[SR-Users] Several Asterisk on this same IP

Kjeld Flarup kfc at viptel.dk
Mon Mar 13 23:21:00 CET 2017


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