20 Aug
2007
20 Aug
'07
7:50 a.m.
Hi All,
Good day, I'm new to this mailing list :-)
I've setup an OpenSER server with TLS implementation, so far I tested
everything works fine,
but there's strange problem for "BYE" request, from the openser's log it
always complaint:
"Aug 19 22:58:40 ser /sbin/openser[16171]: WARNING:get_send_socket:
protocol/port mismatch"
Here's my implementation:
UA -> OpenSER -> Asterisk (as RTP) -> OpenSER -> UA
On OpenSER I've two port opened which is Udp 5060 and TCP 5061 (TLS).
My UA is Eyebeam v1.5 and the CA certificate (self-signed) already
installed into IE "Trusted RootCA Certificates".
So far I've tested p2p and pstn calls (and call to playback
balance/asterisk), but the problem is unless UA disconnect the call
(initiate BYE),
otherwise if remote party (p2p/pstn/asterisk) initiate BYE request, my
UA will forever stay connected until I disconnect the call manually.
So far I've identified the BYE request from remote party will go into
Loose Route, and I've place a xlog to display the R-Uri..
here's the log from openser: (notes: if UA dial '001' will playback
his/her balance from asterisk)
-------------------------------------------------------------------------------------------------------------
Aug 19 22:58:31 ser /sbin/openser[16189]: DBG: INVITE from client
sip:%user%@%domain% (%UA_IP%) - p2p call
Aug 19 22:58:31 ser /sbin/openser[16189]: DBG: INVITE
F-[sip:%user%@%domain%] T-[sip:001@%domain%] IP-[%UA_IP%] SUCCESSFUL
Aug 19 22:58:32 ser /sbin/openser[16189]: DBG: INVITE from client
sip:%user%@%domain% (%UA_IP%)
Aug 19 22:58:32 ser /sbin/openser[16189]: DBG: INVITE from client
sip:%user%@%domain% (%UA_IP%) - p2p call
Aug 19 22:58:32 ser /sbin/openser[16189]: DBG: INVITE
F-[sip:%user%@%domain%] T-[sip:001@%domain%] IP-[%UA_IP%] SUCCESSFUL
Aug 19 22:58:32 ser /sbin/openser[16189]: DBG:
[sip:%user%@%domain%](%UA_IP%) ACK [sip:001@%domain%] relayed LOOSE ROUTE
Aug 19 22:58:32 ser /sbin/openser[16189]: DBG: ruri
[sip:001@%Asterisk_IP%] relayed LOOSE ROUTE
Aug 19 22:58:40 ser /sbin/openser[16171]: DBG:
[sip:001@%domain%](%Asterisk_IP%) BYE [sip:%user%@%domain%] relayed
LOOSE ROUTE
Aug 19 22:58:40 ser /sbin/openser[16171]: DBG: ruri
[sip:%user%@%domain%:%UA_Port%] relayed LOOSE ROUTE
Aug 19 22:58:40 ser /sbin/openser[16171]: WARNING:get_send_socket:
protocol/port mismatch <<<=============
Aug 19 22:58:41 ser /sbin/openser[16177]: DBG:
[sip:001@%domain%](%Asterisk_IP%) BYE [sip:%user%@%domain%] relayed
LOOSE ROUTE
Aug 19 22:58:41 ser /sbin/openser[16177]: DBG: ruri
[sip:%user%@%UA_IP%:%UA_Port%] relayed LOOSE ROUTE
Aug 19 22:58:42 ser /sbin/openser[16181]: DBG:
[sip:001@%domain%](%Asterisk_IP%) BYE [sip:%user%@%domain%] relayed
LOOSE ROUTE
Aug 19 22:58:42 ser /sbin/openser[16181]: DBG: ruri
[sip:%user%@%UA_IP%:%UA_Port%] relayed LOOSE ROUTE
....... (repeat the last 4 lines N times )
-------------------------------------------------------------------------------------------------------------
Below is my openser.cfg, loose route section
-------------------------------------------------------------------------------------------------------------
###############
# Loose route #
###############
if(loose_route()) {
if(has_totag() && (is_method("INVITE") ||
is_method("ACK"))) {
if(nat_uac_test("19") || search("^Route:.*;nat=yes")) {
fix_nated_contact();
}
}
xlog("L_INFO", "DBG: [$fu]($si) $rm [$tu] relayed LOOSE
ROUTE");
xlog("L_INFO", "DBG: ruri [$ru] relayed LOOSE ROUTE");
t_relay();
exit;
}
-------------------------------------------------------------------------------------------------------------
Had anyone encountered the same problems before ? Do you mind to share
your experience with me ?
Your help are greatly appreciated, thanks in advance.
Regards,
David Loh
20 Aug
20 Aug
7:17 p.m.
Hello,
can you paste the BYE here (you can catch it with ngrep if it is not
coming via tls, e.g., from asterisk)? Also, the invite going to asterisk
will be good.
Cheers,
Daniel
On 08/20/07 07:50, David Loh wrote:
Hi All,
Good day, I'm new to this mailing list :-)
I've setup an OpenSER server with TLS implementation, so far I tested
everything works fine,
but there's strange problem for "BYE" request, from the openser's log
it always complaint:
"Aug 19 22:58:40 ser /sbin/openser[16171]: WARNING:get_send_socket:
protocol/port mismatch"
Here's my implementation:
UA -> OpenSER -> Asterisk (as RTP) -> OpenSER -> UA
On OpenSER I've two port opened which is Udp 5060 and TCP 5061 (TLS).
My UA is Eyebeam v1.5 and the CA certificate (self-signed) already
installed into IE "Trusted RootCA Certificates".
So far I've tested p2p and pstn calls (and call to playback
balance/asterisk), but the problem is unless UA disconnect the call
(initiate BYE),
otherwise if remote party (p2p/pstn/asterisk) initiate BYE request, my
UA will forever stay connected until I disconnect the call manually.
So far I've identified the BYE request from remote party will go into
Loose Route, and I've place a xlog to display the R-Uri..
here's the log from openser: (notes: if UA dial '001' will playback
his/her balance from asterisk)
-------------------------------------------------------------------------------------------------------------
Aug 19 22:58:31 ser /sbin/openser[16189]: DBG: INVITE from client
sip:%user%@%domain% (%UA_IP%) - p2p call
Aug 19 22:58:31 ser /sbin/openser[16189]: DBG: INVITE
F-[sip:%user%@%domain%] T-[sip:001@%domain%] IP-[%UA_IP%] SUCCESSFUL
Aug 19 22:58:32 ser /sbin/openser[16189]: DBG: INVITE from client
sip:%user%@%domain% (%UA_IP%)
Aug 19 22:58:32 ser /sbin/openser[16189]: DBG: INVITE from client
sip:%user%@%domain% (%UA_IP%) - p2p call
Aug 19 22:58:32 ser /sbin/openser[16189]: DBG: INVITE
F-[sip:%user%@%domain%] T-[sip:001@%domain%] IP-[%UA_IP%] SUCCESSFUL
Aug 19 22:58:32 ser /sbin/openser[16189]: DBG:
[sip:%user%@%domain%](%UA_IP%) ACK [sip:001@%domain%] relayed LOOSE ROUTE
Aug 19 22:58:32 ser /sbin/openser[16189]: DBG: ruri
[sip:001@%Asterisk_IP%] relayed LOOSE ROUTE
Aug 19 22:58:40 ser /sbin/openser[16171]: DBG:
[sip:001@%domain%](%Asterisk_IP%) BYE [sip:%user%@%domain%] relayed
LOOSE ROUTE
Aug 19 22:58:40 ser /sbin/openser[16171]: DBG: ruri
[sip:%user%@%domain%:%UA_Port%] relayed LOOSE ROUTE
Aug 19 22:58:40 ser /sbin/openser[16171]: WARNING:get_send_socket:
protocol/port mismatch <<<=============
Aug 19 22:58:41 ser /sbin/openser[16177]: DBG:
[sip:001@%domain%](%Asterisk_IP%) BYE [sip:%user%@%domain%] relayed
LOOSE ROUTE
Aug 19 22:58:41 ser /sbin/openser[16177]: DBG: ruri
[sip:%user%@%UA_IP%:%UA_Port%] relayed LOOSE ROUTE
Aug 19 22:58:42 ser /sbin/openser[16181]: DBG:
[sip:001@%domain%](%Asterisk_IP%) BYE [sip:%user%@%domain%] relayed
LOOSE ROUTE
Aug 19 22:58:42 ser /sbin/openser[16181]: DBG: ruri
[sip:%user%@%UA_IP%:%UA_Port%] relayed LOOSE ROUTE
....... (repeat the last 4 lines N times )
-------------------------------------------------------------------------------------------------------------
Below is my openser.cfg, loose route section
-------------------------------------------------------------------------------------------------------------
###############
# Loose route #
###############
if(loose_route()) {
if(has_totag() && (is_method("INVITE") ||
is_method("ACK"))) {
if(nat_uac_test("19") || search("^Route:.*;nat=yes")) {
fix_nated_contact();
}
}
xlog("L_INFO", "DBG: [$fu]($si) $rm [$tu] relayed LOOSE
ROUTE");
xlog("L_INFO", "DBG: ruri [$ru] relayed LOOSE ROUTE");
t_relay();
exit;
}
-------------------------------------------------------------------------------------------------------------
Had anyone encountered the same problems before ? Do you mind to share
your experience with me ?
