Call is terminated in callee side after 36s @Kamailio 3.1 @debian sqeeze64

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Coca

21 Jul 2011 21 Jul '11

10:32 a.m.

Hi List, I have a Kamailio3.1 server and RTPProxy running in WAN. The calls between UA will automatically terminated in Callee UA side 36s after connected, while no one sends a BYE. While Kamailio and UA are in LAN at all , everything is just working well. Is it my rtpproxy doesn't working correctly or something else is wrong? What can I do to fix it. Any hint?? BTW, Kamailio is installed following official guide http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git The kamailio.cfg wasn't changed at all except for below: ----------------------------------------------------------------------------------------- 1) adding the following lines: #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_NAT 2)uncommenting the line below in route[REGISTRAR], setbflag(FLB_NATSIPPING); 3)change rtpproxy port corresponding modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222") ----------------------------------------------------------------------------------------- and my rtpproxy1.2.1 was installed by apt-get install rtpproxy, with 1) /etc/default/rtpproxy changed into: # Defaults for rtpproxy # The control socket. #CONTROL_SOCK="unix:/var/run/rtpproxy/rtpproxy.sock" # To listen on an UDP socket, uncomment this line: CONTROL_SOCK=udp:127.0.0.1:22222 LISTEN_ADDR=xx.xx.xx.xx # Additional options that are passed to the daemon. EXTRA_OPTS="-l ${LISTEN_ADDR}" 2) and started by rtpproxy -l xx.xx.xx.xx -s udp:localhost:22222 -u kamailio Your help will be great appreciated. Coca

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Klaus Darilion

21 Jul 21 Jul

11:47 a.m.

This sounds like a NAT problem, where the callee does not receive the ACK request (INVITE-200OK-ACK). regards klaus Am 21.07.2011 10:32, schrieb Coca:

...

Hi List, I have a Kamailio3.1 server and RTPProxy running in WAN. The calls between UA will automatically terminated in Callee UA side 36s after connected, while no one sends a BYE. While Kamailio and UA are in LAN at all , everything is just working well. Is it my rtpproxy doesn't working correctly or something else is wrong? What can I do to fix it. Any hint?? BTW, Kamailio is installed following official guide http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git The kamailio.cfg wasn't changed at all except for below: ----------------------------------------------------------------------------------------- 1) adding the following lines: #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_NAT 2)uncommenting the line below in route[REGISTRAR], setbflag(FLB_NATSIPPING); 3)change rtpproxy port corresponding modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222 <http://127.0.0.1:22222>") ----------------------------------------------------------------------------------------- and my rtpproxy1.2.1 was installed by apt-get install rtpproxy, with 1) /etc/default/rtpproxy changed into: # Defaults for rtpproxy # The control socket. #CONTROL_SOCK="unix:/var/run/rtpproxy/rtpproxy.sock" # To listen on an UDP socket, uncomment this line: CONTROL_SOCK=udp:127.0.0.1:22222 <http://127.0.0.1:22222> LISTEN_ADDR=xx.xx.xx.xx # Additional options that are passed to the daemon. EXTRA_OPTS="-l ${LISTEN_ADDR}" 2) and started by rtpproxy -l xx.xx.xx.xx -s udp:localhost:22222 -u kamailio Your help will be great appreciated. Coca _______________________________________________ sr-dev mailing list sr-dev(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev

Coca

12:33 p.m.

New subject: Call is terminated in callee side after 36s @Kamailio 3.1 @debian sqeeze64

Dear Klaus, Since I have Usrloc record made for registration of myUA behind nat looks like: Contact: <sip:1234@192.168.10.50:2305 ;transport=TCP;line=7e1d8b95f65b25a>;expires=600;received="sip:27.96.63.122:49202 ;transport=TCP" I thought my rtpproxy is running. However , the call can be established even without NAT enable, and it also ends unusually after 36s. Attachment is the ngrep log in my Kamailio server side on 5060 port. ( I have replaced my server ip as xx.xx.xx.xx and the UA name as myUA) Any hint will be great appreciated. Coca 2011/7/21 Klaus Darilion <klaus.mailinglists(a)pernau.at>

...

This sounds like a NAT problem, where the callee does not receive the ACK request (INVITE-200OK-ACK). regards klaus Am 21.07.2011 10:32, schrieb Coca:

well.

