Hi William,
I would suggest to go back using the $du to force tcp like:
1) in $ru you already have the destination of the call
2) before t_relay() simply do: $du = $ru + ";transport=tcp";
Regards,
Bogdan
On 01/25/2012 03:09 AM, William Edwards wrote:
Thanks for trying. In addition, here's the summary of a packet capture
of me trying to call myself:
http://pastebin.com/raw.php?i=3iczgtXb
X-Lite = x.x.x.x
OpenSIPS = y.y.y.y
FreeSWITCH = z.z.z.z
TCP 1 0.000000 x.x.x.x -> y.y.y.y SIP/SDP Request: INVITE
sip:[email protected];transport=tcp, with session description
TCP 2 0.000042 y.y.y.y -> x.x.x.x TCP sip> 59498 [ACK] Seq=1 Ack=982
Win=864 Len=0
TCP 3 0.000800 y.y.y.y -> x.x.x.x SIP Status: 100 Giving a try
TCP 4 0.000948 y.y.y.y -> z.z.z.z TCP 41695> sip [SYN] Seq=0 Win=5840
Len=0 MSS=1460 TSV=31556561 TSER=0 WS=6
TCP 5 0.001335 z.z.z.z -> y.y.y.y TCP sip> 41695 [SYN, ACK] Seq=0 Ack=1
Win=14480 Len=0 MSS=1460 TSV=31542864 TSER=31556561 WS=6
TCP 6 0.001385 y.y.y.y -> z.z.z.z TCP 41695> sip [ACK] Seq=1 Ack=1
Win=5888 Len=0 TSV=31556562 TSER=31542864
TCP 7 0.001696 y.y.y.y -> z.z.z.z SIP/SDP Request: INVITE
sip:[email protected];transport=tcp, with session description
TCP 8 0.001855 z.z.z.z -> y.y.y.y TCP sip> 41695 [ACK] Seq=1 Ack=1233
Win=16960 Len=0 TSV=31542865 TSER=31556562
TCP 9 0.002719 z.z.z.z -> y.y.y.y SIP Status: 100 Trying
TCP 10 0.002732 y.y.y.y -> z.z.z.z TCP 41695> sip [ACK] Seq=1233 Ack=491
Win=6912 Len=0 TSV=31556563 TSER=31542865
TCP 11 0.215286 x.x.x.x -> y.y.y.y TCP 59498> sip [ACK] Seq=982 Ack=325
Win=4150 Len=0
TCP 12 0.712565 z.z.z.z -> y.y.y.y TCP 48269> sip [SYN] Seq=0 Win=14600
Len=0 MSS=1460 TSV=31543575 TSER=0 WS=6
TCP 13 0.712628 y.y.y.y -> z.z.z.z TCP sip> 48269 [SYN, ACK] Seq=0 Ack=1
Win=5792 Len=0 MSS=1460 TSV=31557273 TSER=31543575 WS=6
TCP 14 0.712944 z.z.z.z -> y.y.y.y TCP 48269> sip [ACK] Seq=1 Ack=1
Win=14656 Len=0 TSV=31543576 TSER=31557273
TCP 15 0.712997 z.z.z.z -> y.y.y.y SIP/SDP Request: INVITE
sip:[email protected]:59498;rinstance=2e3da39d713a676d;transport=TCP, with session
description
TCP 16 0.713009 y.y.y.y -> z.z.z.z TCP sip> 48269 [ACK] Seq=1 Ack=1383
Win=8704 Len=0 TSV=31557273 TSER=31543576
TCP 17 0.714476 y.y.y.y -> z.z.z.z SIP Status: 100 Giving a try
TCP 18 0.714635 z.z.z.z -> y.y.y.y TCP 48269> sip [ACK] Seq=1383 Ack=315
Win=15680 Len=0 TSV=31543577 TSER=31557275
UDP 19 0.714713 y.y.y.y -> x.x.x.x SIP/SDP Request: INVITE
sip:[email protected]:59498;rinstance=2e3da39d713a676d;transport=TCP, with session
description
UDP 20 1.217530 y.y.y.y -> x.x.x.x SIP/SDP Request: INVITE
sip:[email protected]:59498;rinstance=2e3da39d713a676d;transport=TCP, with session
description
UDP 21 2.219539 y.y.y.y -> x.x.x.x SIP/SDP Request: INVITE
sip:[email protected]:59498;rinstance=2e3da39d713a676d;transport=TCP, with session
description
You can see that OpenSIPS is trying to send the request as UDP even
though it originally came from FreeSWITCH as TCP.
Regards,
William
On Tue, Jan 24, 2012 at 4:28 PM, <[email protected]
<mailto:[email protected]>> wrote:
Well hell. Thats all I know to do just from looking at the
documentation. Hopefully one of the other guys can help when they
wake up in their respective timezones.
Sorry I couldn't help.
On , William Edwards <[email protected]
<mailto:[email protected]>> wrote:
> Hey Duane,
>
> Yes, I tried using force_send_socket, but it still sends the
request as UDP instead of the original TCP. I noticed in the
documentation it says that if the protocol doesn't match, the
closest socket of the same protocol is used:
>
>
>
>
>
> "If the protocol doesn't match (e.g. UDP message "forced" to a
TCP socket) the closest socket of the same protocol is used.