Your help are greatly appreciated, thanks in advance.
Regards,
David Loh
_______________________________________________
Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
21 Aug
21 Aug
2:28 p.m.
Hi Daniel,
Thanks for the reply, here's my INVITE and BYE packet, I tcpdump-ed and
and use wireshark to export as txt, so the format might be abit off.
1) Invite from OpenSER to Asterisk via port 5060, asterisk return a 200.
And the call successfully established
------------------------------------------------------------------------
Record-Route: <sip:%OpenSER_IP%;r2=on;lr;ftag=1247871f;nat=yes>
Record-Route:
<sip:%OpenSER_IP%:5061;transport=tls;r2=on;lr;ftag=1247871f;nat=yes>
Via: SIP/2.0/UDP %OpenSER_IP%;branch=z9hG4bKe42e.ff11ae32.0;i=52
Via: SIP/2.0/TLS
%UAC_LAN_IP%:23323;received=%UAC_WAN_IP%;branch=z9hG4bK-d87543-2820ee009262b500-1--d87543-;rport=3135
Max-Forwards: 69
Contact: <sip:davidloh@%UAC_WAN_IP%:3135;transport=TLS>
To: "001"<sip:001@%SIP_Domain%>
From: "ser"<sip:davidloh@%SIP_Domain%>;tag=1247871f
Call-ID: Zjc4OWJhZGNiMDUxNjQ5Y2I2ZDU3NTc4NGY2Y2RmOGQ.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1004p stamp 31962
Content-Length: 239
------------------------------------------------------------------------
2) Here's the BYE, from Asterisk to OpenSER.
------------------------------------------------------------------------
Via: SIP/2.0/UDP %Asterisk_IP%:5060;branch=z9hG4bK0ca1ab09;rport
Route:
<sip:%OpenSER_IP%;r2=on;lr;ftag=1247871f;nat=yes>,<sip:%OpenSER_IP%:5061;transport=tls;r2=on;lr;ftag=1247871f;nat=yes>
From: "001"<sip:001@%SIP_Domain%>;tag=as0492a9d0
To: "ser"<sip:davidloh@%SIP_Domain%>;tag=1247871f
Call-ID: Zjc4OWJhZGNiMDUxNjQ5Y2I2ZDU3NTc4NGY2Y2RmOGQ.
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
------------------------------------------------------------------------
3) And then, OpenSER passing the BYE to UAC...
------------------------------------------------------------------------
Record-Route: <sip:%OpenSER_IP%;lr;ftag=as0492a9d0>
Via: SIP/2.0/UDP %OpenSER_IP%;branch=z9hG4bK5e.1195fd8.0
Via: SIP/2.0/UDP %Asterisk_IP%:5060;branch=z9hG4bK0ca1ab09;rport=5060
From: "001"<sip:001@%SIP_Domain%>;tag=as0492a9d0
To: "ser"<sip:davidloh@%SIP_Domain%>;tag=1247871f
Call-ID: Zjc4OWJhZGNiMDUxNjQ5Y2I2ZDU3NTc4NGY2Y2RmOGQ.
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 69
Content-Length: 0
------------------------------------------------------------------------
I think the "protocol/port mismatch" happened right after (3) because
from my ethereal trace,
after short while (perhaps timeout), Asterisk will re-initiate a "BYE"
and pass to OpenSER (packet #2), and OpenSER will forward it to UAC
(packet #3),
but somehow UAC doesn't response an "ACK" to OpenSER that why from my
ethereal trace, packet #2 and #3 kept looping.
Any thoughts ?
Thanks in advance.
Regards,
David Loh
Daniel-Constantin Mierla wrote:
Hello,
can you paste the BYE here (you can catch it with ngrep if it is not
coming via tls, e.g., from asterisk)? Also, the invite going to
asterisk will be good.
Cheers,
Daniel
On 08/20/07 07:50, David Loh wrote:
Hi All,
Good day, I'm new to this mailing list :-)
I've setup an OpenSER server with TLS implementation, so far I tested
everything works fine,
but there's strange problem for "BYE" request, from the openser's log
it always complaint:
"Aug 19 22:58:40 ser /sbin/openser[16171]: WARNING:get_send_socket:
protocol/port mismatch"
Here's my implementation:
UA -> OpenSER -> Asterisk (as RTP) -> OpenSER -> UA
On OpenSER I've two port opened which is Udp 5060 and TCP 5061 (TLS).
My UA is Eyebeam v1.5 and the CA certificate (self-signed) already
installed into IE "Trusted RootCA Certificates".
So far I've tested p2p and pstn calls (and call to playback
balance/asterisk), but the problem is unless UA disconnect the call
(initiate BYE),
otherwise if remote party (p2p/pstn/asterisk) initiate BYE request,
my UA will forever stay connected until I disconnect the call manually.
So far I've identified the BYE request from remote party will go into
Loose Route, and I've place a xlog to display the R-Uri..
here's the log from openser: (notes: if UA dial '001' will playback
his/her balance from asterisk)
-------------------------------------------------------------------------------------------------------------
Aug 19 22:58:31 ser /sbin/openser[16189]: DBG: INVITE from client
sip:%user%@%domain% (%UA_IP%) - p2p call
Aug 19 22:58:31 ser /sbin/openser[16189]: DBG: INVITE
F-[sip:%user%@%domain%] T-[sip:001@%domain%] IP-[%UA_IP%] SUCCESSFUL
Aug 19 22:58:32 ser /sbin/openser[16189]: DBG: INVITE from client
sip:%user%@%domain% (%UA_IP%)
Aug 19 22:58:32 ser /sbin/openser[16189]: DBG: INVITE from client
sip:%user%@%domain% (%UA_IP%) - p2p call
Aug 19 22:58:32 ser /sbin/openser[16189]: DBG: INVITE
F-[sip:%user%@%domain%] T-[sip:001@%domain%] IP-[%UA_IP%] SUCCESSFUL
Aug 19 22:58:32 ser /sbin/openser[16189]: DBG:
[sip:%user%@%domain%](%UA_IP%) ACK [sip:001@%domain%] relayed LOOSE
ROUTE
Aug 19 22:58:32 ser /sbin/openser[16189]: DBG: ruri
[sip:001@%Asterisk_IP%] relayed LOOSE ROUTE
Aug 19 22:58:40 ser /sbin/openser[16171]: DBG:
[sip:001@%domain%](%Asterisk_IP%) BYE [sip:%user%@%domain%] relayed
LOOSE ROUTE
Aug 19 22:58:40 ser /sbin/openser[16171]: DBG: ruri
[sip:%user%@%domain%:%UA_Port%] relayed LOOSE ROUTE
Aug 19 22:58:40 ser /sbin/openser[16171]: WARNING:get_send_socket:
protocol/port mismatch <<<=============
Aug 19 22:58:41 ser /sbin/openser[16177]: DBG:
[sip:001@%domain%](%Asterisk_IP%) BYE [sip:%user%@%domain%] relayed
LOOSE ROUTE
Aug 19 22:58:41 ser /sbin/openser[16177]: DBG: ruri
[sip:%user%@%UA_IP%:%UA_Port%] relayed LOOSE ROUTE
Aug 19 22:58:42 ser /sbin/openser[16181]: DBG:
[sip:001@%domain%](%Asterisk_IP%) BYE [sip:%user%@%domain%] relayed
LOOSE ROUTE
Aug 19 22:58:42 ser /sbin/openser[16181]: DBG: ruri
[sip:%user%@%UA_IP%:%UA_Port%] relayed LOOSE ROUTE
....... (repeat the last 4 lines N times )
-------------------------------------------------------------------------------------------------------------
Below is my openser.cfg, loose route section
-------------------------------------------------------------------------------------------------------------
###############
# Loose route #
###############
if(loose_route()) {
if(has_totag() && (is_method("INVITE") ||
is_method("ACK"))) {
if(nat_uac_test("19") || search("^Route:.*;nat=yes")) {
fix_nated_contact();
}
}
xlog("L_INFO", "DBG: [$fu]($si) $rm [$tu] relayed LOOSE
ROUTE");
xlog("L_INFO", "DBG: ruri [$ru] relayed LOOSE ROUTE");
t_relay();
exit;
}
-------------------------------------------------------------------------------------------------------------
Had anyone encountered the same problems before ? Do you mind to
share your experience with me ?
