13 Mar
2015
13 Mar
'15
3:52 a.m.
Hi list,
I'm new here, also to Kamailio. My goal is to achieve the "standard
solution": Kamailio for load balancing and high availability of several
Asterisk servers.
For starters, I've set it up with Kamailio+dispatcher+permissions acting
as a proxy and it was working fine, but I'd like to try to keep it much
more simple. For these reasons:
1) I'd like to maintain different peer settings in Asterisk (DTMF, codecs).
2) I don't want/need a database for Kamailio. I think it is mandatory
for permissions module + IP auth. So, let's do Asterisk the
authentication, it's already there.
3) Accounting. I'm using a software where it's really useful for
accounting purposes if every peer/user has its own entry in Asterisk's
sip.conf / in realtime table.
Long story short, my config now looks like this and it seems to be
working fine for me:
request_route {
route(DISPATCHER);
}
route[DISPATCHER] {
if ( method=="REGISTER") {
ds_select_domain("1","4");
sl_send_reply("300","Redirect");
exit;
}
if ( method=="INVITE" ) {
ds_select_domain("1","4");
sl_send_reply("300","Redirect");
exit;
}
if ( method=="SUBSCRIBE" ) {
ds_select_domain("1","4");
sl_send_reply("300","Redirect");
exit;
}
}
My questions:
1) There are other approaches to the DTMF+codecs issue. Like this thread
from 2013:
http://lists.kamailio.org/pipermail/sr-users/2013-January/076379.html -
but I have dozens of changing PSTN peers, so authentication/accounting
based on port would be cumbersome. Are there new/alternative ways?
2) I searched the archive and the Web but did not find many questions
about a "I-don't-want-Kamailio-to-show-as-the-only-peer-to-Asterisk"
solution. Therefore I'm wondering: Is everyone a) running such
homogeneous setups that they can survive with the same codec + DTMF
settings in Asterisk for all their peers + users? Or b) doing some
rewriting magic within Kamailio to deal with it? Or c) I'm overlooking a
simple solution. Which one is it? :)
3) And the biggest question: Everywhere I read basically "Redirect
server for LB + HA is not recommended, better use proxy". Is this still
true, and if so, what are the reasons? Because I should be worried that
not all clients/gateways out there respect a SIP 300 Redirect? Other
reasons?
Thanks so much! :)
Markus
13 Mar
13 Mar
4:47 a.m.
Hello,
On 13/03/15 08:52, Markus wrote:
Hi list,
I'm new here, also to Kamailio. My goal is to achieve the "standard
solution": Kamailio for load balancing and high availability of
several Asterisk servers.
For starters, I've set it up with Kamailio+dispatcher+permissions
acting as a proxy and it was working fine, but I'd like to try to keep
it much more simple. For these reasons:
1) I'd like to maintain different peer settings in Asterisk (DTMF,
codecs).
2) I don't want/need a database for Kamailio. I think it is mandatory
for permissions module + IP auth. So, let's do Asterisk the
authentication, it's already there.
You can use here db_text or db_sqlite, which don't require a heavy mysql
server. You can also have static rules in kamailio, like:
if(src_ip==asterisk_ip1 || src_ip==asterisk_ip1 || ...)
Or, if asterisk servers are in the same subnet, you can have:
if(src_ip==asterisk_net/24)
As you use dispatcher, another alternative is to use ds_is_from_list()
function, which will match if the request was sent by one of addresses
in the dispatcher lists.
3) Accounting. I'm using a software where it's really useful for
accounting purposes if every peer/user has its own entry in Asterisk's
sip.conf / in realtime table.
Long story short, my config now looks like this and it seems to be
working fine for me:
request_route {
route(DISPATCHER);
}
route[DISPATCHER] {
if ( method=="REGISTER") {
ds_select_domain("1","4");
sl_send_reply("300","Redirect");
exit;
}
if ( method=="INVITE" ) {
ds_select_domain("1","4");
sl_send_reply("300","Redirect");
exit;
}
if ( method=="SUBSCRIBE" ) {
ds_select_domain("1","4");
sl_send_reply("300","Redirect");
exit;
}
}
My questions:
1) There are other approaches to the DTMF+codecs issue. Like this
thread from 2013:
http://lists.kamailio.org/pipermail/sr-users/2013-January/076379.html
- but I have dozens of changing PSTN peers, so
authentication/accounting based on port would be cumbersome. Are there
new/alternative ways?
