Dear Kamailio users,
I am trying to set up a simple scenario as follows:
UAC --> Proxy_1 --> Proxy_2 --> Proxy_3 --> UAC
I am new to Kamailio and had the following basic questions that came to my mind after reading the documentation an the default kamilio.cfg config script:
1) Is there any good example of the above scenario in which the UAC and UAS (caller and callee) are modeled using SIPp and proxies are simple kamailio instances run on different machines that simply forward the SIP packets?
2) In specific, I want the proxies to simply forward the messages to their downstream (I do not care about registration or other operations and look for a minimally working simple SIP scenario). Is there any example of such configuration?
3) Let's imagine Proxy_2 in the above example, in the default config, it looks like that with modifying line 164, "listen=udp:10.0.0.10:5060" I can specify the address of its upstream, Proxy_1, but how and where shall I set the specifications of the downstream, Proxy_3?
I suspect it is somewhere in the Routing Logic block (line 449) but not sure how it is exactly done. I see a "route(SIPOUT)" call for example but I am not sure how and where the value of SIPOUT is modified. Is it a representation of the downstream servers in the config file?
Thanks, Alireza
Hello,
listen is to specify local ip address or network interface on which kamailio should listen for sip traffic.
To send to an ip address there are couple of variants, in config file:
$ru = "sip:" + $rU + "@__NEXT_PROXY_IP__"; t_relay(); exit;
Or, if you don't want to change the r-uri, then use:
$du = "sip:__NEXT_PROXY_IP__"; t_relay(); exit;
Of course, you have to replace the __NEXT_PROXY_IP__ with the appropriate value.
More dynamic option would be using dispatcher module.
Cheers, Daniel
On 15/06/14 15:28, AliReza Khoshgoftar Monfared wrote:
Dear Kamailio users,
I am trying to set up a simple scenario as follows:
UAC --> Proxy_1 --> Proxy_2 --> Proxy_3 --> UAC
I am new to Kamailio and had the following basic questions that came to my mind after reading the documentation an the default kamilio.cfg config script:
- Is there any good example of the above scenario in which the UAC
and UAS (caller and callee) are modeled using SIPp and proxies are simple kamailio instances run on different machines that simply forward the SIP packets?
- In specific, I want the proxies to simply forward the messages to
their downstream (I do not care about registration or other operations and look for a minimally working simple SIP scenario). Is there any example of such configuration?
- Let's imagine Proxy_2 in the above example, in the default config,
it looks like that with modifying line 164, "listen=udp:10.0.0.10:5060 http://10.0.0.10:5060" I can specify the address of its upstream, Proxy_1, but how and where shall I set the specifications of the downstream, Proxy_3?
I suspect it is somewhere in the Routing Logic block (line 449) but not sure how it is exactly done. I see a "route(SIPOUT)" call for example but I am not sure how and where the value of SIPOUT is modified. Is it a representation of the downstream servers in the config file?
Thanks, Alireza
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Thanks very much Daniel,
This is solving a part of my issue for now. Here is the routing section of my configuration file (it may be good to share it once complete as a minimally working proxy):
route{
if (!mf_process_maxfwd_header("10")) { sl_reply("483","Too Many Hops"); break; } if (msg:len >= 4096 ) { sl_reply("513", "Message too big");
}
if (!method=="REGISTER") record_route();
if (!loose_route()) {
$du = "sip:10.236.214.86:5060"; #setdsturi("sip:10.236.214.86:5060"); #second way to do it if (!t_relay()) { sl_reply_error(); }
}
}
I am trying to set up a kamailio proxy on an EC2 VM using this configurations and have been running a UAC and a UAS using SIPp on two other machines (UAC <--> Kamailio Proxy <--> UAS):
UAC
sipp -sn uac -nr -r 1 -rp 1000 -d 0 -l 1 -p 5060 -trace_msg -i SELF_IP(UAC) -rsa PROXY_IP:5060 UAS_IP:5060
UAS:
sipp -sn uas -d 0 -p 5060 -i SELF_IP(UAS) -rsa PRXY_IP:5060 -trace_msg
It looks that messages go through, and are received by the server, but what I get back at the UAC (client) is a "dead call" error:
Last Error: Dead call 1-1734@10.140.34.188 (aborted at index 8), receive...
Is there a specific meaning to this "dead call" error? Is there anything that my proxy is missing in its routing or does it have to do with the UAC/UAS configs?
Thanks Alireza
On Tue, Jun 17, 2014 at 5:11 AM, Daniel-Constantin Mierla <miconda@gmail.com
wrote:
Hello,
listen is to specify local ip address or network interface on which kamailio should listen for sip traffic.
To send to an ip address there are couple of variants, in config file:
$ru = "sip:" + $rU + "@__NEXT_PROXY_IP__"; t_relay(); exit;
Or, if you don't want to change the r-uri, then use:
$du = "sip:__NEXT_PROXY_IP__"; t_relay(); exit;
Of course, you have to replace the __NEXT_PROXY_IP__ with the appropriate value.