Your help are greatly appreciated, thanks in advance.
Regards,
David Loh
_______________________________________________
Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
3:21 p.m.
Hello,
please send the request uri as well and the details about the IP
communication (source/destination address, port).
use:
ngrep -qt -W byline port 5060
the replace the IPs with something suggestive to protect your privacy.
Daniel
On 08/21/07 14:28, David Loh wrote:
Hi Daniel,
Thanks for the reply, here's my INVITE and BYE packet, I tcpdump-ed
and and use wireshark to export as txt, so the format might be abit off.
1) Invite from OpenSER to Asterisk via port 5060, asterisk return a
200. And the call successfully established
------------------------------------------------------------------------
Record-Route: <sip:%OpenSER_IP%;r2=on;lr;ftag=1247871f;nat=yes>
Record-Route:
<sip:%OpenSER_IP%:5061;transport=tls;r2=on;lr;ftag=1247871f;nat=yes>
Via: SIP/2.0/UDP %OpenSER_IP%;branch=z9hG4bKe42e.ff11ae32.0;i=52
Via: SIP/2.0/TLS
%UAC_LAN_IP%:23323;received=%UAC_WAN_IP%;branch=z9hG4bK-d87543-2820ee009262b500-1--d87543-;rport=3135
Max-Forwards: 69
Contact: <sip:davidloh@%UAC_WAN_IP%:3135;transport=TLS>
To: "001"<sip:001@%SIP_Domain%>
From: "ser"<sip:davidloh@%SIP_Domain%>;tag=1247871f
Call-ID: Zjc4OWJhZGNiMDUxNjQ5Y2I2ZDU3NTc4NGY2Y2RmOGQ.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1004p stamp 31962
Content-Length: 239
------------------------------------------------------------------------
2) Here's the BYE, from Asterisk to OpenSER.
------------------------------------------------------------------------
Via: SIP/2.0/UDP %Asterisk_IP%:5060;branch=z9hG4bK0ca1ab09;rport
Route:
<sip:%OpenSER_IP%;r2=on;lr;ftag=1247871f;nat=yes>,<sip:%OpenSER_IP%:5061;transport=tls;r2=on;lr;ftag=1247871f;nat=yes>
From: "001"<sip:001@%SIP_Domain%>;tag=as0492a9d0
To: "ser"<sip:davidloh@%SIP_Domain%>;tag=1247871f
Call-ID: Zjc4OWJhZGNiMDUxNjQ5Y2I2ZDU3NTc4NGY2Y2RmOGQ.
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
------------------------------------------------------------------------
3) And then, OpenSER passing the BYE to UAC...
------------------------------------------------------------------------
Record-Route: <sip:%OpenSER_IP%;lr;ftag=as0492a9d0>
Via: SIP/2.0/UDP %OpenSER_IP%;branch=z9hG4bK5e.1195fd8.0
Via: SIP/2.0/UDP %Asterisk_IP%:5060;branch=z9hG4bK0ca1ab09;rport=5060
From: "001"<sip:001@%SIP_Domain%>;tag=as0492a9d0
To: "ser"<sip:davidloh@%SIP_Domain%>;tag=1247871f
Call-ID: Zjc4OWJhZGNiMDUxNjQ5Y2I2ZDU3NTc4NGY2Y2RmOGQ.
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 69
Content-Length: 0
------------------------------------------------------------------------
I think the "protocol/port mismatch" happened right after (3) because
from my ethereal trace,
after short while (perhaps timeout), Asterisk will re-initiate a "BYE"
and pass to OpenSER (packet #2), and OpenSER will forward it to UAC
(packet #3),
but somehow UAC doesn't response an "ACK" to OpenSER that why from my
ethereal trace, packet #2 and #3 kept looping.
Any thoughts ?
Thanks in advance.
Regards,
David Loh
Daniel-Constantin Mierla wrote:
Hello,
can you paste the BYE here (you can catch it with ngrep if it is not
coming via tls, e.g., from asterisk)? Also, the invite going to
asterisk will be good.
Cheers,
Daniel
On 08/20/07 07:50, David Loh wrote:
Hi All,
Good day, I'm new to this mailing list :-)
I've setup an OpenSER server with TLS implementation, so far I
tested everything works fine,
but there's strange problem for "BYE" request, from the openser's
log it always complaint:
"Aug 19 22:58:40 ser /sbin/openser[16171]: WARNING:get_send_socket:
protocol/port mismatch"
Here's my implementation:
UA -> OpenSER -> Asterisk (as RTP) -> OpenSER -> UA
On OpenSER I've two port opened which is Udp 5060 and TCP 5061 (TLS).
My UA is Eyebeam v1.5 and the CA certificate (self-signed) already
installed into IE "Trusted RootCA Certificates".
So far I've tested p2p and pstn calls (and call to playback
balance/asterisk), but the problem is unless UA disconnect the call
(initiate BYE),
otherwise if remote party (p2p/pstn/asterisk) initiate BYE request,
my UA will forever stay connected until I disconnect the call manually.
So far I've identified the BYE request from remote party will go
into Loose Route, and I've place a xlog to display the R-Uri..
