12 Mar
2017
12 Mar
'17
3:47 p.m.
I would like to ask you how you deal with several asterisks and kamailio on
that same IP address, I have installation where i route 5060 to internal
server with kamailio and I would like to route RTP to asterisks, but is any
way to get around a problem with RTP ports collisions?
12 Mar
12 Mar
3:56 p.m.
Can you not assign different Asterisk instances different ranges of RTP ports to allocate
from?
On March 12, 2017 3:47:22 PM EDT, przeqpiciel <przeqpiciel(a)gmail.com> wrote:
I would like to ask you how you deal with several
asterisks and
kamailio on
that same IP address, I have installation where i route 5060 to
internal
server with kamailio and I would like to route RTP to asterisks, but is
any
way to get around a problem with RTP ports collisions?
-- Alex
--
Principal, Evariste Systems LLC (www.evaristesys.com)
Sent from my Google Nexus.
4:08 p.m.
Not entirely sure but you may be able to range them 0-0 for dynamic
allocation and let the kernel handle it.
On Sun, Mar 12, 2017 at 12:56 PM Alex Balashov <abalashov(a)evaristesys.com>
wrote:
Can you not assign different Asterisk instances
different ranges of RTP
ports to allocate from?
On March 12, 2017 3:47:22 PM EDT, przeqpiciel <przeqpiciel(a)gmail.com>
wrote:
--
Sent from Gmail Mobile
I would like to ask you how you deal with several
asterisks and
kamailio on
that same IP address, I have installation where i route 5060 to
internal
server with kamailio and I would like to route RTP to asterisks, but is
any
way to get around a problem with RTP ports collisions?
-- Alex
--
Principal, Evariste Systems LLC (www.evaristesys.com)
Sent from my Google Nexus.
_______________________________________________
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
5:48 p.m.
HI,
Thank you for your respond. Obviously, I could set a port range per
instance of Asterisk but I though about something more dynamic. But when I
think on it more deep there is no any other solution.
2017-03-12 21:08 GMT+01:00 Brandon Armstead <brandon(a)cryy.com>om>:
Not entirely sure but you may be able to range them
0-0 for dynamic
allocation and let the kernel handle it.
On Sun, Mar 12, 2017 at 12:56 PM Alex Balashov <abalashov(a)evaristesys.com>
wrote:
Can you not assign different Asterisk instances
different ranges of RTP
ports to allocate from?
On March 12, 2017 3:47:22 PM EDT, przeqpiciel <przeqpiciel(a)gmail.com>
wrote:
--
Sent from Gmail Mobile
I would like to ask you how you deal with several
asterisks and
kamailio on
that same IP address, I have installation where i route 5060 to
internal
server with kamailio and I would like to route RTP to asterisks, but is
any
way to get around a problem with RTP ports collisions?
-- Alex
--
Principal, Evariste Systems LLC (www.evaristesys.com)
Sent from my Google Nexus.
_______________________________________________
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
7:02 p.m.
On Sun, Mar 12, 2017 at 10:48:37PM +0100, przeqpiciel wrote:
Thank you for your respond. Obviously, I could set a
port range per
instance of Asterisk but I though about something more dynamic. But when I
think on it more deep there is no any other solution.
Trying to run concurrent instances of something like Asterisk on a
single host is very challenging for this, among other reasons.
I recommend containerisation of some sort, e.g. OpenVZ. Give it a very
lightweight "host" environment of its own, with distinct networking.
Then the full range of ports will be available to every instance.
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
7:28 p.m.
We face similar issues, we had 1 * server per VLAN, where each * had the
same IP address, we achieve this with NAT.
<----> Asterisk 1
<----> Asterisk 2
SIP LB <---> NAT <----> Asterisk 3
Your NAT device needs to be SIP aware, in our case we use Cisco ASA
appliance device/ I know there is a ASA virtual device now which runs in
VMware.
Hope it helps
On Sun, Mar 12, 2017 at 4:02 PM, Alex Balashov <abalashov(a)evaristesys.com>
wrote:
On Sun, Mar 12, 2017 at 10:48:37PM +0100, przeqpiciel
wrote:
Thank you for your respond. Obviously, I could
set a port range per
instance of Asterisk but I though about something more dynamic. But when
I
think on it more deep there is no any other
solution.
Trying to run concurrent instances of something like Asterisk on a
single host is very challenging for this, among other reasons.
I recommend containerisation of some sort, e.g. OpenVZ. Give it a very
lightweight "host" environment of its own, with distinct networking.
Then the full range of ports will be available to every instance.
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
_______________________________________________
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
13 Mar
13 Mar
2:08 a.m.
We run multiple Asterisk instances since 1.4 and never configured RTP
ports.
More challenging issues are the Asterisk DB, and the Asteisk home.
Med venlig hilsen / Best regards
Kjeld Flarup (Christensen) M.Sc E.E, Teknisk chef
Viptel ApS, Hammershusvej 16C, DK-7400 Herning
Telefon: +45 46949949, Telefax: +45 46949950, http://viptel.dk
On 03/13/2017 12:02 AM, Alex Balashov wrote:
On Sun, Mar 12, 2017 at 10:48:37PM +0100, przeqpiciel
wrote:
Thank you for your respond. Obviously, I could
set a port range per
instance of Asterisk but I though about something more dynamic. But when I
think on it more deep there is no any other solution.
Trying to run concurrent
instances of something like Asterisk on a
single host is very challenging for this, among other reasons.
I recommend containerisation of some sort, e.g. OpenVZ. Give it a very
lightweight "host" environment of its own, with distinct networking.
Then the full range of ports will be available to every instance.
2:15 a.m.
On Mon, Mar 13, 2017 at 07:08:09AM +0100, Kjeld Flarup wrote:
We run multiple Asterisk instances since 1.4 and never
configured RTP ports.
More challenging issues are the Asterisk DB, and the Asteisk home.
You may not have enough calls for RTP port collisions to become an
issue. Otherwise, I'm not sure how you're avoiding it, since Asterisk
isn't aware of which ports from within the range are in use.
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
8:52 a.m.
Maybe there is an magic device? I know that if we have an asterisk, that
become to us with default configuration of rtp ports sets to 10000_20000.
And each call choose the one port fron that range. So if we have several
asterisks with default configuratiin of rtp, there is possibilities to have
2 concurent calls each through another asterisk instance with this same rtp
port. Am i right?
So mqybe this magic device could see source IP address and route rtp to
correct adterisk?
13.03.2017 7:15 AM "Alex Balashov" <abalashov(a)evaristesys.com>
napisał(a):
On Mon, Mar 13, 2017 at 07:08:09AM +0100, Kjeld Flarup
wrote:
We run multiple Asterisk instances since 1.4 and
never configured RTP
ports.
More challenging issues are the Asterisk DB, and the Asteisk home.
You may not have enough calls for RTP port collisions to become an
issue. Otherwise, I'm not sure how you're avoiding it, since Asterisk
isn't aware of which ports from within the range are in use.
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
_______________________________________________
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
6:04 p.m.
No there is no such thing as magic.
The most obvious way to implement the RTP port handling, is to first
open the next UDP port in the OS, and then report that back in the
Invite/200Ok. If the port cannot be opened, then simply try the next in
line.
