28 Aug
2008
28 Aug
'08
2:49 p.m.
Hi All
I am using using openser 1.3..if I make a call between two softphones on
the same lan or a a pstn call to my mobile phone..and the
called/receiver party does hang-up the call. It works fine in UDP mode
and the call get's hang-up. However in TLS mode this does not work.
Anything I might have missed here? Since both udp and tls use the same
routes, and voice is fine and no one way audio ..etc.
I notice this http://pastebin.com/m38c979f6 on rtp proxy. However in the
logs of openser I can't see any errors.
However on the wire shark I can see icmp destination unreachable...port
unreachable.
I would have said it is a NAT issue. However it works for simple UDP.
However I did notice the following in the logs
Aug 28 13:41:01 [8565] DBG:tm:set_timer: relative timeout is 4000000
Aug 28 13:41:01 [8565] DBG:tm:insert_timer_unsafe: [7]: 0xb6137794
(45900000)
Aug 28 13:41:01 [8565] DBG:tm:retransmission_handler:
retransmission_handler : done
Aug 28 13:41:02 [8564] DBG:core:parse_msg: SIP Request:
Aug 28 13:41:02 [8564] DBG:core:parse_msg: method: <BYE>
Aug 28 13:41:02 [8564] DBG:core:parse_msg: uri:
<sip:michofr@193.227.186.146:3218;transport=UDP;rinstance=D98C1DD404B200
8F980980E97E42F8EC;nat=yes>
Aug 28 13:41:02 [8564] DBG:core:parse_msg: version: <SIP/2.0>
Aug 28 13:41:02 [8564] DBG:core:parse_headers: flags=2
Aug 28 13:41:02 [8564] DBG:core:parse_via_param: found param type 232,
<branch> =
Shouldn't the transport=TLS ?
Regards
28 Aug
28 Aug
3:47 p.m.
New subject: [Kamailio-Users] FW: Call Hangup from Caller-End not working with TLS
Ali Jawad writes:
However I did notice the following in the logs
Aug 28 13:41:02 [8564] DBG:core:parse_msg: uri:
<sip:michofr@193.227.186.146:3218;transport=UDP;rinstance=D98C1DD404B200
8F980980E97E42F8EC;nat=yes>
the party sending bye should use as request uri contact uri of
the other party that it learned during dialog setup (invite, 200 ok).
check that there was transport=tls.
-- juha
3:54 p.m.
New subject: [Kamailio-Users] FW: Call Hangup from Caller-End not working withTLS
Hi
The returning party in this case would be the PSTN GW and it talks UDP.
I.e.
Softphone --TLS-->Openser ---UDP--->Softphone
Should't Openser be responsible for the translation of these parameters
?
Thanks
-----Original Message-----
From: Juha Heinanen [mailto:jh@tutpro.com]
Sent: 2008-08-28 16:47
To: Ali Jawad
Cc: users(a)lists.kamailio.org
Subject: [Kamailio-Users] FW: Call Hangup from Caller-End not working
withTLS
Ali Jawad writes:
However I did notice the following in the logs
Aug 28 13:41:02 [8564] DBG:core:parse_msg: uri:
<sip:michofr@193.227.186.146:3218;transport=UDP;rinstance=D98C1DD404B200
8F980980E97E42F8EC;nat=yes>
the party sending bye should use as request uri contact uri of
the other party that it learned during dialog setup (invite, 200 ok).
check that there was transport=tls.
-- juha
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
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4:04 p.m.
New subject: [Kamailio-Users] FW: Call Hangup from Caller-End not working withTLS
Ali Jawad writes:
The returning party in this case would be the PSTN GW
and it talks UDP.
I.e.
Softphone --TLS-->Openser ---UDP--->Softphone
Should't Openser be responsible for the translation of these
parameters?
i don't see a PSTN gateway anywhere in the figure. opener does not
mess with transport parameters, it uses on the outgoing hop whatever
sender specified as i already mentioned (of course provided that openser
in the figure did record-route the initial request.