here's the log from openser: (notes: if UA dial '001' will playback
his/her balance from asterisk)
-------------------------------------------------------------------------------------------------------------
Aug 19 22:58:31 ser /sbin/openser[16189]: DBG: INVITE from client
sip:%user%@%domain% (%UA_IP%) - p2p call
Aug 19 22:58:31 ser /sbin/openser[16189]: DBG: INVITE
F-[sip:%user%@%domain%] T-[sip:001@%domain%] IP-[%UA_IP%] SUCCESSFUL
Aug 19 22:58:32 ser /sbin/openser[16189]: DBG: INVITE from client
sip:%user%@%domain% (%UA_IP%)
Aug 19 22:58:32 ser /sbin/openser[16189]: DBG: INVITE from client
sip:%user%@%domain% (%UA_IP%) - p2p call
Aug 19 22:58:32 ser /sbin/openser[16189]: DBG: INVITE
F-[sip:%user%@%domain%] T-[sip:001@%domain%] IP-[%UA_IP%] SUCCESSFUL
Aug 19 22:58:32 ser /sbin/openser[16189]: DBG:
[sip:%user%@%domain%](%UA_IP%) ACK [sip:001@%domain%] relayed LOOSE
ROUTE
Aug 19 22:58:32 ser /sbin/openser[16189]: DBG: ruri
[sip:001@%Asterisk_IP%] relayed LOOSE ROUTE
Aug 19 22:58:40 ser /sbin/openser[16171]: DBG:
[sip:001@%domain%](%Asterisk_IP%) BYE [sip:%user%@%domain%] relayed
LOOSE ROUTE
Aug 19 22:58:40 ser /sbin/openser[16171]: DBG: ruri
[sip:%user%@%domain%:%UA_Port%] relayed LOOSE ROUTE
Aug 19 22:58:40 ser /sbin/openser[16171]: WARNING:get_send_socket:
protocol/port mismatch <<<=============
Aug 19 22:58:41 ser /sbin/openser[16177]: DBG:
[sip:001@%domain%](%Asterisk_IP%) BYE [sip:%user%@%domain%] relayed
LOOSE ROUTE
Aug 19 22:58:41 ser /sbin/openser[16177]: DBG: ruri
[sip:%user%@%UA_IP%:%UA_Port%] relayed LOOSE ROUTE
Aug 19 22:58:42 ser /sbin/openser[16181]: DBG:
[sip:001@%domain%](%Asterisk_IP%) BYE [sip:%user%@%domain%] relayed
LOOSE ROUTE
Aug 19 22:58:42 ser /sbin/openser[16181]: DBG: ruri
[sip:%user%@%UA_IP%:%UA_Port%] relayed LOOSE ROUTE
....... (repeat the last 4 lines N times )
-------------------------------------------------------------------------------------------------------------
Below is my openser.cfg, loose route section
-------------------------------------------------------------------------------------------------------------
###############
# Loose route #
###############
if(loose_route()) {
if(has_totag() && (is_method("INVITE") ||
is_method("ACK"))) {
if(nat_uac_test("19") || search("^Route:.*;nat=yes")) {
fix_nated_contact();
}
}
xlog("L_INFO", "DBG: [$fu]($si) $rm [$tu] relayed LOOSE
ROUTE");
xlog("L_INFO", "DBG: ruri [$ru] relayed LOOSE ROUTE");
t_relay();
exit;
}
-------------------------------------------------------------------------------------------------------------
Had anyone encountered the same problems before ? Do you mind to
share your experience with me ?
Your help are greatly appreciated, thanks in advance.
Regards,
David Loh
_______________________________________________
Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
22 Aug
22 Aug
6:01 a.m.
Hi Daniel,
Here's my ngrep result:
# ngrep -qt -W byline port 5060
interface: eth0 (FF.FFF.FFF.0/255.255.255.192)
filter: (ip or ip6) and ( port 5060 )
------------------------------------------------------------------------
U 2007/08/21 21:49:09.562599 %OpenSER_IP%:5060 -> %Asterisk_IP%:5060
INVITE sip:001@%Asterisk_IP%:5060 SIP/2.0.
Record-Route: <sip:%OpenSER_IP%;r2=on;lr;ftag=e803a341;nat=yes>.
Record-Route:
<sip:%OpenSER_IP%:5061;transport=tls;r2=on;lr;ftag=e803a341;nat=yes>.
Via: SIP/2.0/UDP %OpenSER_IP%;branch=z9hG4bK51ee.4fb7f0a3.0;i=7.
Via: SIP/2.0/TLS
%UAC_LAN_IP%:26261;received=%UAC_WAN_IP%;branch=z9hG4bK-d87543-9d18e502fd1ebb4a-1--d87543-;rport=2150.
Max-Forwards: 69.
Contact: <sip:davidloh@%UAC_WAN_IP%:2150;transport=TLS>.
To: "001"<sip:001@%SIP_Domain%>.
From: "ser"<sip:davidloh@%SIP_Domain%>;tag=e803a341.
Call-ID: YjRkZWY0YTZjZjdhOTZmYjE5MTkxZGJlMmFjOGY4YzM..
CSeq: 1 INVITE.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO.
Content-Type: application/sdp.
User-Agent: eyeBeam release 1004p stamp 31962.
Content-Length: 410.
.
v=0.
o=- 8 2 IN IP4 %UAC_LAN_IP%.
s=CounterPath eyeBeam 1.5.
c=IN IP4 %UAC_LAN_IP%.
t=0 0.
m=audio 27394 RTP/SAVP 0 101.
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:SRe4Du/u5MKbHQlbtjHu/zDP9owTDYBjSf/ky7YL .
a=crypto:2 AES_CM_128_HMAC_SHA1_32
inline:vWS2/LxDhDj8pHF6T+FP6aDtjxYBxloEhpp2pmto .
a=fmtp:101 0-15.
a=rtpmap:101 telephone-event/8000.
a=sendrecv.
a=x-rtp-session-id:D2D7D33C12AA4F918DCBDF8601EE6186.
------------------------------------------------------------------------
U 2007/08/21 21:49:09.590902 %Asterisk_IP%:5060 -> %OpenSER_IP%:5060
SIP/2.0 488 Not acceptable here.
Via: SIP/2.0/UDP
%OpenSER_IP%;branch=z9hG4bK51ee.4fb7f0a3.0;i=7;received=%OpenSER_IP%.
Via: SIP/2.0/TLS
%UAC_LAN_IP%:26261;received=%UAC_WAN_IP%;branch=z9hG4bK-d87543-9d18e502fd1ebb4a-1--d87543-;rport=2150.
From: "ser"<sip:davidloh@%SIP_Domain%>;tag=e803a341.
To: "001"<sip:001@%SIP_Domain%>;tag=as43cc1284.
Call-ID: YjRkZWY0YTZjZjdhOTZmYjE5MTkxZGJlMmFjOGY4YzM..
CSeq: 1 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Content-Length: 0.
.
------------------------------------------------------------------------
U 2007/08/21 21:49:09.591253 %OpenSER_IP%:5060 -> %Asterisk_IP%:5060
ACK sip:001@%Asterisk_IP%:5060 SIP/2.0.
Via: SIP/2.0/UDP %OpenSER_IP%;branch=z9hG4bK51ee.4fb7f0a3.0.
From: "ser"<sip:davidloh@%SIP_Domain%>;tag=e803a341.
Call-ID: YjRkZWY0YTZjZjdhOTZmYjE5MTkxZGJlMmFjOGY4YzM..
To: "001"<sip:001@%SIP_Domain%>;tag=as43cc1284.
CSeq: 1 ACK.
Content-Length: 0.
.
------------------------------------------------------------------------
U 2007/08/21 21:49:10.297407 %OpenSER_IP%:5060 -> %Asterisk_IP%:5060
INVITE sip:001@%Asterisk_IP%:5060 SIP/2.0.
Record-Route: <sip:%OpenSER_IP%;r2=on;lr;ftag=3c021055;nat=yes>.
Record-Route:
<sip:%OpenSER_IP%:5061;transport=tls;r2=on;lr;ftag=3c021055;nat=yes>.
Via: SIP/2.0/UDP %OpenSER_IP%;branch=z9hG4bK5916.d8cf7e33.0;i=7.
Via: SIP/2.0/TLS
%UAC_LAN_IP%:26261;received=%UAC_WAN_IP%;branch=z9hG4bK-d87543-42532f6937642019-1--d87543-;rport=2150.
Max-Forwards: 69.
Contact: <sip:davidloh@%UAC_WAN_IP%:2150;transport=TLS>.
To: "001"<sip:001@%SIP_Domain%>.
From: "ser"<sip:davidloh@%SIP_Domain%>;tag=3c021055.
Call-ID: NTc0YmQxMDVlMzU4Y2U3OTE2ZGU1ZDRjMzE0ZGRhYzU..
CSeq: 1 INVITE.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO.
Content-Type: application/sdp.
User-Agent: eyeBeam release 1004p stamp 31962.
Content-Length: 239.
.
v=0.
o=- 5 2 IN IP4 %UAC_LAN_IP%.
s=CounterPath eyeBeam 1.5.
c=IN IP4 %UAC_LAN_IP%.
t=0 0.
m=audio 12508 RTP/AVP 0 101.
a=fmtp:101 0-15.
a=rtpmap:101 telephone-event/8000.
a=sendrecv.
a=x-rtp-session-id:58FD83624F184C9B912F202E04FFAE49.
------------------------------------------------------------------------
U 2007/08/21 21:49:10.325559 %Asterisk_IP%:5060 -> %OpenSER_IP%:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP
%OpenSER_IP%;branch=z9hG4bK5916.d8cf7e33.0;i=7;received=%OpenSER_IP%.