Med venlig hilsen / Best regards
Kjeld Flarup (Christensen) M.Sc E.E, Teknisk chef
Viptel ApS, Hammershusvej 16C, DK-7400 Herning
Telefon: +45 46949949, Telefax: +45 46949950, http://viptel.dk
On 03/13/2017 01:52 PM, przeqpiciel wrote:
Maybe there is an magic device? I know that if we have
an asterisk,
that become to us with default configuration of rtp ports sets to
10000_20000. And each call choose the one port fron that range. So if
we have several asterisks with default configuratiin of rtp, there is
possibilities to have 2 concurent calls each through another asterisk
instance with this same rtp port. Am i right?
So mqybe this magic device could see source IP address and route rtp
to correct adterisk?
13.03.2017 7:15 AM "Alex Balashov" <abalashov(a)evaristesys.com
<mailto:abalashov@evaristesys.com>> napisał(a):
On Mon, Mar 13, 2017 at 07:08:09AM +0100, Kjeld Flarup wrote:
We run multiple Asterisk instances since 1.4 and
never
configured RTP ports.
More challenging issues are the Asterisk DB, and the Asteisk home.
You may not have enough calls for RTP port collisions to become an
issue. Otherwise, I'm not sure how you're avoiding it, since Asterisk
isn't aware of which ports from within the range are in use.
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 <tel:%2B1-706-510-6800> / +1-800-250-5920
<tel:%2B1-800-250-5920> (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
list
sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
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6:05 p.m.
Well, indeed, but a sequential scan of many consecutive ports like this from the bottom of
the same range can be quite a latent operation. So at the very least the allocation
strategy would benefit from being random. Does Asterisk take that approach?
On March 13, 2017 6:04:06 PM EDT, Kjeld Flarup <kfc(a)viptel.dk> wrote:
No there is no such thing as magic.
The most obvious way to implement the RTP port handling, is to first
open the next UDP port in the OS, and then report that back in the
Invite/200Ok. If the port cannot be opened, then simply try the next in
line.
Med venlig hilsen / Best regards
Kjeld Flarup (Christensen) M.Sc E.E, Teknisk chef
Viptel ApS, Hammershusvej 16C, DK-7400 Herning
Telefon: +45 46949949, Telefax: +45 46949950, http://viptel.dk
On 03/13/2017 01:52 PM, przeqpiciel wrote:
-- Alex
--
Principal, Evariste Systems LLC (www.evaristesys.com)
Sent from my Google Nexus.
Maybe there is an magic device? I know that if we
have an asterisk,
that become to us with default configuration of rtp ports sets to
10000_20000. And each call choose the one port fron that range. So if
we have several asterisks with default
configuratiin of rtp, there is
possibilities to have 2 concurent calls each
through another asterisk
instance with this same rtp port. Am i right?
So mqybe this magic device could see source IP address and route rtp
to correct adterisk?
13.03.2017 7:15 AM "Alex Balashov" <abalashov(a)evaristesys.com
<mailto:abalashov@evaristesys.com>> napisał(a):
On Mon, Mar 13, 2017 at 07:08:09AM +0100, Kjeld Flarup wrote:
home.
We run multiple Asterisk instances since 1.4 and
never
configured RTP ports.
>
> More challenging issues are the Asterisk DB, and the Asteisk
You may not have enough calls for RTP port collisions to become
an
issue. Otherwise, I'm not sure how
you're avoiding it, since
Asterisk
isn't aware of which ports from within
the range are in use.
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 <tel:%2B1-706-510-6800> / +1-800-250-5920
<tel:%2B1-800-250-5920> (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
mailing
list
sr-users(a)lists.sip-router.org
<mailto:sr-users@lists.sip-router.org>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>
_______________________________________________
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
6:21 p.m.
As I recall it is sequential, but not from the start everytime, it is
incrementing all the time.
If You are running three servers, then with a 100% identical load, one
would expect an average of 2 failing attempts per call.
The reality I see is however often very different RTP ports, most likely
because load isn't 100% identical.
Med venlig hilsen / Best regards
Kjeld Flarup (Christensen) M.Sc E.E, Teknisk chef
Viptel ApS, Hammershusvej 16C, DK-7400 Herning
Telefon: +45 46949949, Telefax: +45 46949950, http://viptel.dk
On 03/13/2017 11:05 PM, Alex Balashov wrote:
Well, indeed, but a sequential scan of many
consecutive ports like this from the bottom of the same range can be quite a latent
operation. So at the very least the allocation strategy would benefit from being random.
Does Asterisk take that approach?
On March 13, 2017 6:04:06 PM EDT, Kjeld Flarup <kfc(a)viptel.dk> wrote:
No there is no such thing as magic.
The most obvious way to implement the RTP port handling, is to first
open the next UDP port in the OS, and then report that back in the
Invite/200Ok. If the port cannot be opened, then simply try the next in
line.
Med venlig hilsen / Best regards
Kjeld Flarup (Christensen) M.Sc E.E, Teknisk chef
Viptel ApS, Hammershusvej 16C, DK-7400 Herning
Telefon: +45 46949949, Telefax: +45 46949950, http://viptel.dk
On 03/13/2017 01:52 PM, przeqpiciel wrote:
-- Alex
--
Principal, Evariste Systems LLC (www.evaristesys.com)
Sent from my Google Nexus.
_______________________________________________
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 Maybe there is an magic device? I know that if we
have an asterisk,
that become to us with default configuration of rtp ports sets to
10000_20000. And each call choose the one port fron that range. So if
we have several asterisks with default configuratiin of rtp, there is
possibilities to have 2 concurent calls each through another asterisk
instance with this same rtp port. Am i right?
So mqybe this magic device could see source IP address and route rtp
to correct adterisk?
13.03.2017 7:15 AM "Alex Balashov" <abalashov(a)evaristesys.com
<mailto:abalashov@evaristesys.com>> napisał(a):
On Mon, Mar 13, 2017 at 07:08:09AM +0100, Kjeld Flarup wrote:
home.
We run multiple Asterisk instances since 1.4 and
never
configured RTP ports.
>
> More challenging issues are the Asterisk DB, and the Asteisk You may not have enough calls for RTP port
collisions to become
an
issue. Otherwise, I'm not sure how
you're avoiding it, since
Asterisk
isn't aware of which ports from within
the range are in use.
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 <tel:%2B1-706-510-6800> / +1-800-250-5920
<tel:%2B1-800-250-5920> (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
mailing
list
sr-users(a)lists.sip-router.org
<mailto:sr-users@lists.sip-router.org>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>
_______________________________________________
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 
14 Mar
14 Mar
12:14 a.m.
WHy not installing rtpproxy and proxying all rtp to the inside uase
kamailio to load balance them, it will be transparent on the inside perhaps
a cleaner solution?
On Mon, Mar 13, 2017 at 3:21 PM, Kjeld Flarup <kfc(a)viptel.dk> wrote:
As I recall it is sequential, but not from the start
everytime, it is
incrementing all the time.
If You are running three servers, then with a 100% identical load, one
would expect an average of 2 failing attempts per call.
The reality I see is however often very different RTP ports, most likely
because load isn't 100% identical.
Med venlig hilsen / Best regards
Kjeld Flarup (Christensen) M.Sc E.E, Teknisk chef
Viptel ApS, Hammershusvej 16C, DK-7400 Herning
Telefon: +45 46949949, Telefax: +45 46949950, http://viptel.dk
On 03/13/2017 11:05 PM, Alex Balashov wrote:
Well, indeed, but a sequential scan of many
consecutive ports like this
from the bottom of the same range can be quite a latent operation. So at
the very least the allocation strategy would benefit from being random.
Does Asterisk take that approach?
