.101.23.203</a>>;tag=95ffcd055e0f78f7d5d397020e89288dba056a96<br>
CSeq: 46960329 INVITE<br>
Server: Dialogic-SIP/10.5.3.372 IMG2_DEG 2<br>
Content-Length: 0<br>
<br>
------------------------------------------------------------------------<br>
recv 961 bytes from udp/[10.101.23.203]:5060 at 15:19:46.463966:<br>
------------------------------------------------------------------------<br>
SIP/2.0 200 OK<br>
Via: SIP/2.0/UDP 10.101.23.110;rport;branch=z9hG4bK3tjcc8KU636FS;received=10.101.23.110<br>
Call-ID: 1054-453-6232013151830-IMG2_DEG-2-10.101.23.203<br>
From: <<a href="http://sip:ae019032109@10.101.23.110:5060/" target="_blank">sip:ae019032109@10.101.23.110:5060</a>>;tag=B8BUB47FmmerF<br>
To: <<a href="mailto:sip%3A0877857933@10.101.23.203">sip:0877857933@10.101.23.203</a>>;tag=95ffcd055e0f78f7d5d397020e89288dba056a96<br>
Contact: <<a href="http://sip:0877857933@10.101.23.203:5060/" target="_blank">sip:0877857933@10.101.23.203:5060</a>><br>
CSeq: 46960329 INVITE<br>
Server: Dialogic-SIP/10.5.3.372 IMG2_DEG 2<br>
Allow: INVITE, BYE, REGISTER, ACK, OPTIONS, CANCEL, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE<br>
Supported: path, replaces, timer, tdialog<br>
Require: timer<br>
Session-Expires: 1800;refresher=uas<br>
Accept: application/sdp, application/dtmf-relay, text/plain<br>
Content-Type: application/sdp<br>
Content-Length: 239<br>
<br>
v=0<br>
o=Dialogic_SDP 963778 1 IN IP4 10.101.23.203<br>
s=Dialogic-SIP<br>
c=IN IP4 10.101.24.203<br>
t=0 0<br>
m=audio 8332 RTP/AVP 8 101<br>
a=rtpmap:8 PCMA/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-15<br>
a=silenceSupp:off - - - -<br>
a=ptime:20<br>
------------------------------------------------------------------------<br>
recv 961 bytes from udp/[10.101.23.203]:5060 at 15:19:47.101507:<br>
------------------------------------------------------------------------<br>
SIP/2.0 200 OK<br>
Via: SIP/2.0/UDP 10.101.23.110;rport;branch=z9hG4bK3tjcc8KU636FS;received=10.101.23.110<br>
Call-ID: 1054-453-6232013151830-IMG2_DEG-2-10.101.23.203<br>
From: <<a href="http://sip:ae019032109@10.101.23.110:5060/" target="_blank">sip:ae019032109@10.101.23.110:5060</a>>;tag=B8BUB47FmmerF<br>
To: <<a href="mailto:sip%3A0877857933@10.101.23.203">sip:0877857933@10.101.23.203</a>>;tag=95ffcd055e0f78f7d5d397020e89288dba056a96<br>
Contact: <<a href="http://sip:0877857933@10.101.23.203:5060/" target="_blank">sip:0877857933@10.101.23.203:5060</a>><br>
CSeq: 46960329 INVITE<br>
Server: Dialogic-SIP/10.5.3.372 IMG2_DEG 2<br>
Allow: INVITE, BYE, REGISTER, ACK, OPTIONS, CANCEL, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE<br>
Supported: path, replaces, timer, tdialog<br>
Require: timer<br>
Session-Expires: 1800;refresher=uas<br>
Accept: application/sdp, application/dtmf-relay, text/plain<br>
Content-Type: application/sdp<br>
Content-Length: 239<br>
<br>
v=0<br>
o=Dialogic_SDP 963778 1 IN IP4 10.101.23.203<br>
s=Dialogic-SIP<br>
c=IN IP4 10.101.24.203<br>
t=0 0<br>
m=audio 8332 RTP/AVP 8 101<br>
a=rtpmap:8 PCMA/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-15<br>
a=silenceSupp:off - - - -<br>
a=ptime:20<br>
------------------------------------------------------------------------<br>
recv 961 bytes from udp/[10.101.23.203]:5060 at 15:19:48.070746:<br>
------------------------------------------------------------------------<br>