Via: SIP/2.0/TLS
%UAC_LAN_IP%:26261;received=%UAC_WAN_IP%;branch=z9hG4bK-d87543-42532f6937642019-1--d87543-;rport=2150.
From: "ser"<sip:davidloh@%SIP_Domain%>;tag=3c021055.
To: "001"<sip:001@%SIP_Domain%>.
Call-ID: NTc0YmQxMDVlMzU4Y2U3OTE2ZGU1ZDRjMzE0ZGRhYzU..
CSeq: 1 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Contact: <sip:001@%Asterisk_IP%>.
Content-Length: 0.
.
------------------------------------------------------------------------
U 2007/08/21 21:49:10.388885 %Asterisk_IP%:5060 -> %OpenSER_IP%:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP
%OpenSER_IP%;branch=z9hG4bK5916.d8cf7e33.0;i=7;received=%OpenSER_IP%.
Via: SIP/2.0/TLS
%UAC_LAN_IP%:26261;received=%UAC_WAN_IP%;branch=z9hG4bK-d87543-42532f6937642019-1--d87543-;rport=2150.
Record-Route: <sip:%OpenSER_IP%;r2=on;lr;ftag=3c021055;nat=yes>.
Record-Route:
<sip:%OpenSER_IP%:5061;transport=tls;r2=on;lr;ftag=3c021055;nat=yes>.
From: "ser"<sip:davidloh@%SIP_Domain%>;tag=3c021055.
To: "001"<sip:001@%SIP_Domain%>;tag=as5b2df0bb.
Call-ID: NTc0YmQxMDVlMzU4Y2U3OTE2ZGU1ZDRjMzE0ZGRhYzU..
CSeq: 1 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Contact: <sip:001@%Asterisk_IP%>.
Content-Type: application/sdp.
Content-Length: 218.
.
v=0.
o=root 17708 17708 IN IP4 %Asterisk_IP%.
s=session.
c=IN IP4 %Asterisk_IP%.
t=0 0.
m=audio 19350 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
------------------------------------------------------------------------
U 2007/08/21 21:49:11.009760 %OpenSER_IP%:5060 -> %Asterisk_IP%:5060
ACK sip:001@%Asterisk_IP% SIP/2.0.
Record-Route: <sip:%OpenSER_IP%;r2=on;lr;ftag=3c021055;nat=yes>.
Record-Route:
<sip:%OpenSER_IP%:5061;transport=tls;r2=on;lr;ftag=3c021055;nat=yes>.
Via: SIP/2.0/UDP %OpenSER_IP%;branch=z9hG4bK5916.d8cf7e33.2;i=7.
Via: SIP/2.0/TLS
%UAC_LAN_IP%:26261;received=%UAC_WAN_IP%;branch=z9hG4bK-d87543-d1635f327746f900-1--d87543-;rport=2150.
Max-Forwards: 69.
Contact: <sip:davidloh@%UAC_WAN_IP%:2150;transport=TLS>.
To: "001"<sip:001@%SIP_Domain%>;tag=as5b2df0bb.
From: "ser"<sip:davidloh@%SIP_Domain%>;tag=3c021055.
Call-ID: NTc0YmQxMDVlMzU4Y2U3OTE2ZGU1ZDRjMzE0ZGRhYzU..
CSeq: 1 ACK.
User-Agent: eyeBeam release 1004p stamp 31962.
Content-Length: 0.
.
------------------------------------------------------------------------
U 2007/08/21 21:49:18.671026 %Asterisk_IP%:5060 -> %OpenSER_IP%:5060
BYE sip:davidloh@%UAC_WAN_IP%:2150 SIP/2.0.
Via: SIP/2.0/UDP %Asterisk_IP%:5060;branch=z9hG4bK127246b8;rport.
Route:
<sip:%OpenSER_IP%;r2=on;lr;ftag=3c021055;nat=yes>,<sip:%OpenSER_IP%:5061;transport=tls;r2=on;lr;ftag=3c021055;nat=yes>.
From: "001"<sip:001@%SIP_Domain%>;tag=as5b2df0bb.
To: "ser"<sip:davidloh@%SIP_Domain%>;tag=3c021055.
Call-ID: NTc0YmQxMDVlMzU4Y2U3OTE2ZGU1ZDRjMzE0ZGRhYzU..
CSeq: 102 BYE.
User-Agent: Asterisk PBX.
Max-Forwards: 70.
Content-Length: 0.
.
------------------------------------------------------------------------
U 2007/08/21 21:49:18.671397 %OpenSER_IP%:5060 -> %UAC_WAN_IP%:2150
BYE sip:davidloh@%UAC_WAN_IP%:2150 SIP/2.0.
Record-Route: <sip:%OpenSER_IP%;lr;ftag=as5b2df0bb>.
Via: SIP/2.0/UDP %OpenSER_IP%;branch=z9hG4bKe208.73d59fa4.0.
Via: SIP/2.0/UDP %Asterisk_IP%:5060;branch=z9hG4bK127246b8;rport=5060.
From: "001"<sip:001@%SIP_Domain%>;tag=as5b2df0bb.
To: "ser"<sip:davidloh@%SIP_Domain%>;tag=3c021055.
Call-ID: NTc0YmQxMDVlMzU4Y2U3OTE2ZGU1ZDRjMzE0ZGRhYzU..
CSeq: 102 BYE.
User-Agent: Asterisk PBX.
Max-Forwards: 69.
Content-Length: 0.
.
------------------------------------------------------------------------
U 2007/08/21 21:49:19.097425 %OpenSER_IP%:5060 -> %UAC_WAN_IP%:2150
BYE sip:davidloh@%UAC_WAN_IP%:2150 SIP/2.0.
Record-Route: <sip:%OpenSER_IP%;lr;ftag=as5b2df0bb>.
Via: SIP/2.0/UDP %OpenSER_IP%;branch=z9hG4bKe208.73d59fa4.0.
Via: SIP/2.0/UDP %Asterisk_IP%:5060;branch=z9hG4bK127246b8;rport=5060.
From: "001"<sip:001@%SIP_Domain%>;tag=as5b2df0bb.
To: "ser"<sip:davidloh@%SIP_Domain%>;tag=3c021055.
Call-ID: NTc0YmQxMDVlMzU4Y2U3OTE2ZGU1ZDRjMzE0ZGRhYzU..
CSeq: 102 BYE.
User-Agent: Asterisk PBX.
Max-Forwards: 69.
Content-Length: 0.
.
------------------------------------------------------------------------
U 2007/08/21 21:49:19.671474 %Asterisk_IP%:5060 -> %OpenSER_IP%:5060
BYE sip:davidloh@%UAC_WAN_IP%:2150 SIP/2.0.
Via: SIP/2.0/UDP %Asterisk_IP%:5060;branch=z9hG4bK127246b8;rport.
Route:
<sip:%OpenSER_IP%;r2=on;lr;ftag=3c021055;nat=yes>,<sip:%OpenSER_IP%:5061;transport=tls;r2=on;lr;ftag=3c021055;nat=yes>.
From: "001"<sip:001@%SIP_Domain%>;tag=as5b2df0bb.
To: "ser"<sip:davidloh@%SIP_Domain%>;tag=3c021055.
Call-ID: NTc0YmQxMDVlMzU4Y2U3OTE2ZGU1ZDRjMzE0ZGRhYzU..
CSeq: 102 BYE.
User-Agent: Asterisk PBX.
Max-Forwards: 70.
Content-Length: 0.
.
------------------------------------------------------------------------
U 2007/08/21 21:49:20.097952 %OpenSER_IP%:5060 -> %UAC_WAN_IP%:2150
BYE sip:davidloh@%UAC_WAN_IP%:2150 SIP/2.0.
Record-Route: <sip:%OpenSER_IP%;lr;ftag=as5b2df0bb>.
Via: SIP/2.0/UDP %OpenSER_IP%;branch=z9hG4bKe208.73d59fa4.0.