Is it my rtpproxy doesn't working correctly or something else is wrong? What can I do to fix it. Any hint?? BTW, Kamailio is installed following official guide

http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git

The kamailio.cfg wasn't changed at all except for below:

-----------------------------------------------------------------------------------------

1) adding the following lines: #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_NAT 2)uncommenting the line below in route[REGISTRAR], setbflag(FLB_NATSIPPING); 3)change rtpproxy port corresponding modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222 <http://127.0.0.1:22222>")

-----------------------------------------------------------------------------------------

and my rtpproxy1.2.1 was installed by apt-get install rtpproxy, with 1) /etc/default/rtpproxy changed into: # Defaults for rtpproxy # The control socket. #CONTROL_SOCK="unix:/var/run/rtpproxy/rtpproxy.sock" # To listen on an UDP socket, uncomment this line: CONTROL_SOCK=udp:127.0.0.1:22222 <http://127.0.0.1:22222> LISTEN_ADDR=xx.xx.xx.xx # Additional options that are passed to the daemon. EXTRA_OPTS="-l ${LISTEN_ADDR}" 2) and started by rtpproxy -l xx.xx.xx.xx -s udp:localhost:22222 -u kamailio Your help will be great appreciated. Coca _______________________________________________ sr-dev mailing list sr-dev(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev

_______________________________________________ sr-dev mailing list sr-dev(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev

-- -------------------------- Room to Fly, Endless Sky! --Yi Chen

Klaus Darilion

1:58 p.m.

New subject: Call is terminated in callee side after 36s @Kamailio 3.1 @debian sqeeze64

IF you take a look at the trace you see, that the caller does not send ACK after receiving 200 OK. Either it does not send ACK or it sends it to wrong destination. Trace at the caller's phone and watch log file of caller. regards Klaus Am 21.07.2011 12:33, schrieb Coca:

...

Dear Klaus, Since I have Usrloc record made for registration of myUA behind nat looks like: Contact: <sip:1234@192.168.10.50:2305;transport=TCP;line=7e1d8b95f65b25a>;expires=600;received="sip:27.96.63.122:49202;transport=TCP" I thought my rtpproxy is running. However , the call can be established even without NAT enable, and it also ends unusually after 36s. Attachment is the ngrep log in my Kamailio server side on 5060 port. ( I have replaced my server ip as xx.xx.xx.xx and the UA name as myUA) Any hint will be great appreciated. Coca 2011/7/21 Klaus Darilion <klaus.mailinglists(a)pernau.at <mailto:klaus.mailinglists@pernau.at>> This sounds like a NAT problem, where the callee does not receive the ACK request (INVITE-200OK-ACK). regards klaus Am 21.07.2011 10:32, schrieb Coca:

Hi List, I have a Kamailio3.1 server and RTPProxy running in WAN. The calls between UA will automatically terminated in Callee UA

side 36s

after connected, while no one sends a BYE. While Kamailio and UA are in LAN at all , everything is just

working well.

Is it my rtpproxy doesn't working correctly or something else is

wrong?

What can I do to fix it. Any hint?? BTW, Kamailio is installed following official guide

http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git

The kamailio.cfg wasn't changed at all except for below:

-----------------------------------------------------------------------------------------

<http://127.0.0.1:22222>

<http://127.0.0.1:22222>")

-----------------------------------------------------------------------------------------

<http://127.0.0.1:22222>

LISTEN_ADDR=xx.xx.xx.xx # Additional options that are passed to the daemon. EXTRA_OPTS="-l ${LISTEN_ADDR}" 2) and started by rtpproxy -l xx.xx.xx.xx -s udp:localhost:22222 -u kamailio Your help will be great appreciated. Coca _______________________________________________ sr-dev mailing list sr-dev(a)lists.sip-router.org <mailto:sr-dev@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev

_______________________________________________ sr-dev mailing list sr-dev(a)lists.sip-router.org <mailto:sr-dev@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev -- -------------------------- Room to Fly, Endless Sky! --Yi Chen _______________________________________________ sr-dev mailing list sr-dev(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev

Coca

23 Jul 23 Jul

6:49 a.m.

New subject: Call is terminated in callee side after 36s @Kamailio 3.1 @debian sqeeze64

...