SIP/2.0 200 OK<br>
Via: SIP/2.0/UDP 10.101.23.110;rport;branch=z9hG4bK3tjcc8KU636FS;received=10.101.23.110<br>
Call-ID: 1054-453-6232013151830-IMG2_DEG-2-10.101.23.203<br>
From: <<a href="http://sip:ae019032109@10.101.23.110:5060/" target="_blank">sip:ae019032109@10.101.23.110:5060</a>>;tag=B8BUB47FmmerF<br>
To: <<a href="mailto:sip%3A0877857933@10.101.23.203">sip:0877857933@10.101.23.203</a>>;tag=95ffcd055e0f78f7d5d397020e89288dba056a96<br>
Contact: <<a href="http://sip:0877857933@10.101.23.203:5060/" target="_blank">sip:0877857933@10.101.23.203:5060</a>><br>
CSeq: 46960329 INVITE<br>
Server: Dialogic-SIP/10.5.3.372 IMG2_DEG 2<br>
Allow: INVITE, BYE, REGISTER, ACK, OPTIONS, CANCEL, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE<br>
Supported: path, replaces, timer, tdialog<br>
Require: timer<br>
Session-Expires: 1800;refresher=uas<br>
Accept: application/sdp, application/dtmf-relay, text/plain<br>
Content-Type: application/sdp<br>
Content-Length: 239<br>
<br>
v=0<br>
o=Dialogic_SDP 963778 1 IN IP4 10.101.23.203<br>
s=Dialogic-SIP<br>
c=IN IP4 10.101.24.203<br>
t=0 0<br>
m=audio 8332 RTP/AVP 8 101<br>
a=rtpmap:8 PCMA/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-15<br>
a=silenceSupp:off - - - -<br>
a=ptime:20<br>
------------------------------------------------------------------------<br>
send 1175 bytes to tcp/[198.19.255.1]:5060 at 15:19:49.473346:<br>
------------------------------------------------------------------------<br>
INVITE <a href="sip:+35319032109@198.19.255.1:5060;transport=tcp">sip:+35319032109@198.19.255.1:5060;transport=tcp</a> SIP/2.0<br>
Via: SIP/2.0/TCP <a href="tel:78.158.110.24" value="+17815811024">78.158.110.24</a>;rport;branch=z9hG4bKBNF06yNpU187Q<br>
Via: SIP/2.0/TCP <a href="tel:78.158.110.24" value="+17815811024">78.158.110.24</a>;rport;branch=z9hG4bKZ21tSFUQvy38c<br>
Max-Forwards: 69<br>
From: "+353877857933" <<a href="mailto:sip%3A%2B353877857933@78.158.110.24">sip:+353877857933@78.158.110.24</a>>;tag=BK8QvjBeDHXtN<br>
To: <<a href="mailto:sip%3A%2B35319032109@198.19.255.1">sip:+35319032109@198.19.255.1</a>;transport=tcp>;tag=397C7E04-26CA<br>
Call-ID: f9547f1f-6e4d-1231-028b-001f290685a4<br>
CSeq: 46960329 INVITE<br>
Contact: <<a href="sip:+35319032109@78.158.110.24:5060;transport=tcp">sip:+35319032109@78.158.110.24:5060;transport=tcp</a>><br>
User-Agent: PlanNet21 Communications - SBC<br>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY<br>
Supported: timer, precondition, path, replaces<br>
Allow-Events: talk, hold, conference, refer<br>
Privacy: none<br>
Content-Type: application/sdp<br>
Content-Disposition: session<br>
Content-Length: 246<br>
X-FS-Support: update_display,send_info<br>
P-Asserted-Identity: "<a href="tel:%2B353877857933" value="+353877857933">+353877857933</a>" <<a href="mailto:sip%3A%2B353877857933@78.158.110.24">sip:+353877857933@78.158.110.24</a>><br>
<br>
v=0<br>
o=FreeSWITCH 1374569678 1374569680 IN IP4 <a href="tel:78.158.110.24" value="+17815811024">78.158.110.24</a><br>
s=FreeSWITCH<br>
c=IN IP4 <a href="tel:78.158.110.24" value="+17815811024">78.158.110.24</a><br>
t=0 0<br>
m=audio 23108 RTP/AVP 8 101 13<br>
a=rtpmap:8 PCMA/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=rtpmap:13 CN/8000<br>
a=ptime:20<br>
------------------------------------------------------------------------<br>