On March 13, 2017 6:04:06 PM EDT, Kjeld Flarup <kfc(a)viptel.dk> wrote:
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
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http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
No there is no such thing as magic.
The most obvious way to implement the RTP port handling, is to first
open the next UDP port in the OS, and then report that back in the
Invite/200Ok. If the port cannot be opened, then simply try the next in
line.
Med venlig hilsen / Best regards
Kjeld Flarup (Christensen) M.Sc E.E, Teknisk chef
Viptel ApS, Hammershusvej 16C, DK-7400 Herning
Telefon: +45 46949949, Telefax: +45 46949950, http://viptel.dk
On 03/13/2017 01:52 PM, przeqpiciel wrote:
-- Alex
--
Principal, Evariste Systems LLC (www.evaristesys.com)
Sent from my Google Nexus.
_______________________________________________
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
Maybe there is an magic device? I know that if we
have an asterisk,
that become to us with default configuration of rtp ports sets to
10000_20000. And each call choose the one port fron that range. So if
we have several asterisks with default configuratiin of rtp, there is
possibilities to have 2 concurent calls each through another asterisk
instance with this same rtp port. Am i right?
So mqybe this magic device could see source IP address and route rtp
to correct adterisk?
13.03.2017 7:15 AM "Alex Balashov" <abalashov(a)evaristesys.com
<mailto:abalashov@evaristesys.com>> napisał(a):
On Mon, Mar 13, 2017 at 07:08:09AM +0100, Kjeld Flarup wrote:
> We run multiple Asterisk instances since 1.4 and never
configured RTP ports.
>
> More challenging issues are the Asterisk DB, and the Asteisk
home.
You may not have enough calls for RTP port
collisions to become
an
issue. Otherwise, I'm not sure how
you're avoiding it, since
Asterisk
isn't aware of which ports from within
the range are in use.
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 <tel:%2B1-706-510-6800> / +1-800-250-5920
<tel:%2B1-800-250-5920> (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
mailing
list
sr-users(a)lists.sip-router.org
<mailto:sr-users@lists.sip-router.org>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>
_______________________________________________
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list
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2:43 a.m.
> WHy not installing rtpproxy and proxying all
Because I would like to record some calls and I dont know RTPProxy's
features, maybe it could record ?
2017-03-14 5:14 GMT+01:00 anfecora <anfecora(a)gmail.com>om>:
WHy not installing rtpproxy and proxying all rtp to
the inside uase
kamailio to load balance them, it will be transparent on the inside perhaps
a cleaner solution?
On Mon, Mar 13, 2017 at 3:21 PM, Kjeld Flarup <kfc(a)viptel.dk> wrote:
As I recall it is sequential, but not from the
start everytime, it is
incrementing all the time.
If You are running three servers, then with a 100% identical load, one
would expect an average of 2 failing attempts per call.
The reality I see is however often very different RTP ports, most likely
because load isn't 100% identical.
Med venlig hilsen / Best regards
Kjeld Flarup (Christensen) M.Sc E.E, Teknisk chef
Viptel ApS, Hammershusvej 16C, DK-7400 Herning
Telefon: +45 46949949, Telefax: +45 46949950, http://viptel.dk
On 03/13/2017 11:05 PM, Alex Balashov wrote:
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
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http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Well, indeed, but a sequential scan of many
consecutive ports like this
from the bottom of the same range can be quite a latent operation. So at
the very least the allocation strategy would benefit from being random.
Does Asterisk take that approach?
On March 13, 2017 6:04:06 PM EDT, Kjeld Flarup <kfc(a)viptel.dk> wrote:
_______________________________________________
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
No there is no such thing as magic.
The most obvious way to implement the RTP port handling, is to first
open the next UDP port in the OS, and then report that back in the
Invite/200Ok. If the port cannot be opened, then simply try the next in
line.
Med venlig hilsen / Best regards
Kjeld Flarup (Christensen) M.Sc E.E, Teknisk chef
Viptel ApS, Hammershusvej 16C, DK-7400 Herning
Telefon: +45 46949949, Telefax: +45 46949950, http://viptel.dk
On 03/13/2017 01:52 PM, przeqpiciel wrote:
> Maybe there is an magic device? I know that if we have an asterisk,
> that become to us with default configuration of rtp ports sets to
> 10000_20000. And each call choose the one port fron that range. So if
> we have several asterisks with default configuratiin of rtp, there is
> possibilities to have 2 concurent calls each through another asterisk
> instance with this same rtp port. Am i right?
>
> So mqybe this magic device could see source IP address and route rtp
> to correct adterisk?
>
> 13.03.2017 7:15 AM "Alex Balashov" <abalashov(a)evaristesys.com
> <mailto:abalashov@evaristesys.com>> napisał(a):
>
> On Mon, Mar 13, 2017 at 07:08:09AM +0100, Kjeld Flarup wrote:
>
> > We run multiple Asterisk instances since 1.4 and never
> configured RTP ports.
> >
> > More challenging issues are the Asterisk DB, and the Asteisk
>
home.
> You may not have enough calls for RTP port collisions to become
>
an
> issue. Otherwise, I'm not sure how you're avoiding it, since
>
Asterisk
> isn't aware of which ports from within the range are in use.
>
> --
> Alex Balashov | Principal | Evariste Systems LLC
>
> Tel: +1-706-510-6800 <tel:%2B1-706-510-6800> / +1-800-250-5920
> <tel:%2B1-800-250-5920> (toll-free)
> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
>
mailing
> list
> sr-users(a)lists.sip-router.org
>
<mailto:sr-users@lists.sip-router.org>
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>
>
>
>
> _______________________________________________
> 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
>
-- Alex
--
Principal, Evariste Systems LLC (www.evaristesys.com)
Sent from my Google Nexus.
_______________________________________________
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
2:44 a.m.
It can record, as can a number of other media relays.
On March 14, 2017 2:43:15 AM EDT, przeqpiciel <przeqpiciel(a)gmail.com> wrote:
list
> sr-users(a)lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
-- Alex
--
Principal, Evariste Systems LLC (www.evaristesys.com)
Sent from my Google Nexus.
> WHy not
installing rtpproxy and proxying all
Because I would like to record some calls and I
dont know RTPProxy's
features, maybe it could record ?
2017-03-14 5:14 GMT+01:00 anfecora <anfecora(a)gmail.com>om>:
WHy not installing rtpproxy and proxying all rtp
to the inside uase
kamailio to load balance them, it will be transparent on the inside
perhaps
a cleaner solution?
On Mon, Mar 13, 2017 at 3:21 PM, Kjeld Flarup <kfc(a)viptel.dk> wrote:
> As I recall it is sequential, but not from the start everytime, it
is
> incrementing all the time.
>
> If You are running three servers, then with a 100% identical load,
one
> would expect an average of 2 failing attempts
per call.
>
> The reality I see is however often very different RTP ports, most
likely
> because load isn't 100% identical.
>
>
> Med venlig hilsen / Best regards
> Kjeld Flarup (Christensen) M.Sc E.E, Teknisk chef
> Viptel ApS, Hammershusvej 16C, DK-7400 Herning
> Telefon: +45 46949949, Telefax: +45 46949950, http://viptel.dk
>
> On 03/13/2017 11:05 PM, Alex Balashov wrote:
>
>> Well, indeed, but a sequential scan of many consecutive ports like
this
>> from the bottom of the same range can be
quite a latent operation.
So at
>> the very least the allocation strategy
would benefit from being
random.