Via: SIP/2.0/UDP %Asterisk_IP%:5060;branch=z9hG4bK127246b8;rport=5060.
From: "001"<sip:001@%SIP_Domain%>;tag=as5b2df0bb.
To: "ser"<sip:davidloh@%SIP_Domain%>;tag=3c021055.
Call-ID: NTc0YmQxMDVlMzU4Y2U3OTE2ZGU1ZDRjMzE0ZGRhYzU..
CSeq: 102 BYE.
User-Agent: Asterisk PBX.
Max-Forwards: 69.
Content-Length: 0.
.
------------------------------------------------------------------------
U 2007/08/21 21:49:20.670483 %Asterisk_IP%:5060 -> %OpenSER_IP%:5060
BYE sip:davidloh@%UAC_WAN_IP%:2150 SIP/2.0.
Via: SIP/2.0/UDP %Asterisk_IP%:5060;branch=z9hG4bK127246b8;rport.
Route:
<sip:%OpenSER_IP%;r2=on;lr;ftag=3c021055;nat=yes>,<sip:%OpenSER_IP%:5061;transport=tls;r2=on;lr;ftag=3c021055;nat=yes>.
From: "001"<sip:001@%SIP_Domain%>;tag=as5b2df0bb.
To: "ser"<sip:davidloh@%SIP_Domain%>;tag=3c021055.
Call-ID: NTc0YmQxMDVlMzU4Y2U3OTE2ZGU1ZDRjMzE0ZGRhYzU..
CSeq: 102 BYE.
User-Agent: Asterisk PBX.
Max-Forwards: 70.
Content-Length: 0.
.
------------------------------------------------------------------------
U 2007/08/21 21:49:22.099186 %OpenSER_IP%:5060 -> %UAC_WAN_IP%:2150
BYE sip:davidloh@%UAC_WAN_IP%:2150 SIP/2.0.
Record-Route: <sip:%OpenSER_IP%;lr;ftag=as5b2df0bb>.
Via: SIP/2.0/UDP %OpenSER_IP%;branch=z9hG4bKe208.73d59fa4.0.
Via: SIP/2.0/UDP %Asterisk_IP%:5060;branch=z9hG4bK127246b8;rport=5060.
From: "001"<sip:001@%SIP_Domain%>;tag=as5b2df0bb.
To: "ser"<sip:davidloh@%SIP_Domain%>;tag=3c021055.
Call-ID: NTc0YmQxMDVlMzU4Y2U3OTE2ZGU1ZDRjMzE0ZGRhYzU..
CSeq: 102 BYE.
User-Agent: Asterisk PBX.
Max-Forwards: 69.
Content-Length: 0.
.
------------------------------------------------------------------------
U 2007/08/21 21:49:22.671417 %Asterisk_IP%:5060 -> %OpenSER_IP%:5060
BYE sip:davidloh@%UAC_WAN_IP%:2150 SIP/2.0.
Via: SIP/2.0/UDP %Asterisk_IP%:5060;branch=z9hG4bK127246b8;rport.
Route:
<sip:%OpenSER_IP%;r2=on;lr;ftag=3c021055;nat=yes>,<sip:%OpenSER_IP%:5061;transport=tls;r2=on;lr;ftag=3c021055;nat=yes>.
From: "001"<sip:001@%SIP_Domain%>;tag=as5b2df0bb.
To: "ser"<sip:davidloh@%SIP_Domain%>;tag=3c021055.
Call-ID: NTc0YmQxMDVlMzU4Y2U3OTE2ZGU1ZDRjMzE0ZGRhYzU..
CSeq: 102 BYE.
User-Agent: Asterisk PBX.
Max-Forwards: 70.
Content-Length: 0.
.
------------------------------------------------------------------------
U 2007/08/21 21:49:26.101373 %OpenSER_IP%:5060 -> %UAC_WAN_IP%:2150
BYE sip:davidloh@%UAC_WAN_IP%:2150 SIP/2.0.
Record-Route: <sip:%OpenSER_IP%;lr;ftag=as5b2df0bb>.
Via: SIP/2.0/UDP %OpenSER_IP%;branch=z9hG4bKe208.73d59fa4.0.
Via: SIP/2.0/UDP %Asterisk_IP%:5060;branch=z9hG4bK127246b8;rport=5060.
From: "001"<sip:001@%SIP_Domain%>;tag=as5b2df0bb.
To: "ser"<sip:davidloh@%SIP_Domain%>;tag=3c021055.
Call-ID: NTc0YmQxMDVlMzU4Y2U3OTE2ZGU1ZDRjMzE0ZGRhYzU..
CSeq: 102 BYE.
User-Agent: Asterisk PBX.
Max-Forwards: 69.
Content-Length: 0.
.
------------------------------------------------------------------------
U 2007/08/21 21:49:26.671382 %Asterisk_IP%:5060 -> %OpenSER_IP%:5060
BYE sip:davidloh@%UAC_WAN_IP%:2150 SIP/2.0.
Via: SIP/2.0/UDP %Asterisk_IP%:5060;branch=z9hG4bK127246b8;rport.
Route:
<sip:%OpenSER_IP%;r2=on;lr;ftag=3c021055;nat=yes>,<sip:%OpenSER_IP%:5061;transport=tls;r2=on;lr;ftag=3c021055;nat=yes>.
From: "001"<sip:001@%SIP_Domain%>;tag=as5b2df0bb.
To: "ser"<sip:davidloh@%SIP_Domain%>;tag=3c021055.
Call-ID: NTc0YmQxMDVlMzU4Y2U3OTE2ZGU1ZDRjMzE0ZGRhYzU..
CSeq: 102 BYE.
User-Agent: Asterisk PBX.
Max-Forwards: 70.
Content-Length: 0.
.
------------------------------------------------------------------------
As you can see the last few BYE request wasn't able to deliver to UAC
(or UAC failed to response),
and it kept looping. For me the packets look fine, really no clue what's
could go wrong here :(
Thanks,
David Loh
Daniel-Constantin Mierla wrote:
Hello,
please send the request uri as well and the details about the IP
communication (source/destination address, port).
use:
ngrep -qt -W byline port 5060
the replace the IPs with something suggestive to protect your privacy.
Daniel
On 08/21/07 14:28, David Loh wrote:
Hi Daniel,
Thanks for the reply, here's my INVITE and BYE packet, I tcpdump-ed
and and use wireshark to export as txt, so the format might be abit off.
1) Invite from OpenSER to Asterisk via port 5060, asterisk return a
200. And the call successfully established
------------------------------------------------------------------------
Record-Route: <sip:%OpenSER_IP%;r2=on;lr;ftag=1247871f;nat=yes>
Record-Route:
<sip:%OpenSER_IP%:5061;transport=tls;r2=on;lr;ftag=1247871f;nat=yes>
Via: SIP/2.0/UDP %OpenSER_IP%;branch=z9hG4bKe42e.ff11ae32.0;i=52
Via: SIP/2.0/TLS
%UAC_LAN_IP%:23323;received=%UAC_WAN_IP%;branch=z9hG4bK-d87543-2820ee009262b500-1--d87543-;rport=3135
Max-Forwards: 69
Contact: <sip:davidloh@%UAC_WAN_IP%:3135;transport=TLS>
To: "001"<sip:001@%SIP_Domain%>
From: "ser"<sip:davidloh@%SIP_Domain%>;tag=1247871f
Call-ID: Zjc4OWJhZGNiMDUxNjQ5Y2I2ZDU3NTc4NGY2Y2RmOGQ.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1004p stamp 31962
Content-Length: 239
------------------------------------------------------------------------
2) Here's the BYE, from Asterisk to OpenSER.
------------------------------------------------------------------------
Via: SIP/2.0/UDP %Asterisk_IP%:5060;branch=z9hG4bK0ca1ab09;rport
Route:
<sip:%OpenSER_IP%;r2=on;lr;ftag=1247871f;nat=yes>,<sip:%OpenSER_IP%:5061;transport=tls;r2=on;lr;ftag=1247871f;nat=yes>
From: "001"<sip:001@%SIP_Domain%>;tag=as0492a9d0
To: "ser"<sip:davidloh@%SIP_Domain%>;tag=1247871f
Call-ID: Zjc4OWJhZGNiMDUxNjQ5Y2I2ZDU3NTc4NGY2Y2RmOGQ.