recv 484 bytes from tcp/[198.19.255.1]:5060 at 15:19:49.536636:<br>
------------------------------------------------------------------------<br>
SIP/2.0 100 Trying<br>
Via: SIP/2.0/TCP 78.158.110.24;rport;branch=z9hG4bKBNF06yNpU187Q,SIP/2.0/TCP 78.158.110.24;rport;<br>
From: "+353877857933" <<a href="mailto:sip%3A%2B353877857933@78.158.110.24">sip:+353877857933@78.158.110.24</a>>;tag=BK8QvjBeDHXtN<br>
To: <<a href="mailto:sip%3A%2B35319032109@198.19.255.1">sip:+35319032109@198.19.255.1</a>;transport=tcp>;tag=397C7E04-26CA<br>
Date: Tue, 23 Jul 2013 15:19:49 GMT<br>
Call-ID: f9547f1f-6e4d-1231-028b-001f290685a4<br>
CSeq: 46960329 INVITE<br>
Allow-Events: telephone-event<br>
Server: Cisco-SIPGateway/IOS-12.x<br>
Content-Length: 0<br>
<br>
------------------------------------------------------------------------<br>
recv 1073 bytes from tcp/[198.19.255.1]:5060 at 15:19:49.601460:<br>
------------------------------------------------------------------------<br>
SIP/2.0 200 OK<br>
Via: SIP/2.0/TCP 78.158.110.24;rport;branch=z9hG4bKBNF06yNpU187Q,SIP/2.0/TCP 78.158.110.24;rport;<br>
From: "+353877857933" <<a href="mailto:sip%3A%2B353877857933@78.158.110.24">sip:+353877857933@78.158.110.24</a>>;tag=BK8QvjBeDHXtN<br>
To: <<a href="mailto:sip%3A%2B35319032109@198.19.255.1">sip:+35319032109@198.19.255.1</a>;transport=tcp>;tag=397C7E04-26CA<br>
Date: Tue, 23 Jul 2013 15:19:49 GMT<br>
Call-ID: f9547f1f-6e4d-1231-028b-001f290685a4<br>
CSeq: 46960329 INVITE<br>
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER<br>
Allow-Events: telephone-event<br>
Remote-Party-ID: "Ian McGrath" <<a href="mailto:sip%3A%2B1947@198.19.255.1">sip:+1947@198.19.255.1</a>>;party=called;screen=yes;privacy=off<br>
Contact: <sip:<a href="tel:%2B35319032109" value="+35319032109">+35319032109</a>@198.19.255.1:5060;transport=tcp><br>
Supported: replaces<br>
Supported: sdp-anat<br>
Server: Cisco-SIPGateway/IOS-12.x<br>
Supported: timer<br>
Content-Type: application/sdp<br>
Content-Length: 247<br>
<br>
v=0<br>
o=CiscoSystemsSIP-GW-UserAgent 3902 8614 IN IP4 198.19.255.1<br>
s=SIP Call<br>
c=IN IP4 198.19.255.1<br>
t=0 0<br>
m=audio 19050 RTP/AVP 8 101<br>
c=IN IP4 198.19.255.1<br>
a=rtpmap:8 PCMA/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=ptime:20<br>
------------------------------------------------------------------------<br>
send 440 bytes to tcp/[198.19.255.1]:5060 at 15:19:49.603288:<br>
------------------------------------------------------------------------<br>
ACK <a href="sip:+35319032109@198.19.255.1:5060;transport=tcp">sip:+35319032109@198.19.255.1:5060;transport=tcp</a> SIP/2.0<br>
Via: SIP/2.0/TCP <a href="tel:78.158.110.24" value="+17815811024">78.158.110.24</a>;rport;branch=z9hG4bKcy8r8S6SraZtK<br>
Max-Forwards: 70<br>
From: "<a href="tel:%2B353877857933" value="+353877857933">+353877857933</a>" <<a href="mailto:sip%3A%2B353877857933@78.158.110.24">sip:+353877857933@78.158.110.24</a>>;tag=BK8QvjBeDHXtN<br>
To: <<a href="mailto:sip%3A%2B35319032109@198.19.255.1">sip:+35319032109@198.19.255.1</a>;transport=tcp>;tag=397C7E04-26CA<br>
Call-ID: f9547f1f-6e4d-1231-028b-001f290685a4<br>
CSeq: 46960329 ACK<br>
Contact: <<a href="sip:+35319032109@78.158.110.24:5060;transport=tcp">sip:+35319032109@78.158.110.24:5060;transport=tcp</a>><br>
Content-Length: 0<br>
<br>