>> Does Asterisk take that approach?
>>
>> On March 13, 2017 6:04:06 PM EDT, Kjeld Flarup <kfc(a)viptel.dk>
wrote:
>>
>>> No there is no such thing as magic.
>>>
>>> The most obvious way to implement the RTP port handling, is to
first
>>> open the next UDP port in the OS, and
then report that back in the
>>> Invite/200Ok. If the port cannot be opened, then simply try the
next
in
>>>
>>> line.
>>>
>>>
>>> Med venlig hilsen / Best regards
>>> Kjeld Flarup (Christensen) M.Sc E.E, Teknisk chef
>>> Viptel ApS, Hammershusvej 16C, DK-7400 Herning
>>> Telefon: +45 46949949, Telefax: +45 46949950, http://viptel.dk
>>>
>>> On 03/13/2017 01:52 PM, przeqpiciel wrote:
>>>
>>>> Maybe there is an magic device? I know that if we have an
asterisk,
>>>> that become to us with default
configuration of rtp ports sets to
>>>> 10000_20000. And each call choose the one port fron that range.
So
if
>>>> we have several asterisks with
default configuratiin of rtp,
there is
>>>> possibilities to have 2 concurent
calls each through another
asterisk
>>>> instance with this same rtp port.
Am i right?
>>>>
>>>> So mqybe this magic device could see source IP address and route
rtp
>>>> to correct adterisk?
>>>>
>>>> 13.03.2017 7:15 AM "Alex Balashov"
<abalashov(a)evaristesys.com
>>>> <mailto:abalashov@evaristesys.com>> napisał(a):
>>>>
>>>> On Mon, Mar 13, 2017 at 07:08:09AM +0100, Kjeld Flarup
wrote:
>>>>
>>>> > We run multiple Asterisk instances since 1.4 and never
>>>> configured RTP ports.
>>>> >
>>>> > More challenging issues are the Asterisk DB, and the
Asteisk
>>>>
>>> home.
>>>
>>>> You may not have enough calls for RTP port collisions to
become
>>>>
>>> an
>>>
>>>> issue. Otherwise, I'm not sure how you're avoiding it,
since
>>>>
>>> Asterisk
>>>
>>>> isn't aware of which ports from within the range are in use.
>>>>
>>>> --
>>>> Alex Balashov | Principal | Evariste Systems LLC
>>>>
>>>> Tel: +1-706-510-6800 <tel:%2B1-706-510-6800> /
+1-800-250-5920
>>>>
<tel:%2B1-800-250-5920> (toll-free)
>>>> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
>>>>
>>>> _______________________________________________
>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
>>>>
>>> mailing
>>>
>>>> list
>>>> sr-users(a)lists.sip-router.org
>>>>
>>> <mailto:sr-users@lists.sip-router.org>
>>>
>>>>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>
<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> 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
>>>>
>>>
>> -- Alex
>>
>> --
>> Principal, Evariste Systems LLC (www.evaristesys.com)
>>
>> Sent from my Google Nexus.
>>
>> _______________________________________________
>> 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
>>
>
>
> _______________________________________________
> 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
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
2:46 a.m.
So, I can use Kamailio as SBC/Load balancer/registrar, Asterisk as IVR and
application server, and rtpproxy as media relay and recorder ?
2017-03-14 7:44 GMT+01:00 Alex Balashov <abalashov(a)evaristesys.com>om>:
It can record, as can a number of other media relays.
On March 14, 2017 2:43:15 AM EDT, przeqpiciel <przeqpiciel(a)gmail.com>
wrote:
list
> sr-users(a)lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
-- Alex
--
Principal, Evariste Systems LLC (www.evaristesys.com)
Sent from my Google Nexus.
_______________________________________________
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
> WHy
not installing rtpproxy and proxying all
Because I would like to record some calls
and I dont know RTPProxy's
features, maybe it could record ?
2017-03-14 5:14 GMT+01:00 anfecora <anfecora(a)gmail.com>om>:
WHy not installing rtpproxy and proxying all rtp
to the inside uase
kamailio to load balance them, it will be transparent on the inside
perhaps
a cleaner solution?
On Mon, Mar 13, 2017 at 3:21 PM, Kjeld Flarup <kfc(a)viptel.dk> wrote:
> As I recall it is sequential, but not from the start everytime, it
is
> incrementing all the time.
>
> If You are running three servers, then with a 100% identical load,
one
> would expect an average of 2 failing attempts
per call.
>
> The reality I see is however often very different RTP ports, most
likely
> because load isn't 100% identical.
>
>
> Med venlig hilsen / Best regards
> Kjeld Flarup (Christensen) M.Sc E.E, Teknisk chef
> Viptel ApS, Hammershusvej 16C, DK-7400 Herning
> Telefon: +45 46949949, Telefax: +45 46949950, http://viptel.dk
>
> On 03/13/2017 11:05 PM, Alex Balashov wrote:
>
>> Well, indeed, but a sequential scan of many consecutive ports like
this
>> from the bottom of the same range can be
quite a latent operation.
So at
>> the very least the allocation strategy
would benefit from being
random.
>> Does Asterisk take that approach?
>>
>> On March 13, 2017 6:04:06 PM EDT, Kjeld Flarup <kfc(a)viptel.dk>
wrote:
>>
>>> No there is no such thing as magic.
>>>
>>> The most obvious way to implement the RTP port handling, is to
first
>>> open the next UDP port in the OS, and
then report that back in the
>>> Invite/200Ok. If the port cannot be opened, then simply try the
next
in
>>>
>>> line.
>>>
>>>
>>> Med venlig hilsen / Best regards
>>> Kjeld Flarup (Christensen) M.Sc E.E, Teknisk chef
>>> Viptel ApS, Hammershusvej 16C, DK-7400 Herning
>>> Telefon: +45 46949949, Telefax: +45 46949950, http://viptel.dk
>>>
>>> On 03/13/2017 01:52 PM, przeqpiciel wrote:
>>>
>>>> Maybe there is an magic device? I know that if we have an
asterisk,
>>>> that become to us with default
configuration of rtp ports sets to
>>>> 10000_20000. And each call choose the one port fron that range.
So
if
>>>> we have several asterisks with
default configuratiin of rtp,
there is
>>>> possibilities to have 2 concurent
calls each through another
asterisk
>>>> instance with this same rtp port.
Am i right?
>>>>
>>>> So mqybe this magic device could see source IP address and route
rtp
>>>> to correct adterisk?
>>>>
>>>> 13.03.2017 7:15 AM "Alex Balashov"
<abalashov(a)evaristesys.com
>>>> <mailto:abalashov@evaristesys.com>> napisał(a):
>>>>
>>>> On Mon, Mar 13, 2017 at 07:08:09AM +0100, Kjeld Flarup
wrote:
>>>>
>>>> > We run multiple Asterisk instances since 1.4 and never
>>>> configured RTP ports.
>>>> >
>>>> > More challenging issues are the Asterisk DB, and the
Asteisk
>>>>
>>> home.
>>>
>>>> You may not have enough calls for RTP port collisions to
become
>>>>
>>> an
>>>
>>>> issue. Otherwise, I'm not sure how you're avoiding it,
since
>>>>
>>> Asterisk
>>>
>>>> isn't aware of which ports from within the range are in use.