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
------------------------------------------------------------------------
3) And then, OpenSER passing the BYE to UAC...
------------------------------------------------------------------------
Record-Route: <sip:%OpenSER_IP%;lr;ftag=as0492a9d0>
Via: SIP/2.0/UDP %OpenSER_IP%;branch=z9hG4bK5e.1195fd8.0
Via: SIP/2.0/UDP %Asterisk_IP%:5060;branch=z9hG4bK0ca1ab09;rport=5060
From: "001"<sip:001@%SIP_Domain%>;tag=as0492a9d0
To: "ser"<sip:davidloh@%SIP_Domain%>;tag=1247871f
Call-ID: Zjc4OWJhZGNiMDUxNjQ5Y2I2ZDU3NTc4NGY2Y2RmOGQ.
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 69
Content-Length: 0
------------------------------------------------------------------------
I think the "protocol/port mismatch" happened right after (3) because
from my ethereal trace,
after short while (perhaps timeout), Asterisk will re-initiate a
"BYE" and pass to OpenSER (packet #2), and OpenSER will forward it to
UAC (packet #3),
but somehow UAC doesn't response an "ACK" to OpenSER that why from my
ethereal trace, packet #2 and #3 kept looping.
Any thoughts ?
Thanks in advance.
Regards,
David Loh
Daniel-Constantin Mierla wrote:
Hello,
can you paste the BYE here (you can catch it with ngrep if it is not
coming via tls, e.g., from asterisk)? Also, the invite going to
asterisk will be good.
Cheers,
Daniel
On 08/20/07 07:50, David Loh wrote:
Hi All,
Good day, I'm new to this mailing list :-)
I've setup an OpenSER server with TLS implementation, so far I
tested everything works fine,
but there's strange problem for "BYE" request, from the openser's
log it always complaint:
"Aug 19 22:58:40 ser /sbin/openser[16171]: WARNING:get_send_socket:
protocol/port mismatch"
Here's my implementation:
UA -> OpenSER -> Asterisk (as RTP) -> OpenSER -> UA
On OpenSER I've two port opened which is Udp 5060 and TCP 5061 (TLS).
My UA is Eyebeam v1.5 and the CA certificate (self-signed) already
installed into IE "Trusted RootCA Certificates".
So far I've tested p2p and pstn calls (and call to playback
balance/asterisk), but the problem is unless UA disconnect the call
(initiate BYE),
otherwise if remote party (p2p/pstn/asterisk) initiate BYE request,
my UA will forever stay connected until I disconnect the call
manually.
So far I've identified the BYE request from remote party will go
into Loose Route, and I've place a xlog to display the R-Uri..
here's the log from openser: (notes: if UA dial '001' will playback
his/her balance from asterisk)
-------------------------------------------------------------------------------------------------------------
Aug 19 22:58:31 ser /sbin/openser[16189]: DBG: INVITE from client
sip:%user%@%domain% (%UA_IP%) - p2p call
Aug 19 22:58:31 ser /sbin/openser[16189]: DBG: INVITE
F-[sip:%user%@%domain%] T-[sip:001@%domain%] IP-[%UA_IP%] SUCCESSFUL
Aug 19 22:58:32 ser /sbin/openser[16189]: DBG: INVITE from client
sip:%user%@%domain% (%UA_IP%)
Aug 19 22:58:32 ser /sbin/openser[16189]: DBG: INVITE from client
sip:%user%@%domain% (%UA_IP%) - p2p call
Aug 19 22:58:32 ser /sbin/openser[16189]: DBG: INVITE
F-[sip:%user%@%domain%] T-[sip:001@%domain%] IP-[%UA_IP%] SUCCESSFUL
Aug 19 22:58:32 ser /sbin/openser[16189]: DBG:
[sip:%user%@%domain%](%UA_IP%) ACK [sip:001@%domain%] relayed LOOSE
ROUTE
Aug 19 22:58:32 ser /sbin/openser[16189]: DBG: ruri
[sip:001@%Asterisk_IP%] relayed LOOSE ROUTE
Aug 19 22:58:40 ser /sbin/openser[16171]: DBG:
[sip:001@%domain%](%Asterisk_IP%) BYE [sip:%user%@%domain%] relayed
LOOSE ROUTE
Aug 19 22:58:40 ser /sbin/openser[16171]: DBG: ruri
[sip:%user%@%domain%:%UA_Port%] relayed LOOSE ROUTE
Aug 19 22:58:40 ser /sbin/openser[16171]: WARNING:get_send_socket:
protocol/port mismatch <<<=============
Aug 19 22:58:41 ser /sbin/openser[16177]: DBG:
[sip:001@%domain%](%Asterisk_IP%) BYE [sip:%user%@%domain%] relayed
LOOSE ROUTE
Aug 19 22:58:41 ser /sbin/openser[16177]: DBG: ruri
[sip:%user%@%UA_IP%:%UA_Port%] relayed LOOSE ROUTE
Aug 19 22:58:42 ser /sbin/openser[16181]: DBG:
[sip:001@%domain%](%Asterisk_IP%) BYE [sip:%user%@%domain%] relayed
LOOSE ROUTE
Aug 19 22:58:42 ser /sbin/openser[16181]: DBG: ruri
[sip:%user%@%UA_IP%:%UA_Port%] relayed LOOSE ROUTE
....... (repeat the last 4 lines N times )
-------------------------------------------------------------------------------------------------------------
Below is my openser.cfg, loose route section
-------------------------------------------------------------------------------------------------------------
###############
# Loose route #
###############
if(loose_route()) {
if(has_totag() && (is_method("INVITE") ||
is_method("ACK"))) {
if(nat_uac_test("19") || search("^Route:.*;nat=yes")) {
fix_nated_contact();
}
}
xlog("L_INFO", "DBG: [$fu]($si) $rm [$tu] relayed LOOSE
ROUTE");
xlog("L_INFO", "DBG: ruri [$ru] relayed LOOSE ROUTE");
t_relay();
exit;
}
-------------------------------------------------------------------------------------------------------------
Had anyone encountered the same problems before ? Do you mind to
share your experience with me ?
Your help are greatly appreciated, thanks in advance.
Regards,
David Loh
_______________________________________________
Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
23 Aug
23 Aug
12:30 p.m.
Hi Daniel,
I've found this http://bugs.digium.com/view.php?id=1491 , as I'm using
quite ancient version of Asterisk (v1.2.19),
could it be the reason why is the BYE not able to deliver to UAC ?
Regards,
David Loh
David Loh wrote:
Hi Daniel,
Here's my ngrep result:
# ngrep -qt -W byline port 5060
interface: eth0 (FF.FFF.FFF.0/255.255.255.192)
filter: (ip or ip6) and ( port 5060 )
------------------------------------------------------------------------
* trimmed, please refer to earlier post*
.
------------------------------------------------------------------------
As you can see the last few BYE request wasn't able to deliver to UAC
(or UAC failed to response),
and it kept looping. For me the packets look fine, really no clue
what's could go wrong here :(
Thanks,
David Loh
Daniel-Constantin Mierla wrote:
Hello,
please send the request uri as well and the details about the IP
communication (source/destination address, port).
use:
ngrep -qt -W byline port 5060
the replace the IPs with something suggestive to protect your privacy.
Daniel
On 08/21/07 14:28, David Loh wrote:
------------------------------------------------------------------------
_______________________________________________
Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
Hi Daniel,
Thanks for the reply, here's my INVITE and BYE packet, I tcpdump-ed
and and use wireshark to export as txt, so the format might be abit
off.
