[SR-Users] Several Asterisk on this same IP

przeqpiciel przeqpiciel at gmail.com
Tue Mar 14 07:43:15 CET 2017


>> 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
>>>>>
>>>> mailing
>>>>
>>>>>      list
>>>>>      sr-users at lists.sip-router.org
>>>>>
>>>> <mailto:sr-users at 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 at 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 at 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 at 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 at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20170314/859779e6/attachment.html>


More information about the sr-users mailing list