------------------------------------------------------------------------<br>
send 466 bytes to udp/[10.101.23.203]:5060 at 15:19:49.606747:<br>
------------------------------------------------------------------------<br>
ACK <a href="http://sip:0877857933@10.101.23.203:5060/" target="_blank">sip:0877857933@10.101.23.203:5060</a> SIP/2.0<br>
Via: SIP/2.0/UDP 10.101.23.110;rport;branch=z9hG4bK43B5D34y3cX2m<br>
Route: <<a href="sip:0877857933@10.101.23.203:5060;lr">sip:0877857933@10.101.23.203:5060;lr</a>><br>
Max-Forwards: 70<br>
From: <<a href="http://sip:ae019032109@10.101.23.110:5060/" target="_blank">sip:ae019032109@10.101.23.110:5060</a>>;tag=B8BUB47FmmerF<br>
To: <<a href="mailto:sip%3A0877857933@10.101.23.203">sip:0877857933@10.101.23.203</a>>;tag=95ffcd055e0f78f7d5d397020e89288dba056a96<br>
Call-ID: 1054-453-6232013151830-IMG2_DEG-2-10.101.23.203<br>
CSeq: 46960329 ACK<br>
Contact: <<a href="http://sip:0877857933@10.101.23.110:5060/" target="_blank">sip:0877857933@10.101.23.110:5060</a>><br>
Content-Length: 0<br>
<br>
------------------------------------------------------------------------<br>
<div class="HOEnZb"><div class="h5"><br>
<br>
<br>
<br>
On 23 Jul 2013, at 14:47, Brian West <<a href="mailto:brian@freeswitch.org">brian@freeswitch.org</a>> wrote:<br>
<br>
> 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.<br>
><br>
> /b<br>
><br>
> Em Jul 23, 2013, ās 8:01 AM, Anthony McGarry <<a href="mailto:agtmcgarry@gmail.com">agtmcgarry@gmail.com</a>> escreveu:<br>
><br>
>> Anyone using TCP in this scenario? Cannot find what I'm missing.<br>
><br>
><br>
> _________________________________________________________________________<br>
> Professional FreeSWITCH Consulting Services:<br>
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><br>
> FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>
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><br>
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><br>
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<br>
<br>
_________________________________________________________________________<br>
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<br>
Official FreeSWITCH Sites<br>
<a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org</a><br>
<a href="http://wiki.freeswitch.org/" target="_blank">http://wiki.freeswitch.org</a><br>
<a href="http://www.cluecon.com/" target="_blank">http://www.cluecon.com</a><br>
<br>
FreeSWITCH-users mailing list<br>
<a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>
<a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
<a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org</a><br>
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_________________________________________________________________________<br>Professional FreeSWITCH Consulting Services:<br><a href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a><br>http://www.freeswitchsolutions.com<br><br>FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>http://www.cudatel.com<br><br>Official FreeSWITCH Sites<br>http://www.freeswitch.org<br>http://wiki.freeswitch.org<br>http://www.cluecon.com<br><br>FreeSWITCH-users mailing list<br>FreeSWITCH-users@lists.freeswitch.org<br>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>http://www.freeswitch.org<br></blockquote></div><br></div></body></html>