Can you paste this output too?
lsmod | grep -i sip
-- Alex
--
Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard.
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/Vijay Thakur <vijay.thakur(a)loopmethods.com> wrote:Thanks for prompt reply. Here is the out put if command iptables -L -n:
Chain INPUT (policy ACCEPT)
target prot opt source destination
Chain FORWARD (policy ACCEPT)
target prot opt source destination
Chain OUTPUT (policy ACCEPT)
target prot opt source destination
Thanks for prompt reply. This is a Ubuntu 10.04 Kamailio 3.1 Server.
Vijay TH
-==============================================================================
Date: Mon, 06 Aug 2012 06:35:04 -0400
From: Alex Balashov <abalashov(a)evaristesys.com>
Subject: Re: [SR-Users] Kernel Droping SIP packet
To: sr-users(a)lists.sip-router.org
Message-ID: <17t1fd0fbly84yy2vuy8qlcp.1344249304618(a)email.android.com>
Content-Type: text/plain; charset="utf-8"
You might consider pasting the actual output of: iptables -L -n
This lists the actual rules straight from netfilter at runtime. I wouldn't worry too much about what some distro-specific config file or script says. Real truth comes from iptables itself.?
-- Alex
--
Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard.
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/Vijay Thakur <vijay.thakur(a)loopmethods.com> wrote:Hi all,
I have configure Kamailio 3.1.5 Server. All things are working fine.
When i make a call from Soft phone (X-Lite) to iphone, all is working
fine. But in other case call from iphone to Softphone is not working,
even not ringing. During checking the logs i am getting the error:
Aug? 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT=
MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77
DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF
PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763
RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A)
I have not implemented any firewall. You can check the out put of my
iptables : http://pastebin.com/i3zUfVeb
The SIP server is hosted at linnode.
With thanks in advance. Sorry dual posting.
Vijay TH
Ah, you short-circuited where I was going with this. Couldn't remember the name of the module.
Yep, Vijay. What Richard said.
-- Alex
--
Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard.
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/Richard Fuchs <rfuchs(a)sipwise.com> wrote:On 08/06/12 06:08, Vijay Thakur wrote:
> Hi all,
>
> I have configure Kamailio 3.1.5 Server. All things are working fine.
> When i make a call from Soft phone (X-Lite) to iphone, all is working
> fine. But in other case call from iphone to Softphone is not working,
> even not ringing. During checking the logs i am getting the error:
>
> Aug 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT=
> MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77
> DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF
> PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763
> RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A)
This is coming from nf_conntrack_sip, which is a netfilter connection
tracking kernel module for SIP. I've never used it, but judging from
what Google brings up, it seems to be very buggy. You should be able to
just unload it by issuing "rmmod nf_conntrack_sip". If that doesn't work
and/or if you want to keep it from auto-loading, you can blacklist it in
/etc/modprobe.d/ and then reboot.
HTH
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hi all,
I have configure Kamailio 3.1.5 Server. All things are working fine.
When i make a call from Soft phone (X-Lite) to iphone, all is working
fine. But in other case call from iphone to Softphone is not working,
even not ringing. During checking the logs i am getting the error:
Aug 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT=
MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77
DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF
PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763
RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A)
I have not implemented any firewall. You can check the out put of my
iptables : http://pastebin.com/i3zUfVeb
The SIP server is hosted at linnode.
With thanks in advance. Sorry dual posting.
Vijay TH
For your relatively narrow, specific applications in your topology, no. You'd be better off installing SEMS per se.
-- Alex
--
Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard.
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/phillman25 <phillman25(a)gmail.com> wrote:Hello Alex
Will try with SEMS first, found something called sip:provider CE v2.4 from http://www.sipwise.com/news/announcements/spce-v2_4-release/ if i'm not mistaken, this seems to combine Kamailio with SEMS? Do you think that this might be an easier installation rather than installing SEMS on its own as it seems to provide more documentation?
Thanks again!
On Mon, Aug 6, 2012 at 1:33 PM, Alex Balashov <abalashov(a)evaristesys.com> wrote:
The short answer to your latter question is: yes. Cisco media and PSTN gateways have never hairpinned SIP-to-SIP calls well, even when officially supported.
Asterisk has a lower learning curve due to the abundance of information and tutorials, but SEMS would make more sense, since all you need is a signaling B2BUA.
-- Alex
--
Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard.
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/
phillman25 <phillman25(a)gmail.com> wrote:
Hi Alex
Thanks for your prompt reply.