>>>>
>>>> --
>>>> Alex Balashov | Principal | Evariste Systems LLC
>>>>
>>>> Tel: +1-706-510-6800 <tel:%2B1-706-510-6800> /
+1-800-250-5920
>>>>
<tel:%2B1-800-250-5920> (toll-free)
>>>> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
>>>>
>>>> _______________________________________________
>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
>>>>
>>> mailing
>>>
>>>> list
>>>> sr-users(a)lists.sip-router.org
>>>>
>>> <mailto:sr-users@lists.sip-router.org>
>>>
>>>>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>
<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>
>>>>
>>>>
>>>>
>>>>>
_______________________________________________
>>>>> 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
>>>>
>>>>
>>> -- Alex
>>>
>>> --
>>> Principal, Evariste Systems LLC (www.evaristesys.com)
>>>
>>> Sent from my Google Nexus.
>>>
>>> _______________________________________________
>>> 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
>>
>
>
> _______________________________________________
> 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
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
2:47 a.m.
Yes, though of course you would have to correlate the calls (most likely by Call-ID) and
integrate all this.
On March 14, 2017 2:46:27 AM EDT, przeqpiciel <przeqpiciel(a)gmail.com> wrote:
So, I can use Kamailio as SBC/Load balancer/registrar,
Asterisk as IVR
and
application server, and rtpproxy as media relay and recorder ?
2017-03-14 7:44 GMT+01:00 Alex Balashov <abalashov(a)evaristesys.com>om>:
list
> sr-users(a)lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
-- Alex
--
Principal, Evariste Systems LLC (www.evaristesys.com)
Sent from my Google Nexus.
It can record, as can a number of other media
relays.
On March 14, 2017 2:43:15 AM EDT, przeqpiciel <przeqpiciel(a)gmail.com>
wrote:
>>> WHy not installing rtpproxy and proxying all
>Because I would like to record some calls and I dont know RTPProxy's
>features, maybe it could record ?
>
>2017-03-14 5:14 GMT+01:00 anfecora <anfecora(a)gmail.com>om>:
>
>> WHy not installing rtpproxy and proxying all rtp to the inside
uase
>> kamailio to load balance them, it will
be transparent on the
inside
>perhaps
>> a cleaner solution?
>>
>> On Mon, Mar 13, 2017 at 3:21 PM, Kjeld Flarup <kfc(a)viptel.dk>
wrote:
>>
>>> As I recall it is sequential, but not from the start everytime,
it
>is
>>> incrementing all the time.
>>>
>>> If You are running three servers, then with a 100% identical
load,
>one
>>> would expect an average of 2 failing attempts per call.
>>>
>>> The reality I see is however often very different RTP ports, most
>likely
>>> because load isn't 100% identical.
>>>
>>>
>>> Med venlig hilsen / Best regards
>>> Kjeld Flarup (Christensen) M.Sc E.E, Teknisk chef
>>> Viptel ApS, Hammershusvej 16C, DK-7400 Herning
>>> Telefon: +45 46949949, Telefax: +45 46949950, http://viptel.dk
>>>
>>> On 03/13/2017 11:05 PM, Alex Balashov wrote:
>>>
>>>> Well, indeed, but a sequential scan of many consecutive ports
like
>this
>>>> from the bottom of the same range can be quite a latent
operation.
>So at
>>>> the very least the allocation strategy would benefit from being
>random.
>>>> Does Asterisk take that approach?
>>>>
>>>> On March 13, 2017 6:04:06 PM EDT, Kjeld Flarup <kfc(a)viptel.dk>
>wrote:
>>>>
>>>>> No there is no such thing as magic.
>>>>>
>>>>> The most obvious way to implement the RTP port handling, is to
>first
>>>>> open the next UDP port in the OS, and then report that back in
the
>>>>> Invite/200Ok. If the port
cannot be opened, then simply try the
>next in
>>>>>
>>>>> line.
>>>>>
>>>>>
>>>>> Med venlig hilsen / Best regards
>>>>> Kjeld Flarup (Christensen) M.Sc E.E, Teknisk chef
>>>>> Viptel ApS, Hammershusvej 16C, DK-7400 Herning
>>>>> Telefon: +45 46949949, Telefax: +45 46949950, http://viptel.dk
>>>>>
>>>>> On 03/13/2017 01:52 PM, przeqpiciel wrote:
>>>>>
>>>>>> Maybe there is an magic device? I know that if we have an
>asterisk,
>>>>>> that become to us with default configuration of rtp ports sets
to
>>>>>> 10000_20000. And each
call choose the one port fron that
range.
>So if
>>>>>> we have several asterisks with default configuratiin of rtp,
>there is
>>>>>> possibilities to have 2 concurent calls each through another
>asterisk
>>>>>> instance with this same rtp port. Am i right?
>>>>>>
>>>>>> So mqybe this magic device could see source IP address and
route
>rtp
>>>>>> to correct adterisk?
>>>>>>
>>>>>> 13.03.2017 7:15 AM "Alex Balashov"
<abalashov(a)evaristesys.com
>>>>>> <mailto:abalashov@evaristesys.com>> napisał(a):
>>>>>>
>>>>>> On Mon, Mar 13, 2017 at 07:08:09AM +0100, Kjeld Flarup
>wrote:
>>>>>>
>>>>>> > We run multiple Asterisk instances since 1.4 and
never
>>>>>> configured RTP ports.
>>>>>> >
>>>>>> > More challenging issues are the Asterisk DB, and the
>Asteisk
>>>>>>
>>>>> home.
>>>>>
>>>>>> You may not have enough calls for RTP port collisions to
>become
>>>>>>
>>>>> an
>>>>>
>>>>>> issue. Otherwise, I'm not sure how you're avoiding
it,
since
>>>>>>
>>>>> Asterisk
>>>>>
>>>>>> isn't aware of which ports from within the range are in
use.
>>>>>>
>>>>>> --
>>>>>> Alex Balashov | Principal | Evariste Systems LLC
>>>>>>
>>>>>> Tel: +1-706-510-6800 <tel:%2B1-706-510-6800> /
>+1-800-250-5920
>>>>>> <tel:%2B1-800-250-5920> (toll-free)
>>>>>> Web: http://www.evaristesys.com/,
http://www.csrpswitch.com/
>>>>>>
>>>>>> _______________________________________________
>>>>>> SIP Express Router (SER) and Kamailio (OpenSER) -
sr-users
>>>>>>
>>>>> mailing
>>>>>
>>>>>> list
>>>>>> sr-users(a)lists.sip-router.org
>>>>>>
>>>>> <mailto:sr-users@lists.sip-router.org>
>>>>>
>>>>>>
>http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>>>
><http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>
>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> 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
>>>>>>
>>>>>
>>>> -- Alex
>>>>
>>>> --
>>>> Principal, Evariste Systems LLC (www.evaristesys.com)
>>>>
>>>> Sent from my Google Nexus.
>>>>
>>>> _______________________________________________
>>>> 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
>>>>
>>>
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
mailing
list
-- Alex
--
Principal, Evariste Systems LLC (www.evaristesys.com)
Sent from my Google Nexus.
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing > sr-users(a)lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
_______________________________________________
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
>
>
2:55 a.m.
I would like to create PBX platform, at now I faced to make drag&drop ivr
creator. After that I would create option for record calls for client and
this is why I look for solution :)
2017-03-14 7:47 GMT+01:00 Alex Balashov <abalashov(a)evaristesys.com>om>:
Yes, though of course you would have to correlate the
calls (most likely
by Call-ID) and integrate all this.
