It can record, as can a number of other media relays.
On March 14, 2017 2:43:15 AM EDT, przeqpiciel przeqpiciel@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@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@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@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@evaristesys.com mailto:abalashov@evaristesys.com> napisał(a):
On Mon, Mar 13, 2017 at 07:08:09AM +0100, Kjeld Flarupwrote:
> We run multiple Asterisk instances since 1.4 and never configured RTP ports. > > More challenging issues are the Asterisk DB, and theAsteisk
home.
You may not have enough calls for RTP port collisions tobecome
an
issue. Otherwise, I'm not sure how you're avoiding it, sinceAsterisk
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-usersmailing
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