>
> "
>
> Regards,
> William
>
>
> On Tue, Jan 24, 2012 at 4:14 PM, [email protected]
<mailto:[email protected]>> wrote:
>
>
>
>
> What about force_send_socket
>
>
> http://www.opensips.org/Resources/DocsCoreFcn#toc111
>
>
>
>
>
> Have you tried that yet.
>
>
>
>
>
> force_send_socket(udp:192.168.1.10:5060
<http://192.168.1.10:5060>);
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> On , William Edwards [email protected]
<mailto:[email protected]>> wrote:
>
>
> > In my attempt to resolve this problem I have also already come
across that article. Unfortunately the t_relay function does not
support variables as parameters, so I am not able to put in the
full uri (e.g. tcp:70.232.x.x:55085) in the function. From this
article
(http://openser.org/pipermail/users/2010-March/011638.html), it
was suggested to rewrite the $du instead, however I don't believe
it is possible to specify the protocol in the $du.
>
>
> >
>
>
> >
>
>
> >
>
>
> >
>
>
> >
>
>
> > Regards,
>
>
> > William
>
>
> >
>
>
> >
>
>
> > On Tue, Jan 24, 2012 at 3:12 PM, [email protected]
<mailto:[email protected]>> wrote:
>
>
> >
>
>
> >
>
>
> >
>
>
> >
>
>
> > I did a search for "tcp" on Nabble's OpenSIPS section and just
found this
>
>
> >
>
>
> >
http://opensips-open-sip-server.1449251.n2.nabble.com/How-to-change-transport-protocol-of-R-URI-td6376471.html
>
>
> >
>
>
> >
>
>
> >
>
>
> >
>
>
> >
>
>
> > I would say you need to use t_relay() with parameters.
>
>
> >
>
>
> > Hope that helps.
>
>
> >
>
>
> >
>
>
> > On , William Edwards [email protected]
<mailto:[email protected]>> wrote:
>
>
> >
>
>
> >
>
>
> >
>
>
> >
>
>
> > > Hi,
>
>
> > >
>
>
> > > Thanks for your reply. I've looked into $fs, but I haven't
been able to find any concise documentation about how I can use it
to force the protocol to TCP. It's really unclear as to why
OpenSIPS would be rewritting the TCP message to UDP in the first
place. Is there no way to configure OpenSIPS to relay the message
using the same protocol that it was received?
>
>
> >
>
>
> >
>
>
> >
>
>
> >
>
>
> > >
>
>
> > >
>
>
> > >
>
>
> > > Regards,
>
>
> > > William
>
>
> > >
>
>
> > >
>
>
> > > On Tue, Jan 24, 2012 at 10:26 AM, [email protected]
<mailto:[email protected]>> wrote:
>
>
> > >
>
>
> >
>
>
> >
>
>
> >
>
>
> >
>
>
> > >
>
>
> > > Perhaps you need this somewhere in your script
>
>
> > >
>
>
> > > $fs Forced socket:
http://www.opensips.org/Resources/DocsCoreVar16#toc38
>
>
> >
>
>
> >
>
>
> >
>
>
> >
>
>
> > >
>
>
> > >
>
>
> > >
>
>
> > >
>
>
> > >
>
>
> > >
>
>
> > > On , William Edwards [email protected]
<mailto:[email protected]>> wrote:
>
>
> > > > Hi,
>
>
> > > >
>
>
> > > > We currently have two Freeswitch boxes behind OpenSIPS which
>
>
> >
>
>
> >
>
>
> >
>
>
> >
>
>
> > >
>
>
> > >
>
>
> > > > is acting as a SIP proxy/load balancer. Right now we're
facing a problem
>
>
> > > > where OpenSIPS is rewriting incoming INVITE messages from our
>
>
> > > > Freeswitch servers from TCP to UDP. This, of course, is
causing a
>
>
> >
>
>
> >
>
>
> >
>
>
> >
>
>
> > >
>
>
> > >
>
>
> > > > problem because the client has registered using TCP, and
incoming calls
>
>
> > > > are being sent to it over UDP.
>
>
> > > >
>
>
> > > >
>
>
> > > > This is an overview of what is happening:
>
>
> >
>
>
> >
>
>
> >
>
>
> >
>
>
> > > > Freeswitch -TCP-> OpenSIPS -UDP-> Client UA (listening on
TCP)
>
>
> > >
>
>
> > >
>
>
> > > >
>
>
> > > > We would like OpenSIPS to retain the original protocol
that the message was received. Is there any way this is possible?
>
>
> >
>
>
> >
>
>
> >
>
>
> >
>
>
> > > >
>
>
> > > >
>
>
> > > >
>
>
> > > >
>
>
> > > > My current OpenSIPS configuration can be found here:
>
>
> > >
>
>
> > >
>
>
> > > > http://pastebin.com/DSU1f2VS
>
>
> >
>
>
> >
>
>
> >
>
>
> >
>
>
> > > >
>
>
> > > > Regards,
>
>
> > > > William
>
--
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
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