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 > >>
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>>
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>> > 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)
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