1) Invite from OpenSER to Asterisk via port 5060, asterisk return a
200. And the call successfully established
------------------------------------------------------------------------
Record-Route: <sip:%OpenSER_IP%;r2=on;lr;ftag=1247871f;nat=yes>
Record-Route:
<sip:%OpenSER_IP%:5061;transport=tls;r2=on;lr;ftag=1247871f;nat=yes>
Via: SIP/2.0/UDP %OpenSER_IP%;branch=z9hG4bKe42e.ff11ae32.0;i=52
Via: SIP/2.0/TLS
%UAC_LAN_IP%:23323;received=%UAC_WAN_IP%;branch=z9hG4bK-d87543-2820ee009262b500-1--d87543-;rport=3135
Max-Forwards: 69
Contact: <sip:davidloh@%UAC_WAN_IP%:3135;transport=TLS>
To: "001"<sip:001@%SIP_Domain%>
From: "ser"<sip:davidloh@%SIP_Domain%>;tag=1247871f
Call-ID: Zjc4OWJhZGNiMDUxNjQ5Y2I2ZDU3NTc4NGY2Y2RmOGQ.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1004p stamp 31962
Content-Length: 239
------------------------------------------------------------------------
2) Here's the BYE, from Asterisk to OpenSER.
------------------------------------------------------------------------
Via: SIP/2.0/UDP %Asterisk_IP%:5060;branch=z9hG4bK0ca1ab09;rport
Route:
<sip:%OpenSER_IP%;r2=on;lr;ftag=1247871f;nat=yes>,<sip:%OpenSER_IP%:5061;transport=tls;r2=on;lr;ftag=1247871f;nat=yes>
From: "001"<sip:001@%SIP_Domain%>;tag=as0492a9d0
To: "ser"<sip:davidloh@%SIP_Domain%>;tag=1247871f
Call-ID: Zjc4OWJhZGNiMDUxNjQ5Y2I2ZDU3NTc4NGY2Y2RmOGQ.
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
------------------------------------------------------------------------
3) And then, OpenSER passing the BYE to UAC...
------------------------------------------------------------------------
Record-Route: <sip:%OpenSER_IP%;lr;ftag=as0492a9d0>
Via: SIP/2.0/UDP %OpenSER_IP%;branch=z9hG4bK5e.1195fd8.0
Via: SIP/2.0/UDP %Asterisk_IP%:5060;branch=z9hG4bK0ca1ab09;rport=5060
From: "001"<sip:001@%SIP_Domain%>;tag=as0492a9d0
To: "ser"<sip:davidloh@%SIP_Domain%>;tag=1247871f
Call-ID: Zjc4OWJhZGNiMDUxNjQ5Y2I2ZDU3NTc4NGY2Y2RmOGQ.
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 69
Content-Length: 0
------------------------------------------------------------------------
I think the "protocol/port mismatch" happened right after (3)
because from my ethereal trace,
after short while (perhaps timeout), Asterisk will re-initiate a
"BYE" and pass to OpenSER (packet #2), and OpenSER will forward it
to UAC (packet #3),
but somehow UAC doesn't response an "ACK" to OpenSER that why from
my ethereal trace, packet #2 and #3 kept looping.
Any thoughts ?
Thanks in advance.
Regards,
David Loh
Daniel-Constantin Mierla wrote:
Hello,
can you paste the BYE here (you can catch it with ngrep if it is
not coming via tls, e.g., from asterisk)? Also, the invite going to
asterisk will be good.
Cheers,
Daniel
On 08/20/07 07:50, David Loh wrote:
> Hi All,
>
> Good day, I'm new to this mailing list :-)
>
> I've setup an OpenSER server with TLS implementation, so far I
> tested everything works fine,
> but there's strange problem for "BYE" request, from the openser's
> log it always complaint:
> "Aug 19 22:58:40 ser /sbin/openser[16171]:
> WARNING:get_send_socket: protocol/port mismatch"
>
> Here's my implementation:
> UA -> OpenSER -> Asterisk (as RTP) -> OpenSER -> UA
> On OpenSER I've two port opened which is Udp 5060 and TCP 5061 (TLS).
> My UA is Eyebeam v1.5 and the CA certificate (self-signed) already
> installed into IE "Trusted RootCA Certificates".
> So far I've tested p2p and pstn calls (and call to playback
> balance/asterisk), but the problem is unless UA disconnect the
> call (initiate BYE),
> otherwise if remote party (p2p/pstn/asterisk) initiate BYE
> request, my UA will forever stay connected until I disconnect the
> call manually.
>
> So far I've identified the BYE request from remote party will go
> into Loose Route, and I've place a xlog to display the R-Uri..
> here's the log from openser: (notes: if UA dial '001' will
> playback his/her balance from asterisk)
>
-------------------------------------------------------------------------------------------------------------
>
> Aug 19 22:58:31 ser /sbin/openser[16189]: DBG: INVITE from client
> sip:%user%@%domain% (%UA_IP%) - p2p call
> Aug 19 22:58:31 ser /sbin/openser[16189]: DBG: INVITE
> F-[sip:%user%@%domain%] T-[sip:001@%domain%] IP-[%UA_IP%] SUCCESSFUL
> Aug 19 22:58:32 ser /sbin/openser[16189]: DBG: INVITE from client
> sip:%user%@%domain% (%UA_IP%)
> Aug 19 22:58:32 ser /sbin/openser[16189]: DBG: INVITE from client
> sip:%user%@%domain% (%UA_IP%) - p2p call
> Aug 19 22:58:32 ser /sbin/openser[16189]: DBG: INVITE
> F-[sip:%user%@%domain%] T-[sip:001@%domain%] IP-[%UA_IP%] SUCCESSFUL
> Aug 19 22:58:32 ser /sbin/openser[16189]: DBG:
> [sip:%user%@%domain%](%UA_IP%) ACK [sip:001@%domain%] relayed
> LOOSE ROUTE
> Aug 19 22:58:32 ser /sbin/openser[16189]: DBG: ruri
> [sip:001@%Asterisk_IP%] relayed LOOSE ROUTE
> Aug 19 22:58:40 ser /sbin/openser[16171]: DBG:
> [sip:001@%domain%](%Asterisk_IP%) BYE [sip:%user%@%domain%]
> relayed LOOSE ROUTE
> Aug 19 22:58:40 ser /sbin/openser[16171]: DBG: ruri
> [sip:%user%@%domain%:%UA_Port%] relayed LOOSE ROUTE
> Aug 19 22:58:40 ser /sbin/openser[16171]: WARNING:get_send_socket:
> protocol/port mismatch <<<=============
> Aug 19 22:58:41 ser /sbin/openser[16177]: DBG:
> [sip:001@%domain%](%Asterisk_IP%) BYE [sip:%user%@%domain%]
> relayed LOOSE ROUTE
> Aug 19 22:58:41 ser /sbin/openser[16177]: DBG: ruri
> [sip:%user%@%UA_IP%:%UA_Port%] relayed LOOSE ROUTE
> Aug 19 22:58:42 ser /sbin/openser[16181]: DBG:
> [sip:001@%domain%](%Asterisk_IP%) BYE [sip:%user%@%domain%]
> relayed LOOSE ROUTE
> Aug 19 22:58:42 ser /sbin/openser[16181]: DBG: ruri
> [sip:%user%@%UA_IP%:%UA_Port%] relayed LOOSE ROUTE
> ....... (repeat the last 4 lines N times )
>
-------------------------------------------------------------------------------------------------------------
>
> Below is my openser.cfg, loose route section
>
-------------------------------------------------------------------------------------------------------------
>
> ###############
> # Loose route #
> ###############
> if(loose_route()) {
> if(has_totag() && (is_method("INVITE") ||
is_method("ACK"))) {
> if(nat_uac_test("19") || search("^Route:.*;nat=yes"))
{
> fix_nated_contact();
> }
> }
>
> xlog("L_INFO", "DBG: [$fu]($si) $rm [$tu] relayed LOOSE
> ROUTE");
> xlog("L_INFO", "DBG: ruri [$ru] relayed LOOSE ROUTE");
> t_relay();
>
> exit;
> }
>
-------------------------------------------------------------------------------------------------------------
>
>
> Had anyone encountered the same problems before ? Do you mind to
> share your experience with me ?
> Your help are greatly appreciated, thanks in advance.
>
> Regards,
> David Loh
>
>
> _______________________________________________
> Users mailing list
> Users(a)openser.org
> http://openser.org/cgi-bin/mailman/listinfo/users
>
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participants (2)
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David Loh