On March 14, 2017 2:46:27 AM EDT, przeqpiciel <przeqpiciel(a)gmail.com>
wrote:
So, I can use Kamailio as SBC/Load
balancer/registrar, Asterisk as IVR
and
application server, and rtpproxy as media relay and recorder ?
2017-03-14 7:44 GMT+01:00 Alex Balashov <abalashov(a)evaristesys.com>om>:
list
> sr-users(a)lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
-- Alex
--
Principal, Evariste Systems LLC (www.evaristesys.com)
Sent from my Google Nexus.
_______________________________________________
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
It can record, as can a number of other media
relays.
On March 14, 2017 2:43:15 AM EDT, przeqpiciel <przeqpiciel(a)gmail.com>
wrote:
>>> WHy not installing rtpproxy and proxying all
>Because I would like to record some calls and I dont know RTPProxy's
>features, maybe it could record ?
>
>2017-03-14 5:14 GMT+01:00 anfecora <anfecora(a)gmail.com>om>:
>
>> WHy not installing rtpproxy and proxying all rtp to the inside
uase
>> kamailio to load balance them, it will
be transparent on the
inside
>perhaps
>> a cleaner solution?
>>
>> On Mon, Mar 13, 2017 at 3:21 PM, Kjeld Flarup <kfc(a)viptel.dk>
wrote:
>>
>>> As I recall it is sequential, but not from the start everytime,
it
>is
>>> incrementing all the time.
>>>
>>> If You are running three servers, then with a 100% identical
load,
>one
>>> would expect an average of 2 failing attempts per call.
>>>
>>> The reality I see is however often very different RTP ports, most
>likely
>>> because load isn't 100% identical.
>>>
>>>
>>> Med venlig hilsen / Best regards
>>> Kjeld Flarup (Christensen) M.Sc E.E, Teknisk chef
>>> Viptel ApS, Hammershusvej 16C, DK-7400 Herning
>>> Telefon: +45 46949949, Telefax: +45 46949950, http://viptel.dk
>>>
>>> On 03/13/2017 11:05 PM, Alex Balashov wrote:
>>>
>>>> Well, indeed, but a sequential scan of many consecutive ports
like
>this
>>>> from the bottom of the same range can be quite a latent
operation.
>So at
>>>> the very least the allocation strategy would benefit from being
>random.
>>>> Does Asterisk take that approach?
>>>>
>>>> On March 13, 2017 6:04:06 PM EDT, Kjeld Flarup <kfc(a)viptel.dk>
>wrote:
>>>>
>>>>> No there is no such thing as magic.
>>>>>
>>>>> The most obvious way to implement the RTP port handling, is to
>first
>>>>> open the next UDP port in the OS, and then report that back in
the
>>>>> Invite/200Ok. If the port
cannot be opened, then simply try the
>next in
>>>>>
>>>>> line.
>>>>>
>>>>>
>>>>> Med venlig hilsen / Best regards
>>>>> Kjeld Flarup (Christensen) M.Sc E.E, Teknisk chef
>>>>> Viptel ApS, Hammershusvej 16C, DK-7400 Herning
>>>>> Telefon: +45 46949949, Telefax: +45 46949950, http://viptel.dk
>>>>>
>>>>> On 03/13/2017 01:52 PM, przeqpiciel wrote:
>>>>>
>>>>>> Maybe there is an magic device? I know that if we have an
>asterisk,
>>>>>> that become to us with default configuration of rtp ports sets
to
>>>>>> 10000_20000. And each
call choose the one port fron that
range.
>So if
>>>>>> we have several asterisks with default configuratiin of rtp,
>there is
>>>>>> possibilities to have 2 concurent calls each through another
>asterisk
>>>>>> instance with this same rtp port. Am i right?
>>>>>>
>>>>>> So mqybe this magic device could see source IP address and
route
>rtp
>>>>>> to correct adterisk?
>>>>>>
>>>>>> 13.03.2017 7:15 AM "Alex Balashov"
<abalashov(a)evaristesys.com
>>>>>> <mailto:abalashov@evaristesys.com>> napisał(a):
>>>>>>
>>>>>> On Mon, Mar 13, 2017 at 07:08:09AM +0100, Kjeld Flarup
>wrote:
>>>>>>
>>>>>> > We run multiple Asterisk instances since 1.4 and
never
>>>>>> configured RTP ports.
>>>>>> >
>>>>>> > More challenging issues are the Asterisk DB, and the
>Asteisk
>>>>>>
>>>>> home.
>>>>>
>>>>>> You may not have enough calls for RTP port collisions to
>become
>>>>>>
>>>>> an
>>>>>
>>>>>> issue. Otherwise, I'm not sure how you're avoiding
it,
since
>>>>>>
>>>>> Asterisk
>>>>>
>>>>>> isn't aware of which ports from within the range are in
use.
>>>>>>
>>>>>> --
>>>>>> Alex Balashov | Principal | Evariste Systems LLC
>>>>>>
>>>>>> Tel: +1-706-510-6800 <tel:%2B1-706-510-6800> /
>+1-800-250-5920
>>>>>> <tel:%2B1-800-250-5920> (toll-free)
>>>>>> Web: http://www.evaristesys.com/,
http://www.csrpswitch.com/
>>>>>>
>>>>>> _______________________________________________
>>>>>> SIP Express Router (SER) and Kamailio (OpenSER) -
sr-users
>>>>>>
>>>>> mailing
>>>>>
>>>>>> list
>>>>>> sr-users(a)lists.sip-router.org
>>>>>>
>>>>> <mailto:sr-users@lists.sip-router.org>
>>>>>
>>>>>>
>http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>>>
><http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>
>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> 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
>>>>>>
>>>>>
>>>> -- Alex
>>>>
>>>> --
>>>> Principal, Evariste Systems LLC (www.evaristesys.com)
>>>>
>>>> Sent from my Google Nexus.
>>>>
>>>> _______________________________________________
>>>> 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
>>>>
>>>
>>>
>>> _______________________________________________
>>> 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
>>
>
>
> _______________________________________________
> 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
>
>
-- Alex
--
Principal, Evariste Systems LLC (www.evaristesys.com)
Sent from my Google Nexus.
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
2:56 a.m.
An ambitious endeavour. Are you sure it's an economically sensible one, given that
there are a variety of solutions already out there?
On March 14, 2017 2:55:31 AM EDT, przeqpiciel <przeqpiciel(a)gmail.com> wrote:
I would like to create PBX platform, at now I faced to
make drag&drop
ivr
creator. After that I would create option for record calls for client
and
this is why I look for solution :)
2017-03-14 7:47 GMT+01:00 Alex Balashov <abalashov(a)evaristesys.com>om>:
-- Alex
--
Principal, Evariste Systems LLC (www.evaristesys.com)
Sent from my Google Nexus.
_______________________________________________
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
>
-- Alex
--
Principal, Evariste Systems LLC (www.evaristesys.com)
Sent from my Google Nexus.
Yes, though of course you would have to correlate
the calls (most
likely
by Call-ID) and integrate all this.
On March 14, 2017 2:46:27 AM EDT, przeqpiciel <przeqpiciel(a)gmail.com>
wrote:
>So, I can use Kamailio as SBC/Load balancer/registrar, Asterisk as
IVR
>and
>application server, and rtpproxy as media relay and recorder ?
>
>2017-03-14 7:44 GMT+01:00 Alex Balashov <abalashov(a)evaristesys.com>om>:
>
>> It can record, as can a number of other media relays.