-- juha
4:05 p.m.
New subject: [Kamailio-Users] FW: Call Hangup from Caller-End not working withTLS
Sorry my fault..I think got brain busted..
The picture should be
Softphone --TLS-->Openser ---UDP-->PSTNGW
From: Juha Heinanen [mailto:jh@tutpro.com]
Sent: 2008-08-28 17:04
To: Ali Jawad
Cc: users(a)lists.kamailio.org
Subject: RE: [Kamailio-Users] FW: Call Hangup from Caller-End not
working withTLS
Ali Jawad writes:
The returning party in this case would be the PSTN GW
and it talks
UDP.
I.e.
Softphone --TLS-->Openser ---UDP--->Softphone
Should't Openser be responsible for the translation of these
parameters?
i don't see a PSTN gateway anywhere in the figure. opener does not
mess with transport parameters, it uses on the outgoing hop whatever
sender specified as i already mentioned (of course provided that openser
in the figure did record-route the initial request.
-- juha
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.
4:07 p.m.
New subject: [Kamailio-Users] FW: Call Hangup from Caller-End not working withTLS
Ali Jawad writes:
Sorry my fault..I think got brain busted..
The picture should be
Softphone --TLS-->Openser ---UDP-->PSTNGW
that does not change my response. softphone sends bye to openser using
tls and openser sends bye to pstngw using whatever transport protocol
request uri specifies.
-- juha
4:10 p.m.
New subject: [Kamailio-Users] FW: Call Hangup from Caller-End not working withTLS
True..However my problem occurs when the mobile phone closes the call,
so in this case PSTN GW send's it in UDP to Openser, and Openser will
send it in TLS to the softphone, right ?
If that is true..where could the culprit be ?
Thanks
Fax: +961 1 375554
-----Original Message-----
From: Juha Heinanen [mailto:jh@tutpro.com]
Sent: 2008-08-28 17:08
To: Ali Jawad
Cc: users(a)lists.kamailio.org
Subject: RE: [Kamailio-Users] FW: Call Hangup from Caller-End not
working withTLS
Ali Jawad writes:
Sorry my fault..I think got brain busted..
The picture should be
Softphone --TLS-->Openser ---UDP-->PSTNGW
that does not change my response. softphone sends bye to openser using
tls and openser sends bye to pstngw using whatever transport protocol
request uri specifies.
-- juha
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.
4:28 p.m.
New subject: [Kamailio-Users] FW: Call Hangup from Caller-End not working withTLS
Ali Jawad writes:
True..However my problem occurs when the mobile phone
closes the call,
so in this case PSTN GW send's it in UDP to Openser, and Openser will
send it in TLS to the softphone, right ?
yes, it request uri from pstn gw specifies tls transport.
-- juha
5:13 p.m.
Hi!
Send us an ngrep dump: ngrep -P "" -W byline port 5060
Although this will show us only the UDP part (as TLS is encrypted) but
may still show as the problem.
Which SIP client do you use?
regards
klaus
Ali Jawad schrieb:
Hi All
I am using using openser 1.3..if I make a call between two softphones on
the same lan or a a pstn call to my mobile phone..and the
called/receiver party does hang-up the call. It works fine in UDP mode
and the call get’s hang-up. However in TLS mode this does not work.
Anything I might have missed here? Since both udp and tls use the same
routes, and voice is fine and no one way audio ..etc.
I notice this http://pastebin.com/m38c979f6 on rtp proxy. However in the
logs of openser I can’t see any errors.
However on the wire shark I can see icmp destination unreachable…port
unreachable.
I would have said it is a NAT issue. However it works for simple UDP.