The PGW 2200 solution is used as our core PSTN gateway where its currently handling many SS7, H.323 and SIP interconnections. However, there are a few scenarios like the example described below, that the call is originating from Kamailio being sent to the PGW and then back to Kamailio for termination and this scenario doesn't seem to work.
Do you think that by implementing SEMS or Asterisk in between the PGW and Kamailio could resolve this issue for these specific scenarios?
From your experience what do you think would be a better solution?
Thanks again!
Phillip
========================
Message: 2
Date: Mon, 06 Aug 2012 04:26:28 -0400
From: Alex Balashov <abalashov(a)evaristesys.com>
Subject: Re: [SR-Users] B2BUA issues
To: sr-users(a)lists.sip-router.org
Message-ID: <501F7FB4.8040700(a)evaristesys.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
What is the larger objective? Are you using the PGW purely as a B2BUA?
If so, that's a colossally overblown waste of resources; just use
something like SEMS or Asterisk.
On 08/06/2012 04:24 AM, phillman25 wrote:
> Dear List
>
> I am trying to accomplish the following:
>
>
> Asterisk PABX (192.168.10.189) ==> Kamailio(xxx.xxx.xxx.xxx) ==> Cisco
> PGW 2200 (PSTN gateway) (yyy.yyy.yyy.yyy) ==>
> Kamailio(xxx.xxx.xxx.xxx) ==> Asterisk PABX (192.168.10.189)
>
> When trying the above scenario, the call is silent and drops after a few
> seconds. In syslog i observe the following error:
>
> *ERROR: <core> [parser/parse_rr.c:84]: parse_rr(): Error while parsing
> name-addr (sip:22030305@192.168.10.189:5060
> <http://sip:22030305@192.168.10.189:5060>>)*
>
> Looking at the sip trace i see that his might be caused by the ACK
> message received from the ASTERISK PABX? :
>
> ACK sip:22030305@192.168.10.189:5060
> <http://sip:22030305@192.168.10.189:5060> SIP/2.0
> Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK3c80f516;rport
> Route:
> <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea;did=97b.66623da5>,<sip:22030305@
> yyy.yyy.yyy.yyy;pgw-call=call-55bc4>,<sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea>
> Max-Forwards: 70
> From: "22498045" <sip:22498045@192.168.10.189
> <mailto:sip%3A22498045@192.168.10.189>>;tag=as166b1eea
> To: <sip:22030305@xxx.xxx.xxx.xxx>;tag=as6d578713
> Contact: <sip:22498045@192.168.10.189:5060
> <http://sip:22498045@192.168.10.189:5060>>
> Call-ID: 5e2d61160bd1bec9214e2d7d04e5a778@192.168.10.189:5060
> <http://5e2d61160bd1bec9214e2d7d04e5a778@192.168.10.189:5060>
> CSeq: 102 ACK
> User-Agent: FPBX-2.8.1(1.8.12.0)
> Content-Length: 0
>
>
> After contacting Cisco they informed us that issue is cause by B2BUA
> from our current version of Cisco PGW 2200 that doesn't support this
> feature. Is there a module, solution that i can implement on Kamailio
> that could temporarily resolve this issue?
>
> Thanking you in advance.
>
> Phillip
Thanks for prompt reply. Here is the out put if command iptables -L -n:
*Chain INPUT (policy ACCEPT)
target prot opt source destination
Chain FORWARD (policy ACCEPT)
target prot opt source destination
Chain OUTPUT (policy ACCEPT)
target prot opt source destination
Thanks for prompt reply. This is a Ubuntu 10.04 Kamailio 3.1 Server.
Vijay TH
-==============================================================================
*
Date: Mon, 06 Aug 2012 06:35:04 -0400
From: Alex Balashov<abalashov(a)evaristesys.com>
Subject: Re: [SR-Users] Kernel Droping SIP packet
To:sr-users@lists.sip-router.org
Message-ID:<17t1fd0fbly84yy2vuy8qlcp.1344249304618(a)email.android.com>
Content-Type: text/plain; charset="utf-8"
You might consider pasting the actual output of: iptables -L -n
This lists the actual rules straight from netfilter at runtime. I wouldn't worry too much about what some distro-specific config file or script says. Real truth comes from iptables itself.?
-- Alex
--
Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard.
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Web:http://www.evaristesys.com/Vijay Thakur<vijay.thakur(a)loopmethods.com> wrote:Hi all,
I have configure Kamailio 3.1.5 Server. All things are working fine.