>>
>> On March 14, 2017 2:43:15 AM EDT, przeqpiciel
<przeqpiciel(a)gmail.com>
>> wrote:
>> >>> WHy not installing rtpproxy and proxying all
>> >Because I would like to record some calls and I dont know
RTPProxy's
>> >features, maybe it could record ?
>> >
>> >2017-03-14 5:14 GMT+01:00 anfecora <anfecora(a)gmail.com>om>:
>> >
>> >> WHy not installing rtpproxy and proxying all rtp to the inside
>uase
>> >> kamailio to load balance them, it will be transparent on the
>inside
>> >perhaps
>> >> a cleaner solution?
>> >>
>> >> On Mon, Mar 13, 2017 at 3:21 PM, Kjeld Flarup <kfc(a)viptel.dk>
>wrote:
>> >>
>> >>> As I recall it is sequential, but not from the start
everytime,
>it
>> >is
>> >>> incrementing all the time.
>> >>>
>> >>> If You are running three servers, then with a 100% identical
>load,
>> >one
>> >>> would expect an average of 2 failing attempts per call.
>> >>>
>> >>> The reality I see is however often very different RTP ports,
most
>> >likely
>> >>> because load isn't 100% identical.
>> >>>
>> >>>
>> >>> Med venlig hilsen / Best regards
>> >>> Kjeld Flarup (Christensen) M.Sc E.E, Teknisk chef
>> >>> Viptel ApS, Hammershusvej 16C, DK-7400 Herning
>> >>> Telefon: +45 46949949, Telefax: +45 46949950, http://viptel.dk
>> >>>
>> >>> On 03/13/2017 11:05 PM, Alex Balashov wrote:
>> >>>
>> >>>> Well, indeed, but a sequential scan of many consecutive ports
>like
>> >this
>> >>>> from the bottom of the same range can be quite a latent
>operation.
>> >So at
>> >>>> the very least the allocation strategy would benefit from
being
>> >random.
>> >>>> Does Asterisk take that approach?
>> >>>>
>> >>>> On March 13, 2017 6:04:06 PM EDT, Kjeld Flarup
<kfc(a)viptel.dk>
>> >wrote:
>> >>>>
>> >>>>> No there is no such thing as magic.
>> >>>>>
>> >>>>> The most obvious way to implement the RTP port handling, is
to
>> >first
>> >>>>> open the next UDP port in the OS, and then report that back
in
>the
>> >>>>> Invite/200Ok. If the port cannot be opened, then simply try
the
>> >next in
>> >>>>>
>> >>>>> line.
>> >>>>>
>> >>>>>
>> >>>>> Med venlig hilsen / Best regards
>> >>>>> Kjeld Flarup (Christensen) M.Sc E.E, Teknisk chef
>> >>>>> Viptel ApS, Hammershusvej 16C, DK-7400 Herning
>> >>>>> Telefon: +45 46949949, Telefax: +45 46949950,
http://viptel.dk
>> >>>>>
>> >>>>> On 03/13/2017 01:52 PM, przeqpiciel wrote:
>> >>>>>
>> >>>>>> Maybe there is an magic device? I know that if we have
an
>> >asterisk,
>> >>>>>> that become to us with default configuration of rtp
ports
sets
>to
>> >>>>>> 10000_20000. And each call choose the one port fron
that
>range.
>> >So if
>> >>>>>> we have several asterisks with default configuratiin of
rtp,
>> >there is
>> >>>>>> possibilities to have 2 concurent calls each through
another
>> >asterisk
>> >>>>>> instance with this same rtp port. Am i right?
>> >>>>>>
>> >>>>>> So mqybe this magic device could see source IP address
and
>route
>> >rtp
>> >>>>>> to correct adterisk?
>> >>>>>>
>> >>>>>> 13.03.2017 7:15 AM "Alex Balashov"
<abalashov(a)evaristesys.com
>> >>>>>>
<mailto:abalashov@evaristesys.com>> napisał(a):
>> >>>>>>
>> >>>>>> On Mon, Mar 13, 2017 at 07:08:09AM +0100, Kjeld
Flarup
>> >wrote:
>> >>>>>>
>> >>>>>> > We run multiple Asterisk instances since 1.4
and
never
>> >>>>>> configured
RTP ports.
>> >>>>>> >
>> >>>>>> > More challenging issues are the Asterisk DB,
and the
>> >Asteisk
>> >>>>>>
>> >>>>> home.
>> >>>>>
>> >>>>>> You may not have enough calls for RTP port
collisions
to
>> >become
>> >>>>>>
>> >>>>> an
>> >>>>>
>> >>>>>> issue. Otherwise, I'm not sure how you're
avoiding it,
>since
>> >>>>>>
>> >>>>> Asterisk
>> >>>>>
>> >>>>>> isn't aware of which ports from within the
range are
in
>use.
>> >>>>>>
>> >>>>>> --
>> >>>>>> Alex Balashov | Principal | Evariste Systems LLC
>> >>>>>>
>> >>>>>> Tel: +1-706-510-6800 <tel:%2B1-706-510-6800>
/
>> >+1-800-250-5920
>> >>>>>> <tel:%2B1-800-250-5920> (toll-free)
>> >>>>>> Web: http://www.evaristesys.com/,
>http://www.csrpswitch.com/
>> >>>>>>
>> >>>>>> _______________________________________________
>> >>>>>> SIP Express Router (SER) and Kamailio (OpenSER) -
>sr-users
>> >>>>>>
>> >>>>> mailing
>> >>>>>
>> >>>>>> list
>> >>>>>> sr-users(a)lists.sip-router.org
>> >>>>>>
>> >>>>> <mailto:sr-users@lists.sip-router.org>
>> >>>>>
>> >>>>>>
>> >http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>> >>>>>>
>> ><http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>
>> >>>>>>
>> >>>>>>
>> >>>>>>
>> >>>>>> _______________________________________________
>> >>>>>> 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
>> >>>>>>
>> >>>>>
>> >>>> -- Alex
>> >>>>
>> >>>> --
>> >>>> Principal, Evariste Systems LLC (www.evaristesys.com)
>> >>>>
>> >>>> Sent from my Google Nexus.
>> >>>>
>> >>>> _______________________________________________
>> >>>> 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
>> >>>>
>> >>>
>> >>>
>> >>> _______________________________________________
>> >>> 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
>> >>>
>> >>
>> >>
>> >> _______________________________________________
>> >> 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
>>
>>
-- Alex
--
Principal, Evariste Systems LLC (www.evaristesys.com)
Sent from my Google Nexus.
_______________________________________________
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
>
4:05 a.m.
even if it is not an economically sensible then It will be pretty
situations to learn how to configure Kamailio and RTPProxy ;)
2017-03-14 7:56 GMT+01:00 Alex Balashov <abalashov(a)evaristesys.com>om>:
An ambitious endeavour. Are you sure it's an
economically sensible one,
given that there are a variety of solutions already out there?
On March 14, 2017 2:55:31 AM EDT, przeqpiciel <przeqpiciel(a)gmail.com>
wrote:
I would like to create PBX platform, at now I
faced to make drag&drop
ivr
creator. After that I would create option for record calls for client
and
this is why I look for solution :)
2017-03-14 7:47 GMT+01:00 Alex Balashov <abalashov(a)evaristesys.com>om>:
list
> sr-users(a)lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
-- Alex
--
Principal, Evariste Systems LLC (www.evaristesys.com)
Sent from my Google Nexus.
_______________________________________________
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
Yes, though of course you would have to correlate
the calls (most
likely
by Call-ID) and integrate all this.
