[SR-Users] Several Asterisk on this same IP

Tue Mar 14 07:46:27 CET 2017

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 at evaristesys.com>:

> It can record, as can a number of other media relays.
>
> On March 14, 2017 2:43:15 AM EDT, przeqpiciel <przeqpiciel at 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 at 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 at 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 at 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 at evaristesys.com
> >>>>>> <mailto:abalashov at 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
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> >>>>>>
> >>>>> <mailto:sr-users at lists.sip-router.org>
> >>>>>
> >>>>>>
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> >>>>>>
> ><http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>> _______________________________________________
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> >>>>>>
> >>>>>
> >>>> -- Alex
> >>>>
> >>>> --
> >>>> Principal, Evariste Systems LLC (www.evaristesys.com)
> >>>>
> >>>> Sent from my Google Nexus.
> >>>>
> >>>> _______________________________________________
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> >>>>
> >>>
> >>>
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> >>
> >>
> >> _______________________________________________
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> >>
>
>
> -- Alex
>
> --
> Principal, Evariste Systems LLC (www.evaristesys.com)
>
> Sent from my Google Nexus.
>
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