When i make a call from Soft phone (X-Lite) to iphone, all is working
fine. But in other case call from iphone to Softphone is not working,
even not ringing. During checking the logs i am getting the error:
Aug? 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT=
MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77
DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF
PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763
RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A)
I have not implemented any firewall. You can check the out put of my
iptables :http://pastebin.com/i3zUfVeb
The SIP server is hosted at linnode.
With thanks in advance. Sorry dual posting.
Vijay TH
*
*
The short answer to your latter question is: yes. Cisco media and PSTN gateways have never hairpinned SIP-to-SIP calls well, even when officially supported.
Asterisk has a lower learning curve due to the abundance of information and tutorials, but SEMS would make more sense, since all you need is a signaling B2BUA.
-- Alex
--
Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard.
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/phillman25 <phillman25(a)gmail.com> wrote:Hi Alex
Thanks for your prompt reply.
The PGW 2200 solution is used as our core PSTN gateway where its currently handling many SS7, H.323 and SIP interconnections. However, there are a few scenarios like the example described below, that the call is originating from Kamailio being sent to the PGW and then back to Kamailio for termination and this scenario doesn't seem to work.
Do you think that by implementing SEMS or Asterisk in between the PGW and Kamailio could resolve this issue for these specific scenarios?
From your experience what do you think would be a better solution?
Thanks again!
Phillip
========================
Message: 2
Date: Mon, 06 Aug 2012 04:26:28 -0400
From: Alex Balashov <abalashov(a)evaristesys.com>
Subject: Re: [SR-Users] B2BUA issues
To: sr-users(a)lists.sip-router.org
Message-ID: <501F7FB4.8040700(a)evaristesys.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
What is the larger objective? Are you using the PGW purely as a B2BUA?
If so, that's a colossally overblown waste of resources; just use
something like SEMS or Asterisk.
On 08/06/2012 04:24 AM, phillman25 wrote:
> Dear List
>
> I am trying to accomplish the following:
>
>
> Asterisk PABX (192.168.10.189) ==> Kamailio(xxx.xxx.xxx.xxx) ==> Cisco
> PGW 2200 (PSTN gateway) (yyy.yyy.yyy.yyy) ==>
> Kamailio(xxx.xxx.xxx.xxx) ==> Asterisk PABX (192.168.10.189)
>
> When trying the above scenario, the call is silent and drops after a few
> seconds. In syslog i observe the following error:
>
> *ERROR: <core> [parser/parse_rr.c:84]: parse_rr(): Error while parsing
> name-addr (sip:22030305@192.168.10.189:5060
> <http://sip:22030305@192.168.10.189:5060>>)*
>
> Looking at the sip trace i see that his might be caused by the ACK
> message received from the ASTERISK PABX? :
>
> ACK sip:22030305@192.168.10.189:5060
> <http://sip:22030305@192.168.10.189:5060> SIP/2.0
> Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK3c80f516;rport
> Route:
> <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea;did=97b.66623da5>,<sip:22030305@
> yyy.yyy.yyy.yyy;pgw-call=call-55bc4>,<sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea>
> Max-Forwards: 70
> From: "22498045" <sip:22498045@192.168.10.189
> <mailto:sip%3A22498045@192.168.10.189>>;tag=as166b1eea
> To: <sip:22030305@xxx.xxx.xxx.xxx>;tag=as6d578713
> Contact: <sip:22498045@192.168.10.189:5060
> <http://sip:22498045@192.168.10.189:5060>>
> Call-ID: 5e2d61160bd1bec9214e2d7d04e5a778@192.168.10.189:5060
> <http://5e2d61160bd1bec9214e2d7d04e5a778@192.168.10.189:5060>
> CSeq: 102 ACK
> User-Agent: FPBX-2.8.1(1.8.12.0)
> Content-Length: 0
>
>
> After contacting Cisco they informed us that issue is cause by B2BUA
> from our current version of Cisco PGW 2200 that doesn't support this
> feature. Is there a module, solution that i can implement on Kamailio
> that could temporarily resolve this issue?
>
> Thanking you in advance.
>
> Phillip
You might consider pasting the actual output of: iptables -L -n
This lists the actual rules straight from netfilter at runtime. I wouldn't worry too much about what some distro-specific config file or script says. Real truth comes from iptables itself.
-- Alex
--
Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard.
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/Vijay Thakur <vijay.thakur(a)loopmethods.com> wrote:Hi all,
I have configure Kamailio 3.1.5 Server. All things are working fine.
When i make a call from Soft phone (X-Lite) to iphone, all is working
fine. But in other case call from iphone to Softphone is not working,
even not ringing. During checking the logs i am getting the error:
Aug 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT=
MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77
DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF
PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763
RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A)
I have not implemented any firewall. You can check the out put of my
iptables : http://pastebin.com/i3zUfVeb
The SIP server is hosted at linnode.