More dynamic option would be using dispatcher module.
Cheers, Daniel
On 15/06/14 15:28, AliReza Khoshgoftar Monfared wrote:
Dear Kamailio users,
I am trying to set up a simple scenario as follows:
UAC --> Proxy_1 --> Proxy_2 --> Proxy_3 --> UAC
I am new to Kamailio and had the following basic questions that came to my mind after reading the documentation an the default kamilio.cfg config script:
- Is there any good example of the above scenario in which the UAC and
UAS (caller and callee) are modeled using SIPp and proxies are simple kamailio instances run on different machines that simply forward the SIP packets?
- In specific, I want the proxies to simply forward the messages to
their downstream (I do not care about registration or other operations and look for a minimally working simple SIP scenario). Is there any example of such configuration?
- Let's imagine Proxy_2 in the above example, in the default config, it
looks like that with modifying line 164, "listen=udp:10.0.0.10:5060" I can specify the address of its upstream, Proxy_1, but how and where shall I set the specifications of the downstream, Proxy_3?
I suspect it is somewhere in the Routing Logic block (line 449) but not sure how it is exactly done. I see a "route(SIPOUT)" call for example but I am not sure how and where the value of SIPOUT is modified. Is it a representation of the downstream servers in the config file?
Thanks, Alireza
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Shedding some more light on the situation, here are some further facts I could figure out:
1) I did the same test without the proxy in between (UAC <--> UAS) purely in SIPp, everything was fine, no dead call error
2) With the Kamailio proxy in between, I can see that there is a problem in call termination. The client is not receiving any "200 OK" for the "BYE"s it is sending.
Is there anything wrong in my routing logic that is preventing these ACKs?
On Thu, Jun 19, 2014 at 4:09 PM, AliReza Khoshgoftar Monfared < khoshgoftar@gmail.com> wrote:
Thanks very much Daniel,
This is solving a part of my issue for now. Here is the routing section of my configuration file (it may be good to share it once complete as a minimally working proxy):
route{
if (!mf_process_maxfwd_header("10")) { sl_reply("483","Too Many Hops"); break; } if (msg:len >= 4096 ) { sl_reply("513", "Message too big");}if (!method=="REGISTER") record_route();if (!loose_route()) {$du = "sip:10.236.214.86:5060"; #setdsturi("sip:10.236.214.86:5060"); #second way to doit if (!t_relay()) { sl_reply_error(); }
}}
I am trying to set up a kamailio proxy on an EC2 VM using this configurations and have been running a UAC and a UAS using SIPp on two other machines (UAC <--> Kamailio Proxy <--> UAS):
UAC
sipp -sn uac -nr -r 1 -rp 1000 -d 0 -l 1 -p 5060 -trace_msg -i SELF_IP(UAC) -rsa PROXY_IP:5060 UAS_IP:5060
UAS:
sipp -sn uas -d 0 -p 5060 -i SELF_IP(UAS) -rsa PRXY_IP:5060 -trace_msg
It looks that messages go through, and are received by the server, but what I get back at the UAC (client) is a "dead call" error:
Last Error: Dead call 1-1734@10.140.34.188 (aborted at index 8),
receive...
Is there a specific meaning to this "dead call" error? Is there anything that my proxy is missing in its routing or does it have to do with the UAC/UAS configs?
Thanks Alireza
On Tue, Jun 17, 2014 at 5:11 AM, Daniel-Constantin Mierla < miconda@gmail.com> wrote:
Hello,
listen is to specify local ip address or network interface on which kamailio should listen for sip traffic.
To send to an ip address there are couple of variants, in config file:
$ru = "sip:" + $rU + "@__NEXT_PROXY_IP__"; t_relay(); exit;
Or, if you don't want to change the r-uri, then use:
$du = "sip:__NEXT_PROXY_IP__"; t_relay(); exit;
Of course, you have to replace the __NEXT_PROXY_IP__ with the appropriate value.
More dynamic option would be using dispatcher module.
Cheers, Daniel
On 15/06/14 15:28, AliReza Khoshgoftar Monfared wrote:
Dear Kamailio users,
I am trying to set up a simple scenario as follows:
UAC --> Proxy_1 --> Proxy_2 --> Proxy_3 --> UAC
I am new to Kamailio and had the following basic questions that came to my mind after reading the documentation an the default kamilio.cfg config script:
- Is there any good example of the above scenario in which the UAC and
UAS (caller and callee) are modeled using SIPp and proxies are simple kamailio instances run on different machines that simply forward the SIP packets?
- In specific, I want the proxies to simply forward the messages to
their downstream (I do not care about registration or other operations and look for a minimally working simple SIP scenario). Is there any example of such configuration?
- Let's imagine Proxy_2 in the above example, in the default config,
it looks like that with modifying line 164, "listen=udp:10.0.0.10:5060" I can specify the address of its upstream, Proxy_1, but how and where shall I set the specifications of the downstream, Proxy_3?