On March 14, 2017 2:46:27 AM EDT, przeqpiciel <przeqpiciel(a)gmail.com>
wrote:
>So, I can use Kamailio as SBC/Load balancer/registrar, Asterisk as
IVR
>and
>application server, and rtpproxy as media relay and recorder ?
>
>2017-03-14 7:44 GMT+01:00 Alex Balashov <abalashov(a)evaristesys.com>om>:
>
>> It can record, as can a number of other media relays.
>>
>> On March 14, 2017 2:43:15 AM EDT, przeqpiciel
<przeqpiciel(a)gmail.com>
>> wrote:
>> >>> WHy not installing rtpproxy and proxying all
>> >Because I would like to record some calls and I dont know
RTPProxy's
>> >features, maybe it could record ?
>> >
>> >2017-03-14 5:14 GMT+01:00 anfecora <anfecora(a)gmail.com>om>:
>> >
>> >> WHy not installing rtpproxy and proxying all rtp to the inside
>uase
>> >> kamailio to load balance them, it will be transparent on the
>inside
>> >perhaps
>> >> a cleaner solution?
>> >>
>> >> On Mon, Mar 13, 2017 at 3:21 PM, Kjeld Flarup <kfc(a)viptel.dk>
>wrote:
>> >>
>> >>> As I recall it is sequential, but not from the start
everytime,
>it
>> >is
>> >>> incrementing all the time.
>> >>>
>> >>> If You are running three servers, then with a 100% identical
>load,
>> >one
>> >>> would expect an average of 2 failing attempts per call.
>> >>>
>> >>> The reality I see is however often very different RTP ports,
most
>> >likely
>> >>> because load isn't 100% identical.
>> >>>
>> >>>
>> >>> Med venlig hilsen / Best regards
>> >>> Kjeld Flarup (Christensen) M.Sc E.E, Teknisk chef
>> >>> Viptel ApS, Hammershusvej 16C, DK-7400 Herning
>> >>> Telefon: +45 46949949, Telefax: +45 46949950, http://viptel.dk
>> >>>
>> >>> On 03/13/2017 11:05 PM, Alex Balashov wrote:
>> >>>
>> >>>> Well, indeed, but a sequential scan of many consecutive ports
>like
>> >this
>> >>>> from the bottom of the same range can be quite a latent
>operation.
>> >So at
>> >>>> the very least the allocation strategy would benefit from
being
>> >random.
>> >>>> Does Asterisk take that approach?
>> >>>>
>> >>>> On March 13, 2017 6:04:06 PM EDT, Kjeld Flarup
<kfc(a)viptel.dk>
>> >wrote:
>> >>>>
>> >>>>> No there is no such thing as magic.
>> >>>>>
>> >>>>> The most obvious way to implement the RTP port handling, is
to
>> >first
>> >>>>> open the next UDP port in the OS, and then report that back
in
>the
>> >>>>> Invite/200Ok. If the port cannot be opened, then simply try
the
>> >next in
>> >>>>>
>> >>>>> line.
>> >>>>>
>> >>>>>
>> >>>>> Med venlig hilsen / Best regards
>> >>>>> Kjeld Flarup (Christensen) M.Sc E.E, Teknisk chef
>> >>>>> Viptel ApS, Hammershusvej 16C, DK-7400 Herning
>> >>>>> Telefon: +45 46949949, Telefax: +45 46949950,
http://viptel.dk
>> >>>>>
>> >>>>> On 03/13/2017 01:52 PM, przeqpiciel wrote:
>> >>>>>
>> >>>>>> Maybe there is an magic device? I know that if we have
an
>> >asterisk,
>> >>>>>> that become to us with default configuration of rtp
ports
sets
>to
>> >>>>>> 10000_20000. And each call choose the one port fron
that
>range.
>> >So if
>> >>>>>> we have several asterisks with default configuratiin of
rtp,
>> >there is
>> >>>>>> possibilities to have 2 concurent calls each through
another
>> >asterisk
>> >>>>>> instance with this same rtp port. Am i right?
>> >>>>>>
>> >>>>>> So mqybe this magic device could see source IP address
and
>route
>> >rtp
>> >>>>>> to correct adterisk?
>> >>>>>>
>> >>>>>> 13.03.2017 7:15 AM "Alex Balashov"
<abalashov(a)evaristesys.com
>> >>>>>>
<mailto:abalashov@evaristesys.com>> napisał(a):
>> >>>>>>
>> >>>>>> On Mon, Mar 13, 2017 at 07:08:09AM +0100, Kjeld
Flarup
>> >wrote:
>> >>>>>>
>> >>>>>> > We run multiple Asterisk instances since 1.4
and
never
>> >>>>>> configured
RTP ports.
>> >>>>>> >
>> >>>>>> > More challenging issues are the Asterisk DB,
and the
>> >Asteisk
>> >>>>>>
>> >>>>> home.
>> >>>>>
>> >>>>>> You may not have enough calls for RTP port
collisions
to
>> >become
>> >>>>>>
>> >>>>> an
>> >>>>>
>> >>>>>> issue. Otherwise, I'm not sure how you're
avoiding it,
>since
>> >>>>>>
>> >>>>> Asterisk
>> >>>>>
>> >>>>>> isn't aware of which ports from within the
range are
in
>use.
>> >>>>>>
>> >>>>>> --
>> >>>>>> Alex Balashov | Principal | Evariste Systems LLC
>> >>>>>>
>> >>>>>> Tel: +1-706-510-6800 <tel:%2B1-706-510-6800>
/
>> >+1-800-250-5920
>> >>>>>> <tel:%2B1-800-250-5920> (toll-free)
>> >>>>>> Web: http://www.evaristesys.com/,
>http://www.csrpswitch.com/
>> >>>>>>
>> >>>>>> _______________________________________________
>> >>>>>> SIP Express Router (SER) and Kamailio (OpenSER) -
>sr-users
>> >>>>>>
>> >>>>> mailing
>> >>>>>
>> >>>>>> list
>> >>>>>> sr-users(a)lists.sip-router.org
>> >>>>>>
>> >>>>> <mailto:sr-users@lists.sip-router.org>
>> >>>>>
>> >>>>>>
>> >http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>> >>>>>>
>> ><http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>
>> >>>>>>
>> >>>>>>
>> >>>>>>
>> >>>>>> _______________________________________________
>> >>>>>> 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
>> >>>>>>
>> >>>>>
>> >>>> -- Alex
>> >>>>
>> >>>> --
>> >>>> Principal, Evariste Systems LLC (www.evaristesys.com)
>> >>>>
>> >>>> Sent from my Google Nexus.
>> >>>>
>> >>>> _______________________________________________
>> >>>> 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
>> >>>>
>> >>>
>> >>>
>> >>> _______________________________________________
>> >>> 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
>> >>>
>> >>
>> >>
>> >> _______________________________________________
>> >> 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
> >>
> >>
>
>
> -- Alex
>
> --
> Principal, Evariste Systems LLC (www.evaristesys.com)
>
> Sent from my Google Nexus.
>
> _______________________________________________
> 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
>
-- Alex
--
Principal, Evariste Systems LLC (www.evaristesys.com)
Sent from my Google Nexus.
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
2884
days inactive
2886
days old
19 comments
6 participants
participants (6)
-
Alex Balashov -
anfecora -
Brandon Armstead -
Gonzalo Gasca Meza -
Kjeld Flarup -
przeqpiciel