[SR-Users] Several Asterisk on this same IP

Alex Balashov abalashov at evaristesys.com
Tue Mar 14 07:47:55 CET 2017


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 at 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 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>> napisaƂ(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
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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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>> >>
>> >>
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>> >>
>>
>>
>> -- Alex
>>
>> --
>> Principal, Evariste Systems LLC (www.evaristesys.com)
>>
>> Sent from my Google Nexus.
>>
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>list
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>>


-- Alex

--
Principal, Evariste Systems LLC (www.evaristesys.com)

Sent from my Google Nexus.



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