2) I searched the archive and the Web but did not find many questions
about a "I-don't-want-Kamailio-to-show-as-the-only-peer-to-Asterisk"
solution. Therefore I'm wondering: Is everyone a) running such
homogeneous setups that they can survive with the same codec + DTMF
settings in Asterisk for all their peers + users? Or b) doing some
rewriting magic within Kamailio to deal with it? Or c) I'm overlooking
a simple solution. Which one is it? :)
Can asterisk make some decision on headers? IIRC, someone was mentioning
at some point to use Remote-Party-ID (could be now P-Asserted-Identity)
header to match on a specific profile.
3) And the biggest question: Everywhere I read basically "Redirect
server for LB + HA is not recommended, better use proxy". Is this
still true, and if so, what are the reasons? Because I should be
worried that not all clients/gateways out there respect a SIP 300
Redirect? Other reasons?
I think this is something specific for your environment.
Kamailio
doesn't have any issues sending redirect replies. A proxy might be
required in case of natted endpoints when they cannot communicate each
other. But otherwise, if you don't want the new INVITE to pass again via
LB, then you can send 3xx.
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, May 27-29, 2015
Berlin, Germany - http://www.kamailioworld.com
11:39 a.m.
Hi Daniel,
Am 13.03.2015 um 09:47 schrieb Daniel-Constantin Mierla:
Can asterisk make some decision on headers? IIRC,
someone was mentioning
at some point to use Remote-Party-ID (could be now P-Asserted-Identity)
header to match on a specific profile.
I don't know. I'll ask in the Asterisk list. But based on your answer it
really seems this is a scenario that not many people need or ask for.
I'd love to find out why this is the case. I would have estimated 9 out
of 10 people need to fine-tune DTMF + codec settings in Asterisk.
I think this is something specific for your
environment. Kamailio
doesn't have any issues sending redirect replies. A proxy might be
required in case of natted endpoints when they cannot communicate each
other. But otherwise, if you don't want the new INVITE to pass again via
LB, then you can send 3xx.
Other than communication between natted endpoints you don't see any
disadvantage of using Kamailio for LB+HA as Redirect server instead as
proxy server?
Thanks!
Markus
12:14 p.m.
On 13 Mar 2015, at 16:39, Markus <universe(a)truemetal.org> wrote:
Hi Daniel,
Am 13.03.2015 um 09:47 schrieb Daniel-Constantin Mierla:
The most common solution
I've seen is to have kamailio use 10 different ports and have 10 different matching
peers in the Asterisk SIP config to select settings.
/O
Can asterisk make some decision on headers? IIRC,
someone was mentioning
at some point to use Remote-Party-ID (could be now P-Asserted-Identity)
header to match on a specific profile.
I don't know. I'll ask in the Asterisk list. But based on your answer it really
seems this is a scenario that not many people need or ask for.
I'd love to find out why this is the case. I would have estimated 9 out of 10 people
need to fine-tune DTMF + codec settings in Asterisk. 
1:45 p.m.
On Friday 13 March 2015 17:14:31 Olle E. Johansson wrote:
Cumbersome, why not just use some custom headers (stored in usr_preferences)?
kamailio stuff:
if($avp(src_dtmfmode)!="")
{
append_hf("X-DTMFMode: $avp(src_dtmfmode)\r\n");
}
asterisk stuff:
exten => _[+X].,n,ExecIf($[${LEN(${SIP_HEADER(X-DTMFMode)})} >
0]?SIPDtmfMode(${SIP_HEADER(X-DTMFMode)}))
The other way around is bit more tricky. YOu have to dial using the M option
(i'm using AGI instead of RT):
M(setdtmftx^".$row['dtmftx'].")
