It can record, as can a number of other media relays.
On March 14, 2017 2:43:15 AM EDT, przeqpiciel <przeqpiciel(a)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(a)gmail.com>om>:
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(a)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(a)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(a)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/,
http://www.csrpswitch.com/
>>>>
>>>> _______________________________________________
>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
>>>>
>>> mailing
>>>
>>>> list
>>>> sr-users(a)lists.sip-router.org
>>>>
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>>>
>>>>
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>>>>
<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>
>>>>
>>>>
>>>>
>>>>>
_______________________________________________
>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
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>>>>
>>>> list
>>>>
>>>>> sr-users(a)lists.sip-router.org
>>>>>
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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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