With thanks in advance. Sorry dual posting.
Vijay TH
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hi Alex
Thanks for your prompt reply.
The PGW 2200 solution is used as our core PSTN gateway where its currently
handling many SS7, H.323 and SIP interconnections. However, there are a few
scenarios like the example described below, that the call is originating
from Kamailio being sent to the PGW and then back to Kamailio for
termination and this scenario doesn't seem to work.
Do you think that by implementing SEMS or Asterisk in between the PGW and
Kamailio could resolve this issue for these specific scenarios?
>From your experience what do you think would be a better solution?
Thanks again!
Phillip
========================
Message: 2
Date: Mon, 06 Aug 2012 04:26:28 -0400
From: Alex Balashov <abalashov(a)evaristesys.com>
Subject: Re: [SR-Users] B2BUA issues
To: sr-users(a)lists.sip-router.org
Message-ID: <501F7FB4.8040700(a)evaristesys.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
What is the larger objective? Are you using the PGW purely as a B2BUA?
If so, that's a colossally overblown waste of resources; just use
something like SEMS or Asterisk.
On 08/06/2012 04:24 AM, phillman25 wrote:
> Dear List
>
> I am trying to accomplish the following:
>
>
> Asterisk PABX (192.168.10.189) ==> Kamailio(xxx.xxx.xxx.xxx) ==> Cisco
> PGW 2200 (PSTN gateway) (yyy.yyy.yyy.yyy) ==>
> Kamailio(xxx.xxx.xxx.xxx) ==> Asterisk PABX (192.168.10.189)
>
> When trying the above scenario, the call is silent and drops after a few
> seconds. In syslog i observe the following error:
>
> *ERROR: <core> [parser/parse_rr.c:84]: parse_rr(): Error while parsing
> name-addr (sip:22030305@192.168.10.189:5060
> <http://sip:22030305@192.168.10.189:5060>>)*
>
> Looking at the sip trace i see that his might be caused by the ACK
> message received from the ASTERISK PABX? :
>
> ACK sip:22030305@192.168.10.189:5060
> <http://sip:22030305@192.168.10.189:5060> SIP/2.0
> Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK3c80f516;rport
> Route:
> <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea;did=97b.
66623da5>,<sip:22030305@
> yyy.yyy.yyy.yyy;pgw-call=call-55bc4>,<sip:xxx.xxx.xxx.xxx;
lr=on;ftag=as166b1eea>
> Max-Forwards: 70
> From: "22498045" <sip:22498045@192.168.10.189
> <mailto:sip%3A22498045@192.168.10.189>>;tag=as166b1eea
> To: <sip:22030305@xxx.xxx.xxx.xxx>;tag=as6d578713
> Contact: <sip:22498045@192.168.10.189:5060
> <http://sip:22498045@192.168.10.189:5060>>
> Call-ID: 5e2d61160bd1bec9214e2d7d04e5a778@192.168.10.189:5060
> <http://5e2d61160bd1bec9214e2d7d04e5a778@192.168.10.189:5060>
> CSeq: 102 ACK
> User-Agent: FPBX-2.8.1(1.8.12.0)
> Content-Length: 0
>
>
> After contacting Cisco they informed us that issue is cause by B2BUA
> from our current version of Cisco PGW 2200 that doesn't support this
> feature. Is there a module, solution that i can implement on Kamailio
> that could temporarily resolve this issue?
>
> Thanking you in advance.
>
> Phillip
Dear List
I am trying to accomplish the following:
Asterisk PABX (192.168.10.189) ==> Kamailio(xxx.xxx.xxx.xxx) ==> Cisco PGW
2200 (PSTN gateway) (yyy.yyy.yyy.yyy) ==> Kamailio(xxx.xxx.xxx.xxx) ==>
Asterisk PABX (192.168.10.189)
When trying the above scenario, the call is silent and drops after a few
seconds. In syslog i observe the following error:
*ERROR: <core> [parser/parse_rr.c:84]: parse_rr(): Error while parsing
name-addr (sip:22030305@192.168.10.189:5060>)*
Looking at the sip trace i see that his might be caused by the ACK message
received from the ASTERISK PABX? :
ACK sip:22030305@192.168.10.189:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK3c80f516;rport
Route:
<sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea;did=97b.66623da5>,<sip:22030305@
yyy.yyy.yyy.yyy
;pgw-call=call-55bc4>,<sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea>
Max-Forwards: 70
From: "22498045" <sip:22498045@192.168.10.189>;tag=as166b1eea
To: <sip:22030305@xxx.xxx.xxx.xxx>;tag=as6d578713
Contact: <sip:22498045@192.168.10.189:5060>
Call-ID: 5e2d61160bd1bec9214e2d7d04e5a778@192.168.10.189:5060
CSeq: 102 ACK
User-Agent: FPBX-2.8.1(1.8.12.0)
Content-Length: 0
After contacting Cisco they informed us that issue is cause by B2BUA from
our current version of Cisco PGW 2200 that doesn't support this feature. Is
there a module, solution that i can implement on Kamailio that could
temporarily resolve this issue?