[macro-setdtmftx]
exten => s,1,NoOp()
exten => s,n,SIPDtmfMode(${ARG1})
exten => s,n,MacroExit
--
Telefoon: 088 0100 700
Sales: sales(a)pocos.nl | Service: servicedesk(a)pocos.nl
http://www.pocos.nl/ | Croy 9c, 5653 LC Eindhoven | Kamer van Koophandel
17097024
I'd love
to find out why this is the case. I would have estimated 9 out of
10 people need to fine-tune DTMF + codec settings in Asterisk.
The most common
solution I've seen is to have kamailio use 10 different
ports and have 10 different matching peers in the Asterisk SIP config to
select settings.
3:16 p.m.
On 13 Mar 2015, at 18:45, Daniel Tryba <d.tryba(a)pocos.nl> wrote:
On Friday 13 March 2015 17:14:31 Olle E. Johansson
wrote:
Cumbersome, why not just use some custom headers (stored in usr_preferences)?
There
are settings you can't set from the dialplan, but when you can this is a good
idea.
/O
I'd
love to find out why this is the case. I would have estimated 9 out of
10 people need to fine-tune DTMF + codec settings in Asterisk.
The most common
solution I've seen is to have kamailio use 10 different
ports and have 10 different matching peers in the Asterisk SIP config to
select settings.
kamailio stuff:
if($avp(src_dtmfmode)!="")
{
append_hf("X-DTMFMode: $avp(src_dtmfmode)\r\n");
}
asterisk stuff:
exten => _[+X].,n,ExecIf($[${LEN(${SIP_HEADER(X-DTMFMode)})} >
0]?SIPDtmfMode(${SIP_HEADER(X-DTMFMode)}))
The other way around is bit more tricky. YOu have to dial using the M option
(i'm using AGI instead of RT):
M(setdtmftx^".$row['dtmftx'].")
[macro-setdtmftx]
exten => s,1,NoOp()
exten => s,n,SIPDtmfMode(${ARG1})
exten => s,n,MacroExit
--
Telefoon: 088 0100 700
Sales: sales(a)pocos.nl | Service: servicedesk(a)pocos.nl
http://www.pocos.nl/ | Croy 9c, 5653 LC Eindhoven | Kamer van Koophandel
17097024
3:34 p.m.
On Fri, Mar 13, 2015 at 08:16:50PM +0100, Olle E. Johansson wrote:
With modern Asterisks is has been a long time since I needed anything
that wasn't available from the dialplan (can't think of anything).
Cumbersome,
why not just use some custom headers (stored in usr_preferences)?
There are
settings you can't set from the dialplan, but when you can this is a good
idea.
15 Mar
15 Mar
3:22 a.m.
On 13 Mar 2015, at 20:34, Daniel Tryba <d.tryba(a)pocos.nl> wrote:
On Fri, Mar 13, 2015 at 08:16:50PM +0100, Olle E.
Johansson wrote:
With modern Asterisks is has been a long time since I needed anything
that wasn't available from the dialplan (can't think of anything).
Good. Then we've done a good job with Asterisk. :-)
Thank you for the feedback.
/O
Cumbersome, why not just use some custom headers (stored in usr_preferences)?
There
are settings you can't set from the dialplan, but when you can this is a good
idea.
18 Mar
18 Mar
4:25 a.m.
On 13/03/15 16:39, Markus wrote:
[...]
accounting
for new calls (resulted from redirect) is no longer possible
on the sip server, but I guess that is not something you look for in
this scenario.
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, May 27-29, 2015
Berlin, Germany - http://www.kamailioworld.com
I think this is something specific for your
environment. Kamailio
doesn't have any issues sending redirect replies. A proxy might be
required in case of natted endpoints when they cannot communicate each
other. But otherwise, if you don't want the new INVITE to pass again via
LB, then you can send 3xx.
Other than communication between natted endpoints you don't see any
disadvantage of using Kamailio for LB+HA as Redirect server instead as
proxy server?
From Kamailio point of view, there is nothing. When doing redirect, the
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8 comments
4 participants
participants (4)
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Daniel Tryba -
Daniel-Constantin Mierla -
Markus -
Olle E. Johansson