However I did notice the following in the logs
Aug 28 13:41:01 [8565] DBG:tm:set_timer: relative timeout is 4000000
Aug 28 13:41:01 [8565] DBG:tm:insert_timer_unsafe: [7]: 0xb6137794
(45900000)
Aug 28 13:41:01 [8565] DBG:tm:retransmission_handler:
retransmission_handler : done
Aug 28 13:41:02 [8564] DBG:core:parse_msg: SIP Request:
Aug 28 13:41:02 [8564] DBG:core:parse_msg: method: <BYE>
Aug 28 13:41:02 [8564] DBG:core:parse_msg: uri:
<sip:michofr@193.227.186.146:3218;transport=UDP;rinstance=D98C1DD404B2008F980980E97E42F8EC;nat=yes>
Aug 28 13:41:02 [8564] DBG:core:parse_msg: version: <SIP/2.0>
Aug 28 13:41:02 [8564] DBG:core:parse_headers: flags=2
Aug 28 13:41:02 [8564] DBG:core:parse_via_param: found param type 232,
<branch> =
Shouldn’t the transport=TLS ?
Regards
------------------------------------------------------------------------
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Users mailing list
Users(a)lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
29 Aug
29 Aug
9:31 a.m.
New subject: [Kamailio-Users] FW: Call Hangup from Caller-End not working with TLS
Hi
Please note that UDP is working just fine, however I will try to
duplicate this on TCP, I do use two clients..one custom built and
eyebeam.
Regards
-----Original Message-----
From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at]
Sent: 2008-08-28 18:13
To: Ali Jawad
Cc: users(a)lists.kamailio.org
Subject: Re: [Kamailio-Users] FW: Call Hangup from Caller-End not
working with TLS
Hi!
Send us an ngrep dump: ngrep -P "" -W byline port 5060
Although this will show us only the UDP part (as TLS is encrypted) but
may still show as the problem.
Which SIP client do you use?
regards
klaus
Ali Jawad schrieb:
Hi All
I am using using openser 1.3..if I make a call between two softphones
on
the same lan or a a pstn call to my mobile phone..and
the
called/receiver party does hang-up the call. It works fine in UDP mode
and the call get's hang-up. However in TLS mode
this does not work.
Anything I might have missed here? Since both udp and tls use the same
routes, and voice is fine and no one way audio ..etc.
I notice this http://pastebin.com/m38c979f6 on rtp proxy. However in
the
logs of openser I can't see any errors.
However on the wire shark I can see icmp destination
unreachable...port
unreachable.
I would have said it is a NAT issue. However it works for simple UDP.
However I did notice the following in the logs
Aug 28 13:41:01 [8565] DBG:tm:set_timer: relative timeout is 4000000
Aug 28 13:41:01 [8565] DBG:tm:insert_timer_unsafe: [7]: 0xb6137794
(45900000)
Aug 28 13:41:01 [8565] DBG:tm:retransmission_handler:
retransmission_handler : done
Aug 28 13:41:02 [8564] DBG:core:parse_msg: SIP Request:
Aug 28 13:41:02 [8564] DBG:core:parse_msg: method: <BYE>
Aug 28 13:41:02 [8564] DBG:core:parse_msg: uri:
<sip:michofr@193.227.186.146:3218;transport=UDP;rinstance=D98C1DD404B200
8F980980E97E42F8EC;nat=yes>
Aug 28 13:41:02 [8564] DBG:core:parse_msg: version: <SIP/2.0>
Aug 28 13:41:02 [8564] DBG:core:parse_headers: flags=2
Aug 28 13:41:02 [8564] DBG:core:parse_via_param: found param type 232,
<branch> =
Shouldn't the transport=TLS ?
Regards
------------------------------------------------------------------------
_______________________________________________
Users mailing list
Users(a)lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
--
This message has been scanned for viruses and
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1:27 p.m.
New subject: [Kamailio-Users] FW: Call Hangup from Caller-End not working with TLS
Dear All
Please find below the call setup from my softphone to my cell phone, The
setup is as follows:
LAN -> Office Gateway <--TLS--->Openser<--UDP-->PSTN GW
I got the trace below by applying
ngrep -W byline -T username -q -d eth0
http://pastebin.com/m2a78a27f
I did the same trace for a udp call and it seemed identical to me, as
you can see in the lower part of the trace that a BYE packet is being
sent to the softphone however the transport is being indicated as UDP
not TLS..is this normal ?
