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?
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@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.
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@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@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.
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,
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@cryy.com:
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@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@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.
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
-- Sent from Gmail Mobile
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.
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@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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.
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.
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@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@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@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@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@lists.sip-router.org 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@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@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 Asteiskhome.
You may not have enough calls for RTP port collisions to becomean
issue. Otherwise, I'm not sure how you're avoiding it, sinceAsterisk
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-usersmailing
list sr-users@lists.sip-router.orgmailto: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@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.
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@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@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 Asteiskhome.
You may not have enough calls for RTP port collisions to becomean
issue. Otherwise, I'm not sure how you're avoiding it, sinceAsterisk
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-usersmailing
list sr-users@lists.sip-router.orgmailto: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@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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@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@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@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 Asteiskhome.
You may not have enough calls for RTP port collisions to becomean
issue. Otherwise, I'm not sure how you're avoiding it, sinceAsterisk
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-usersmailing
list sr-users@lists.sip-router.orgmailto: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@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@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@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@gmail.com:
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@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@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@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 Asteiskhome.
You may not have enough calls for RTP port collisions to becomean
issue. Otherwise, I'm not sure how you're avoiding it, sinceAsterisk
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-usersmailing
list sr-users@lists.sip-router.orgmailto: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@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@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@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@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@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@gmail.com:
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@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@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@evaristesys.com mailto:abalashov@evaristesys.com> napisał(a):
On Mon, Mar 13, 2017 at 07:08:09AM +0100, Kjeld Flarupwrote:
> We run multiple Asterisk instances since 1.4 and never configured RTP ports. > > More challenging issues are the Asterisk DB, and theAsteisk
home.
You may not have enough calls for RTP port collisions tobecome
an
issue. Otherwise, I'm not sure how you're avoiding it, sinceAsterisk
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-usersmailing
list sr-users@lists.sip-router.orghttp://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@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@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@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@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.
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@evaristesys.com:
It can record, as can a number of other media relays.
On March 14, 2017 2:43:15 AM EDT, przeqpiciel przeqpiciel@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@gmail.com:
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@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@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@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@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@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@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@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@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@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@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@evaristesys.com:
It can record, as can a number of other media relays.
On March 14, 2017 2:43:15 AM EDT, przeqpiciel przeqpiciel@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@gmail.com:
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@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@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@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/,
>> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) -
sr-users
>> > mailing > >> list >> sr-users@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@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@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@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@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@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.
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@evaristesys.com:
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@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@evaristesys.com:
It can record, as can a number of other media relays.
On March 14, 2017 2:43:15 AM EDT, przeqpiciel przeqpiciel@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@gmail.com:
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@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@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@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/,
>>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) -
sr-users
>>> >> mailing >> >>> list >>> sr-users@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@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@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@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@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@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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@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@evaristesys.com:
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@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@evaristesys.com:
It can record, as can a number of other media relays.
On March 14, 2017 2:43:15 AM EDT, przeqpiciel
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@gmail.com:
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@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
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,
>>> >>> 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@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/,
>>>> >>>> _______________________________________________ >>>> SIP Express Router (SER) and Kamailio (OpenSER) -
sr-users
>>>> >>> mailing >>> >>>> list >>>> sr-users@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@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@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@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@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@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@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.
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@evaristesys.com:
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@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@evaristesys.com:
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@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@evaristesys.com:
It can record, as can a number of other media relays.
On March 14, 2017 2:43:15 AM EDT, przeqpiciel
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@gmail.com:
> 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@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
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,
>>>> >>>> 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@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/,
>>>>> >>>>> _______________________________________________ >>>>> SIP Express Router (SER) and Kamailio (OpenSER) -
sr-users
>>>>> >>>> mailing >>>> >>>>> list >>>>> sr-users@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@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@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@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@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@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@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-
Alex Balashov -
anfecora -
Brandon Armstead -
Gonzalo Gasca Meza -
Kjeld Flarup -
przeqpiciel