I suspect it is somewhere in the Routing Logic block (line 449) but not sure how it is exactly done. I see a "route(SIPOUT)" call for example but I am not sure how and where the value of SIPOUT is modified. Is it a representation of the downstream servers in the config file?
Thanks, Alireza
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hi,
I have another update. The problem that I had is fixed now and it is fixed in an unexpected way!
The fix seems to be in changing the SIPp default client scenario. In specific, once I change the UAC.xml in SIPp trunk to generate the "BYE" message's "To" field like this:
To: sut sip:[call_number]@[remote_ip]:[remote_port]
instead of:
To: sut sip:[call_number]@[remote_ip]:[remote_port][peer_tag_param]
The problem is solved and Kamailio proxy forwards BYEs to the server. I am not sure why this is happening and if it more of a SIPp issue or a Kamailio issue. Neither I am sure of the meaning of the "[peer_tag_param]" in SIPp default client, but I thought it might be helpful to document the issue here in case that somebody else was searching the same issue.
On Thu, Jun 19, 2014 at 8:42 PM, AliReza Khoshgoftar Monfared < khoshgoftar@gmail.com> wrote:
Shedding some more light on the situation, here are some further facts I could figure out:
- I did the same test without the proxy in between (UAC <--> UAS) purely
in SIPp, everything was fine, no dead call error
- With the Kamailio proxy in between, I can see that there is a problem
in call termination. The client is not receiving any "200 OK" for the "BYE"s it is sending.
Is there anything wrong in my routing logic that is preventing these ACKs?
On Thu, Jun 19, 2014 at 4:09 PM, AliReza Khoshgoftar Monfared < khoshgoftar@gmail.com> wrote:
Thanks very much Daniel,
This is solving a part of my issue for now. Here is the routing section of my configuration file (it may be good to share it once complete as a minimally working proxy):
route{
if (!mf_process_maxfwd_header("10")) { sl_reply("483","Too Many Hops"); break; } if (msg:len >= 4096 ) { sl_reply("513", "Message too big");}if (!method=="REGISTER") record_route();if (!loose_route()) {$du = "sip:10.236.214.86:5060"; #setdsturi("sip:10.236.214.86:5060"); #second way to doit if (!t_relay()) { sl_reply_error(); }
}}
I am trying to set up a kamailio proxy on an EC2 VM using this configurations and have been running a UAC and a UAS using SIPp on two other machines (UAC <--> Kamailio Proxy <--> UAS):
UAC
sipp -sn uac -nr -r 1 -rp 1000 -d 0 -l 1 -p 5060 -trace_msg -i SELF_IP(UAC) -rsa PROXY_IP:5060 UAS_IP:5060
UAS:
sipp -sn uas -d 0 -p 5060 -i SELF_IP(UAS) -rsa PRXY_IP:5060 -trace_msg
It looks that messages go through, and are received by the server, but what I get back at the UAC (client) is a "dead call" error:
Last Error: Dead call 1-1734@10.140.34.188 (aborted at index 8),
receive...
Is there a specific meaning to this "dead call" error? Is there anything that my proxy is missing in its routing or does it have to do with the UAC/UAS configs?
Thanks Alireza
On Tue, Jun 17, 2014 at 5:11 AM, Daniel-Constantin Mierla < miconda@gmail.com> wrote:
Hello,
listen is to specify local ip address or network interface on which kamailio should listen for sip traffic.
To send to an ip address there are couple of variants, in config file:
$ru = "sip:" + $rU + "@__NEXT_PROXY_IP__"; t_relay(); exit;
Or, if you don't want to change the r-uri, then use:
$du = "sip:__NEXT_PROXY_IP__"; t_relay(); exit;
Of course, you have to replace the __NEXT_PROXY_IP__ with the appropriate value.
More dynamic option would be using dispatcher module.
Cheers, Daniel
On 15/06/14 15:28, AliReza Khoshgoftar Monfared wrote:
Dear Kamailio users,
I am trying to set up a simple scenario as follows:
UAC --> Proxy_1 --> Proxy_2 --> Proxy_3 --> UAC
I am new to Kamailio and had the following basic questions that came to my mind after reading the documentation an the default kamilio.cfg config script:
- Is there any good example of the above scenario in which the UAC
and UAS (caller and callee) are modeled using SIPp and proxies are simple kamailio instances run on different machines that simply forward the SIP packets?
- In specific, I want the proxies to simply forward the messages to
their downstream (I do not care about registration or other operations and look for a minimally working simple SIP scenario). Is there any example of such configuration?
- Let's imagine Proxy_2 in the above example, in the default config,
it looks like that with modifying line 164, "listen=udp:10.0.0.10:5060" I can specify the address of its upstream, Proxy_1, but how and where shall I set the specifications of the downstream, Proxy_3?
I suspect it is somewhere in the Routing Logic block (line 449) but not sure how it is exactly done. I see a "route(SIPOUT)" call for example but I am not sure how and where the value of SIPOUT is modified. Is it a representation of the downstream servers in the config file?
Thanks, Alireza
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-
AliReza Khoshgoftar Monfared -
Daniel-Constantin Mierla