Any clues apart from that ?
Thanks
-----Original Message-----
From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at]
Sent: 2008-08-28 18:13
To: Ali Jawad
Cc: users(a)lists.kamailio.org
Subject: Re: [Kamailio-Users] FW: Call Hangup from Caller-End not
working with TLS
Hi!
Send us an ngrep dump: ngrep -P "" -W byline port 5060
Although this will show us only the UDP part (as TLS is encrypted) but
may still show as the problem.
Which SIP client do you use?
regards
klaus
Ali Jawad schrieb:
Hi All
I am using using openser 1.3..if I make a call between two softphones
on
the same lan or a a pstn call to my mobile phone..and
the
called/receiver party does hang-up the call. It works fine in UDP mode
and the call get's hang-up. However in TLS mode
this does not work.
Anything I might have missed here? Since both udp and tls use the same
routes, and voice is fine and no one way audio ..etc.
I notice this http://pastebin.com/m38c979f6 on rtp proxy. However in
the
logs of openser I can't see any errors.
However on the wire shark I can see icmp destination
unreachable...port
unreachable.
I would have said it is a NAT issue. However it works for simple UDP.
However I did notice the following in the logs
Aug 28 13:41:01 [8565] DBG:tm:set_timer: relative timeout is 4000000
Aug 28 13:41:01 [8565] DBG:tm:insert_timer_unsafe: [7]: 0xb6137794
(45900000)
Aug 28 13:41:01 [8565] DBG:tm:retransmission_handler:
retransmission_handler : done
Aug 28 13:41:02 [8564] DBG:core:parse_msg: SIP Request:
Aug 28 13:41:02 [8564] DBG:core:parse_msg: method: <BYE>
Aug 28 13:41:02 [8564] DBG:core:parse_msg: uri:
<sip:michofr@193.227.186.146:3218;transport=UDP;rinstance=D98C1DD404B200
8F980980E97E42F8EC;nat=yes>
Aug 28 13:41:02 [8564] DBG:core:parse_msg: version: <SIP/2.0>
Aug 28 13:41:02 [8564] DBG:core:parse_headers: flags=2
Aug 28 13:41:02 [8564] DBG:core:parse_via_param: found param type 232,
<branch> =
Shouldn't the transport=TLS ?
Regards
------------------------------------------------------------------------
_______________________________________________
Users mailing list
Users(a)lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.
1:33 p.m.
1. INVITE:
Contact:
<sip:username@IP.OF.LAN.GW:2949;transport=UDP;rinstance=D98C1DD404B2008F980980E97E42F8EC;nat=yes>.
As you see the caller announces UDP as contact. Either a bug in the
caller client or do you rewrite the contact in openser?
Further it is strange that the caller sends frmo 127.0.0.1 (Via header,
SDP) but announces a different IP in contact.
regards
klaus
Ali Jawad schrieb:
Dear All
Please find below the call setup from my softphone to my cell phone, The
setup is as follows:
LAN -> Office Gateway <--TLS--->Openser<--UDP-->PSTN GW
I got the trace below by applying
ngrep -W byline -T username -q -d eth0
http://pastebin.com/m2a78a27f
I did the same trace for a udp call and it seemed identical to me, as
you can see in the lower part of the trace that a BYE packet is being
sent to the softphone however the transport is being indicated as UDP
not TLS..is this normal ?
Any clues apart from that ?
Thanks
-----Original Message-----
From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at]
Sent: 2008-08-28 18:13
To: Ali Jawad
Cc: users(a)lists.kamailio.org
Subject: Re: [Kamailio-Users] FW: Call Hangup from Caller-End not
working with TLS
Hi!