Dear Klaus, Since I have Usrloc record made for registration of myUA behind nat looks like: Contact: <sip:1234@192.168.10.50:2305

;transport=TCP;line=7e1d8b95f65b25a>;expires=600;received="sip:27.96.63.122:49202 ;transport=TCP"

I thought my rtpproxy is running. However , the call can be established even without NAT enable, and it also ends unusually after 36s. Attachment is the ngrep log in my Kamailio server side on 5060 port. ( I have replaced my server ip as xx.xx.xx.xx and the UA name as myUA) Any hint will be great appreciated. Coca 2011/7/21 Klaus Darilion <klaus.mailinglists(a)pernau.at <mailto:klaus.mailinglists@pernau.at>> This sounds like a NAT problem, where the callee does not receive the ACK request (INVITE-200OK-ACK). regards klaus Am 21.07.2011 10:32, schrieb Coca:

Hi List, I have a Kamailio3.1 server and RTPProxy running in WAN. The calls between UA will automatically terminated in Callee UA

side 36s

after connected, while no one sends a BYE. While Kamailio and UA are in LAN at all , everything is just

working well.

Is it my rtpproxy doesn't working correctly or something else is

wrong?

What can I do to fix it. Any hint?? BTW, Kamailio is installed following official guide

http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git

The kamailio.cfg wasn't changed at all except for below:

-----------------------------------------------------------------------------------------

<http://127.0.0.1:22222>

<http://127.0.0.1:22222>")

-----------------------------------------------------------------------------------------

<http://127.0.0.1:22222>

_______________________________________________ sr-dev mailing list sr-dev(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev

-- -------------------------- Room to Fly, Endless Sky! --Yi Chen

Klaus Darilion

11:35 a.m.

New subject: Call is terminated in callee side after 36s @Kamailio 3.1 @debian sqeeze64

If it is "your" client, then you should be able to debug your client to find out if ACK is sent, and if yes, where the ACK is sent to. I inspected the last trace and I didn't spotted an error. regards klaus Coca wrote:

...

Dear Klaus, Thank you for your email. I tried using my UA connecting to the servers provided by other free sip service provider like Opensips in WAN. And the same problem didn't happen. My UA did sent ACK after the 200 OK in the invite process. It looks like the problem is my Kamailio server 's NAT setting. I wonder how I can do to verify whether the NAT setting is correctly done. What kind of method did you use in this issue? Your help will be great appreciated. Coca 2011/7/21 Klaus Darilion <klaus.mailinglists(a)pernau.at <mailto:klaus.mailinglists@pernau.at>> IF you take a look at the trace you see, that the caller does not send ACK after receiving 200 OK. Either it does not send ACK or it sends it to wrong destination. Trace at the caller's phone and watch log file of caller. regards Klaus Am 21.07.2011 12:33, schrieb Coca:

Dear Klaus, Since I have Usrloc record made for registration of myUA behind nat looks like: Contact:

<sip:1234@192.168.10.50:2305;transport=TCP;line=7e1d8b95f65b25a>;expires=600;received="sip:27.96.63.122:49202;transport=TCP"

myUA)

Any hint will be great appreciated. Coca 2011/7/21 Klaus Darilion <klaus.mailinglists(a)pernau.at

<mailto:klaus.mailinglists@pernau.at>

<mailto:klaus.mailinglists@pernau.at

<mailto:klaus.mailinglists@pernau.at>>>

This sounds like a NAT problem, where the callee does not

receive the

ACK request (INVITE-200OK-ACK). regards klaus Am 21.07.2011 10:32, schrieb Coca:

Hi List, I have a Kamailio3.1 server and RTPProxy running in WAN. The calls between UA will automatically terminated in Callee UA

side 36s

after connected, while no one sends a BYE. While Kamailio and UA are in LAN at all , everything is just

working well. > > Is it my rtpproxy doesn't working correctly or something

else is

wrong?