Thanking you in advance.
Phillip
Hi all,
I have a one question regarding ACK and Proxy-Authorization header.
I am testing Kamailio 3.3 as SIP proxy,default config. I made one call with
2 phones where one has Public IP while other is behind NAT. SIP clients are
Panasonic KX-UT133 and other is 1-NET (ex Sweden Mobile&CDMA provider ,over
50M users) Everything is working just fine, both RTP and SIP. On the other
hand one of the clients,1-NET one, send Proxy-Authorization in ACK when he
receives 200 OK from Kamailio.
PUBLIC_IP_USER1 - user 1000
PUBLIC_IP_USER2 - user 1001
KAMAILIO ==> 1-NET
SIP/2.0 200 OKr\n
Record-Route: <sip: <sip:KAMAILIO_PUBLIC_IP;lr=on;nat=yes>
KAMAILIO_PUBLIC_IP;lr=on;nat=yes>
Via: SIP/2.0/UDP PUBLIC_IP_USER1:5064;rport=5064;branch=z9hG4bK934894606
Call-ID: 221374358
From: blabla <sip:1000@ <sip:1000@KAMAILIO_PUBLIC_IP>
KAMAILIO_PUBLIC_IP>;tag=507511069
To: <sip:1001@ <sip:1001@KAMAILIO_PUBLIC_IP:5060>
KAMAILIO_PUBLIC_IP:5060>;tag=2148378512
CSeq: 21 INVITE
Allow: INVITE,ACK,CANCEL,BYE,INFO,UPDATE,OPTIONS,NOTIFY,REFER
Contact: <sip:1001@ <sip:1001@PUBLIC_IP_USER2:1027>
PUBLIC_IP_USER2:1027>
Require: replaces
Content-Type: application/sdp
Server: Panasonic_KX-UT133NE/01.081 (0080f0cedd83)
Content-Length: 182
v=0
o=- 1343736535 1343736535 IN IP4 KAMAILIO_PUBLIC_IP
s=-
c=IN IP4 KAMAILIO_PUBLIC_IP
t=0 0
m=audio 20412 RTP/AVP 18
a=rtpmap:18 G729/8000
a=sendrecv
a=ptime:20
a=nortpproxy:yes
1-NET ==> KAMAILIO
ACK sip:1001@ <sip:1001@PUBLIC_IP_USER2:1027> PUBLIC_IP_USER2:1027
SIP/2.0
Via: SIP/2.0/UDP PUBLIC_IP_USER1:5064;rport;branch=z9hG4bK642028490
Route: <sip: <sip:KAMAILIO_PUBLIC_IP;lr=on;nat=yes>
KAMAILIO_PUBLIC_IP;lr=on;nat=yes>
From: blabla <sip:1000@ <sip:1000@KAMAILIO_PUBLIC_IP>
KAMAILIO_PUBLIC_IP>;tag=507511069
To: <sip:1001@ <sip:1001@KAMAILIO_PUBLIC_IP:5060>
KAMAILIO_PUBLIC_IP:5060>;tag=2148378512
Call-ID: 221374358
CSeq: 21 ACK
Contact: <sip:1000@ <sip:1000@PUBLIC_IP_USER1:5064>
PUBLIC_IP_USER1:5064>
[truncated] Proxy-Authorization: Digest username="1000",
realm="KAMAILIO_PUBLIC_IP", nonce="UBfL8lAXysb5tJCs80ZnthyPl9IzmRZk",
uri="sip:1001@KAMAILIO_PUBLIC_IP:5060",
response="7824519cdad9f1c2c79027a2d7522344", algorithm=MD5,
cnonce="0a4f113b", q
Max-Forwards: 70
User-Agent: Serbia_2.00
Content-Length: 0
I attach txt file with call flow, I can send pcap also.
I think that the issue is related with bad client but I need another
opinion.
Does anyone has an idea about this issue?
Best Regards,
Ivan