Send us an ngrep dump: ngrep -P "" -W byline port 5060
Although this will show us only the UDP part (as TLS is encrypted) but
may still show as the problem.
Which SIP client do you use?
regards
klaus
Ali Jawad schrieb:
Hi All
I am using using openser 1.3..if I make a call between two softphones
on
the same lan or a a pstn call to my mobile
phone..and the
called/receiver party does hang-up the call. It works fine in UDP mode
and the call get's hang-up. However in TLS
mode this does not work.
Anything I might have missed here? Since both udp and tls use the same
routes, and voice is fine and no one way audio
..etc.
I notice this http://pastebin.com/m38c979f6 on rtp proxy. However in
the
logs of openser I can't see any errors.
However on the wire shark I can see icmp destination
unreachable...port
unreachable.
I would have said it is a NAT issue. However it works for simple UDP.
However I did notice the following in the logs
Aug 28 13:41:01 [8565] DBG:tm:set_timer: relative timeout is 4000000
Aug 28 13:41:01 [8565] DBG:tm:insert_timer_unsafe: [7]: 0xb6137794
(45900000)
Aug 28 13:41:01 [8565] DBG:tm:retransmission_handler:
retransmission_handler : done
Aug 28 13:41:02 [8564] DBG:core:parse_msg: SIP Request:
Aug 28 13:41:02 [8564] DBG:core:parse_msg: method: <BYE>
Aug 28 13:41:02 [8564] DBG:core:parse_msg: uri:
<sip:michofr@193.227.186.146:3218;transport=UDP;rinstance=D98C1DD404B200
8F980980E97E42F8EC;nat=yes>
Aug 28 13:41:02 [8564] DBG:core:parse_msg:
version: <SIP/2.0>
Aug 28 13:41:02 [8564] DBG:core:parse_headers: flags=2
Aug 28 13:41:02 [8564] DBG:core:parse_via_param: found param type 232,
<branch> =
Shouldn't the transport=TLS ?
Regards
------------------------------------------------------------------------
_______________________________________________
Users mailing list
Users(a)lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
2:14 p.m.
New subject: [Kamailio-Users] FW: Call Hangup from Caller-End not working with TLS
Hi Klaus
You are referring to line 17 right ? That part of traffic is from
openser to the pstn gw ..and both of those are UDP..should it be
transport=tls there or transport=udp?
Thanks
With Regards
Ali Jawad
System Administrator
Splendor Telecom (www.splendor.net)
Beirut, Lebanon
Phone: +961 1 373725
Fax: +961 1 375554
-----Original Message-----
From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at]
Sent: 2008-08-29 14:34
To: Ali Jawad
Cc: users(a)lists.kamailio.org
Subject: Re: [Kamailio-Users] FW: Call Hangup from Caller-End not
working with TLS
1. INVITE:
Contact:
<sip:username@IP.OF.LAN.GW:2949;transport=UDP;rinstance=D98C1DD404B2008F
980980E97E42F8EC;nat=yes>.
As you see the caller announces UDP as contact. Either a bug in the
caller client or do you rewrite the contact in openser?
Further it is strange that the caller sends frmo 127.0.0.1 (Via header,
SDP) but announces a different IP in contact.
regards
klaus
Ali Jawad schrieb:
<sip:michofr@193.227.186.146:3218;transport=UDP;rinstance=D98C1DD404B200
------------------------------------------------------------------------
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.
Dear All
Please find below the call setup from my softphone to my cell phone,
The
setup is as follows:
LAN -> Office Gateway <--TLS--->Openser<--UDP-->PSTN GW
I got the trace below by applying
ngrep -W byline -T username -q -d eth0
http://pastebin.com/m2a78a27f
I did the same trace for a udp call and it seemed identical to me, as
you can see in the lower part of the trace that a BYE packet is being
sent to the softphone however the transport is being indicated as UDP
not TLS..is this normal ?
Any clues apart from that ?