What can I do to fix it. Any hint?? BTW, Kamailio is installed following official guide

http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git

The kamailio.cfg wasn't changed at all except for below:

-----------------------------------------------------------------------------------------

> 1) adding the following lines: > #!define WITH_MYSQL > #!define WITH_AUTH > #!define WITH_USRLOCDB > #!define WITH_NAT > > 2)uncommenting the line below in route[REGISTRAR], > setbflag(FLB_NATSIPPING); > > 3)change rtpproxy port corresponding > modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222

<http://127.0.0.1:22222>

<http://127.0.0.1:22222>")

-----------------------------------------------------------------------------------------

> > and my rtpproxy1.2.1 was installed by apt-get install rtpproxy, > with > 1) /etc/default/rtpproxy changed into: > > # Defaults for rtpproxy > > > # The control socket. > > #CONTROL_SOCK="unix:/var/run/rtpproxy/rtpproxy.sock" > > # To listen on an UDP socket, uncomment this line: > > CONTROL_SOCK=udp:127.0.0.1:22222 <http://127.0.0.1:22222>

<http://127.0.0.1:22222>

<http://127.0.0.1:22222> > > LISTEN_ADDR=xx.xx.xx.xx > > > # Additional options that are passed to the daemon. > > EXTRA_OPTS="-l ${LISTEN_ADDR}" > > 2) and started by > rtpproxy -l xx.xx.xx.xx -s udp:localhost:22222 -u kamailio > > > > Your help will be great appreciated. > > Coca > > > > _______________________________________________ > sr-dev mailing list > sr-dev(a)lists.sip-router.org

<mailto:sr-dev@lists.sip-router.org> <mailto:sr-dev@lists.sip-router.org <mailto:sr-dev@lists.sip-router.org>>

http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev

_______________________________________________ sr-dev mailing list sr-dev(a)lists.sip-router.org

<mailto:sr-dev@lists.sip-router.org> <mailto:sr-dev@lists.sip-router.org <mailto:sr-dev@lists.sip-router.org>>

http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev -- -------------------------- Room to Fly, Endless Sky! --Yi Chen _______________________________________________ sr-dev mailing list sr-dev(a)lists.sip-router.org <mailto:sr-dev@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev

_______________________________________________ sr-dev mailing list sr-dev(a)lists.sip-router.org <mailto:sr-dev@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev -- -------------------------- Room to Fly, Endless Sky! --Yi Chen ------------------------------------------------------------------------ _______________________________________________ sr-dev mailing list sr-dev(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev

Coca

5:54 p.m.

New subject: Call is terminated in callee side after 36s @Kamailio 3.1 @debian sqeeze64

Dear Klaus, Sorry for the late response. I check the log of the caller and found out: The message header of 200 OK (with session description) from the server to the caller , right before the RTP starts, has the value <sip:10.150.175.210;lr=on;nat=yes> of Record-Route . Then the caller sent ACK to ip 10.150.175.210. The ip 10.150.175.210 is the private ip of the amazon server I am using , which is unreachable for the caller. Is it kinda installing Kamailio inside the NAT issue? What should I do? Your help is great appreciated. Thanks, Coca 2011/7/23 Klaus Darilion <klaus.mailinglists(a)pernau.at>

...

Dear Klaus, Thank you for your email. I tried using my UA connecting to the servers provided by other free sip service provider like Opensips in WAN. And the same problem didn't happen. My UA did sent ACK after the 200 OK in the invite process. It looks like the problem is my Kamailio server 's NAT setting. I wonder how I can do to verify whether the NAT setting is correctly done. What kind of method did you use in this issue? Your help will be great appreciated. Coca 2011/7/21 Klaus Darilion <klaus.mailinglists(a)pernau.at <mailto: klaus.mailinglists(a)**pernau.at <klaus.mailinglists(a)pernau.at>>> IF you take a look at the trace you see, that the caller does not send ACK after receiving 200 OK. Either it does not send ACK or it sends it to wrong destination. Trace at the caller's phone and watch log file of caller. regards Klaus Am 21.07.2011 12:33, schrieb Coca:

Dear Klaus, Since I have Usrloc record made for registration of myUA behind nat looks like: Contact:

<sip:1234@192.168.10.50:2305;**transport=TCP;line=** 7e1d8b95f65b25a>;expires=600;**received="sip:27.96.63.122:** 49202;transport=TCP"

I thought my rtpproxy is running. However , the call can be established even without NAT enable, and

also ends unusually after 36s. Attachment is the ngrep log in my Kamailio server side on 5060 port. ( I have replaced my server ip as xx.xx.xx.xx and the UA name as

myUA)

