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
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
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@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
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
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@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
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
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@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
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
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
Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
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@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@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
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@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@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
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
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@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@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
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@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:
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
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@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@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
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@lists.kamailio.org Subject: Re: [Kamailio-Users] FW: Call Hangup from Caller-End not working with TLS
- 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:
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
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@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@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
Klaus Darilion-2 wrote:
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@lists.kamailio.org Subject: Re: [Kamailio-Users] FW: Call Hangup from Caller-End not working with TLS
- 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:
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
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@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
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Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
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.
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
-
Ali Jawad -
jh＠tutpro.com -
Klaus Darilion -
toly