Thanks
-----Original Message-----
From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at]
Sent: 2008-08-28 18:13
To: Ali Jawad
Cc: users(a)lists.kamailio.org
Subject: Re: [Kamailio-Users] FW: Call Hangup from Caller-End not
working with TLS
Hi!
Send us an ngrep dump: ngrep -P "" -W byline port 5060
Although this will show us only the UDP part (as TLS is encrypted) but
may still show as the problem.
Which SIP client do you use?
regards
klaus
Ali Jawad schrieb:
mode
Hi All
I am using using openser 1.3..if I make a call between two softphones
on
> the same lan or a a pstn call to my mobile phone..and the
> called/receiver party does hang-up the call. It works fine in UDP
> and the call get's hang-up. However in TLS mode this does not work.
> Anything I might have missed here? Since both udp and tls use the
same
routes, and voice is fine and no one way audio
..etc.
I notice this http://pastebin.com/m38c979f6 on rtp proxy. However in
the
logs of openser I can't see any errors.
However on the wire shark I can see icmp destination
unreachable...port
unreachable.
I would have said it is a NAT issue. However it works for simple UDP.
However I did notice the following in the logs
Aug 28 13:41:01 [8565] DBG:tm:set_timer: relative timeout is 4000000
Aug 28 13:41:01 [8565] DBG:tm:insert_timer_unsafe: [7]: 0xb6137794
(45900000)
Aug 28 13:41:01 [8565] DBG:tm:retransmission_handler:
retransmission_handler : done
Aug 28 13:41:02 [8564] DBG:core:parse_msg: SIP Request:
Aug 28 13:41:02 [8564] DBG:core:parse_msg: method: <BYE>
Aug 28 13:41:02 [8564] DBG:core:parse_msg: uri:
8F980980E97E42F8EC;nat=yes>
> Aug 28 13:41:02 [8564] DBG:core:parse_msg: version: <SIP/2.0>
>
> Aug 28 13:41:02 [8564] DBG:core:parse_headers: flags=2
>
> Aug 28 13:41:02 [8564] DBG:core:parse_via_param: found param type
232,
<branch> =
Shouldn't the transport=TLS ?
Regards
_______________________________________________
Users mailing list
Users(a)lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
2:38 p.m.
Ali Jawad schrieb:
Hi Klaus
You are referring to line 17 right ? That part of traffic is from
openser to the pstn gw ..and both of those are UDP..should it be
transport=tls there or transport=udp?
The Contact is the contact of the caller. Thus, there should be the
IP:socket:protocol which is used by the caller (TLS).
klaus
Thanks
With Regards
Ali Jawad
System Administrator
Splendor Telecom (www.splendor.net)
Beirut, Lebanon
Phone: +961 1 373725
Fax: +961 1 375554
-----Original Message-----
From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at]
Sent: 2008-08-29 14:34
To: Ali Jawad
Cc: users(a)lists.kamailio.org
Subject: Re: [Kamailio-Users] FW: Call Hangup from Caller-End not
working with TLS
1. INVITE:
Contact:
<sip:username@IP.OF.LAN.GW:2949;transport=UDP;rinstance=D98C1DD404B2008F
980980E97E42F8EC;nat=yes>.
As you see the caller announces UDP as contact. Either a bug in the
caller client or do you rewrite the contact in openser?
Further it is strange that the caller sends frmo 127.0.0.1 (Via header,
SDP) but announces a different IP in contact.
regards
klaus
Ali Jawad schrieb:
<sip:michofr@193.227.186.146:3218;transport=UDP;rinstance=D98C1DD404B200
Dear All
Please find below the call setup from my softphone to my cell phone,
The
setup is as follows:
LAN -> Office Gateway <--TLS--->Openser<--UDP-->PSTN GW
I got the trace below by applying
ngrep -W byline -T username -q -d eth0
http://pastebin.com/m2a78a27f
I did the same trace for a udp call and it seemed identical to me, as
you can see in the lower part of the trace that a BYE packet is being
sent to the softphone however the transport is being indicated as UDP
not TLS..is this normal ?