Any hint will be great appreciated. Coca 2011/7/21 Klaus Darilion <klaus.mailinglists(a)pernau.at

<mailto:klaus.mailinglists@**pernau.at <klaus.mailinglists(a)pernau.at>>

<mailto:klaus.mailinglists@**pernau.at<klaus.mailinglists@pernau.at>

<mailto:klaus.mailinglists@**pernau.at <klaus.mailinglists(a)pernau.at>

This sounds like a NAT problem, where the callee does not

receive the

ACK request (INVITE-200OK-ACK). regards klaus Am 21.07.2011 10:32, schrieb Coca: > Hi List,

> I have a Kamailio3.1 server and RTPProxy running in WAN.

> The calls between UA will automatically terminated in Callee

side 36s

after connected, while no one sends a BYE. While Kamailio and UA are in LAN at all , everything is just

working well.

> Is it my rtpproxy doesn't working correctly or something

else is

wrong?

What can I do to fix it. Any hint?? BTW, Kamailio is installed following official guide

http://www.kamailio.org/**dokuwiki/doku.php/install:**

kamailio-3.1.x-from-git<http://www.kamailio.org/dokuwiki/doku.php/instal…

The kamailio.cfg wasn't changed at all except for below:

------------------------------**

------------------------------**-----------------------------

> 1) adding the following lines: > #!define WITH_MYSQL > #!define WITH_AUTH > #!define WITH_USRLOCDB > #!define WITH_NAT

> 2)uncommenting the line below in route[REGISTRAR], > setbflag(FLB_NATSIPPING);

> 3)change rtpproxy port corresponding > modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222

<http://127.0.0.1:22222>

<http://127.0.0.1:22222>")

------------------------------**

------------------------------**-----------------------------

> and my rtpproxy1.2.1 was installed by apt-get install

rtpproxy,

> with > 1) /etc/default/rtpproxy changed into:

> # Defaults for rtpproxy

> # The control socket.

> #CONTROL_SOCK="unix:/var/run/**rtpproxy/rtpproxy.sock"

> # To listen on an UDP socket, uncomment this line:

> CONTROL_SOCK=udp:127.0.0.1:**22222 <http://127.0.0.1:22222> <

http://127.0.0.1:22222> <http://127.0.0.1:22222>

<http://127.0.0.1:22222>

> LISTEN_ADDR=xx.xx.xx.xx

> # Additional options that are passed to the daemon.

> EXTRA_OPTS="-l ${LISTEN_ADDR}"

> 2) and started by > rtpproxy -l xx.xx.xx.xx -s udp:localhost:22222 -u kamailio

> Your help will be great appreciated.

> Coca

> ______________________________**_________________ > sr-dev mailing list > sr-dev(a)lists.sip-router.org

<mailto:sr-dev@lists.sip-**router.org <sr-dev(a)lists.sip-router.org>> <mailto:sr-dev@lists.sip-**router.org <sr-dev(a)lists.sip-router.org> <mailto:sr-dev@lists.sip-**router.org <sr-dev(a)lists.sip-router.org>>>

> http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**

dev <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev>

______________________________**_________________ sr-dev mailing list sr-dev(a)lists.sip-router.org

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-- -------------------------- Room to Fly, Endless Sky! --Yi Chen

Klaus Darilion

25 Jul 25 Jul

10:02 a.m.

New subject: Call is terminated in callee side after 36s @Kamailio 3.1 @debian sqeeze64

Am 23.07.2011 17:54, schrieb Coca:

...

Is it kinda installing Kamailio inside the NAT issue?

Coca

11:39 a.m.

New subject: Call is terminated in callee side after 36s @Kamailio 3.1 @debian sqeeze64

Dear Klaus, Thanks to your advice. I've already had it resolved. Coca 2011/7/25 Klaus Darilion <klaus.mailinglists(a)pernau.at>

...

Am 23.07.2011 17:54, schrieb Coca:

Is it kinda installing Kamailio inside the NAT issue?

Yes. You have to set http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:3.1.x#advertised_ad… and add the public IP address to alias=.... Basically a SIP proxy behind NAT sucks and may you give some more headaches. regards klaus _______________________________________________ sr-dev mailing list sr-dev(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev

-- -------------------------- Room to Fly, Endless Sky! --Yi Chen

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Coca
Klaus Darilion