Any clues apart from that ?
Thanks
-----Original Message-----
From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at]
Sent: 2008-08-28 18:13
To: Ali Jawad
Cc: users(a)lists.kamailio.org
Subject: Re: [Kamailio-Users] FW: Call Hangup from Caller-End not
working with TLS
Hi!
Send us an ngrep dump: ngrep -P "" -W byline port 5060
Although this will show us only the UDP part (as TLS is encrypted) but
may still show as the problem.
Which SIP client do you use?
regards
klaus
Ali Jawad schrieb:
mode
Hi All
I am using using openser 1.3..if I make a call between two softphones
on
> the same lan or a a pstn call to my mobile phone..and the
> called/receiver party does hang-up the call. It works fine in UDP > and the call get's hang-up. However in
TLS mode this does not work.
> Anything I might have missed here? Since both udp and tls use the
same
routes,
and voice is fine and no one way audio ..etc.
I notice this http://pastebin.com/m38c979f6 on rtp proxy. However in
the
logs of openser I can't see any errors.
However on the wire shark I can see icmp destination
unreachable...port
> unreachable.
>
> I would have said it is a NAT issue. However it works for simple UDP.
>
>
>
> However I did notice the following in the logs
>
>
>
> Aug 28 13:41:01 [8565] DBG:tm:set_timer: relative timeout is 4000000
>
> Aug 28 13:41:01 [8565] DBG:tm:insert_timer_unsafe: [7]: 0xb6137794
> (45900000)
>
> Aug 28 13:41:01 [8565] DBG:tm:retransmission_handler:
> retransmission_handler : done
>
> Aug 28 13:41:02 [8564] DBG:core:parse_msg: SIP Request:
>
> Aug 28 13:41:02 [8564] DBG:core:parse_msg: method: <BYE>
>
> Aug 28 13:41:02 [8564] DBG:core:parse_msg: uri:
> 8F980980E97E42F8EC;nat=yes>
> Aug 28 13:41:02 [8564] DBG:core:parse_msg: version: <SIP/2.0>
>
> Aug 28 13:41:02 [8564] DBG:core:parse_headers: flags=2
>
> Aug 28 13:41:02 [8564] DBG:core:parse_via_param: found param type
232,
> <branch> =
>
>
>
> Shouldn't the transport=TLS ?
>
>
>
>
>
> Regards
>
>
>
------------------------------------------------------------------------
>
_______________________________________________
> Users mailing list
> Users(a)lists.kamailio.org
> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
5 Sep
5 Sep
3:48 a.m.
Klaus Darilion-2 wrote:
Ali Jawad schrieb:
If you make tls/tcp call from sip client having tls/tcp transport to GW
which has udp transport, you have to add transport=tls before you forward
bye back to client.
--
View this message in context:
http://www.nabble.com/FW%3A-Call-Hangup-from-Caller-End-not-working-with-TL…
Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.
Hi Klaus
You are referring to line 17 right ? That part of traffic is from
openser to the pstn gw ..and both of those are UDP..should it be
transport=tls there or transport=udp?
The Contact is the contact of the caller. Thus, there should be the
IP:socket:protocol which is used by the caller (TLS).
klaus
Thanks
With Regards
Ali Jawad
System Administrator
Splendor Telecom (www.splendor.net)
Beirut, Lebanon
Phone: +961 1 373725
Fax: +961 1 375554
-----Original Message-----
From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at]
Sent: 2008-08-29 14:34
To: Ali Jawad
Cc: users(a)lists.kamailio.org
Subject: Re: [Kamailio-Users] FW: Call Hangup from Caller-End not
working with TLS
1. INVITE:
Contact:
<sip:username@IP.OF.LAN.GW:2949;transport=UDP;rinstance=D98C1DD404B2008F
980980E97E42F8EC;nat=yes>.
As you see the caller announces UDP as contact. Either a bug in the
caller client or do you rewrite the contact in openser?
Further it is strange that the caller sends frmo 127.0.0.1 (Via header,
SDP) but announces a different IP in contact.
regards
klaus
Ali Jawad schrieb:
<sip:michofr@193.227.186.146:3218;transport=UDP;rinstance=D98C1DD404B200
_______________________________________________
Users mailing list
Users(a)lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
Dear All
Please find below the call setup from my softphone to my cell phone,
The
setup is as follows:
LAN -> Office Gateway <--TLS--->Openser<--UDP-->PSTN GW
I got the trace below by applying
ngrep -W byline -T username -q -d eth0
http://pastebin.com/m2a78a27f
I did the same trace for a udp call and it seemed identical to me, as
you can see in the lower part of the trace that a BYE packet is being
sent to the softphone however the transport is being indicated as UDP
not TLS..is this normal ?
Any clues apart from that ?
Thanks
-----Original Message-----
From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at]
Sent: 2008-08-28 18:13
To: Ali Jawad
Cc: users(a)lists.kamailio.org
Subject: Re: [Kamailio-Users] FW: Call Hangup from Caller-End not
working with TLS
Hi!
Send us an ngrep dump: ngrep -P "" -W byline port 5060
Although this will show us only the UDP part (as TLS is encrypted) but
may still show as the problem.
Which SIP client do you use?
regards
klaus
Ali Jawad schrieb:
mode
Hi All
I am using using openser 1.3..if I make a call between two softphones
on
> the same lan or a a pstn call to my mobile phone..and the
> called/receiver party does hang-up the call. It works fine in UDP > and the call get's hang-up. However in
TLS mode this does not work.
> Anything I might have missed here? Since both udp and tls use the
same
routes,
and voice is fine and no one way audio ..etc.
I notice this http://pastebin.com/m38c979f6 on rtp proxy. However in
the
logs of openser I can't see any errors.
However on the wire shark I can see icmp destination
unreachable...port
> unreachable.
>
> I would have said it is a NAT issue. However it works for simple UDP.
>
>
>
> However I did notice the following in the logs
>
>
>
> Aug 28 13:41:01 [8565] DBG:tm:set_timer: relative timeout is 4000000
>
> Aug 28 13:41:01 [8565] DBG:tm:insert_timer_unsafe: [7]: 0xb6137794
> (45900000)
>
> Aug 28 13:41:01 [8565] DBG:tm:retransmission_handler:
> retransmission_handler : done
>
> Aug 28 13:41:02 [8564] DBG:core:parse_msg: SIP Request:
>
> Aug 28 13:41:02 [8564] DBG:core:parse_msg: method: <BYE>
>
> Aug 28 13:41:02 [8564] DBG:core:parse_msg: uri:
> 8F980980E97E42F8EC;nat=yes>
> Aug 28 13:41:02 [8564] DBG:core:parse_msg: version: <SIP/2.0>
>
> Aug 28 13:41:02 [8564] DBG:core:parse_headers: flags=2
>
> Aug 28 13:41:02 [8564] DBG:core:parse_via_param: found param type
232,
> <branch> =
>
>
>
> Shouldn't the transport=TLS ?
>
>
>
>
>
> Regards
>
>
>
------------------------------------------------------------------------
>
_______________________________________________
> Users mailing list
> Users(a)lists.kamailio.org
> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
8:17 a.m.
If you make tls/tcp call from sip client having
tls/tcp transport to GW
which has udp transport, you have to add transport=tls before you forward
bye back to client.
Only if the client is buggy. The client should add this parameter to the
Contact URI.
Nevertheless, using fix_nated_contact and adding the transport=tls
parameter is indeed a workaround
regards
klaus
5711
days inactive
5719
days old
15 comments
4 participants
participants (4)
-
Ali Jawad -
jh＠tutpro.com -
Klaus Darilion -
toly