etc in the offer
options to specify which address you wish to present for each scenario
you are handling.
This also applies to -A however you should only need that if running
those addresses behind NAT.
Hope that helps to move you forward!
Callum
On Wed, 19 Apr 2023 at 15:01, Gavin Mur
,
Gavin
On Wednesday, April 19, 2023 at 02:09:13 PM ADT, Gavin Murphy via Users
wrote:
Yes. I am assuming the rtpproxy_offer("ei") should be used when the INVITE
comes in from the external network, and rtpproxy_answer("ie") should be used in
the onreply_route to handle
I have fixed most of the issues I had. I think that in the process of testing
I mixed up an ie<->ei. Things are set up properly now and audio is flowing to
the client on my mobile phone.
Thanks for the help.
Gavin
On Wednesday, April 19, 2023 at 02:19:44 PM ADT, Gavin Murphy
nternal and external addresses
so that it can present the public IP to the client and private to the
application server.
You probably don't need fix_nated_sdp as rtpproxy/rtpengine will do
that for you, you'll just need to tell it which address to use which
differs for each product.
Good l
main route for a non-LR
INVITE and rtp_answer("ie") in the on reply route (when handling the
183 response).
Is that what you're doing?
On Wed, 19 Apr 2023 at 17:08, Gavin Murphy wrote:
>
> Hi Callum,
>
> thanks for the additional info. I removed the -A option from th
Hello,
I'm trying to set up an instance of opensips to support a testing SIP phone
calling into my simulated network. The client is running from a mobile phone.
The connection from the client comes in from the public network, but the client
sees its own IP as private (192.0.0.2). My test net
Hi all,
I did a search on this and the most recent discussion I saw was
from 2010, so I figured I would pose the question again: is there a way
to have OpenSIPS automatically adjust the transport from UDP to TCP when
the request size is within 200 bytes of the MTU size, as per section
18.
Hi Chris,
it would seem (based on the description) that the fr_timer setting
is for the final timeout, and not related to retransmissions, but please
correct me if I have interpreted the description incorrectly. In RFC
3261, it appears that Timer E is the one that governs the
retransmissi
Hi all,
is it possible use a provisional response to suppress the
retransmission of SIP MESSAGEs (over UDP)? A test whereby a 100 Trying
was sent didn't seem to do the trick. In our case there could be a
significant amount of time (many seconds) between the time the original
MESSAGE is sen
ng, therefore, if any script or module command
that consumes time e.g. custom sql query, external script or shell
command etc. would block opensips and any responses received during that
time are likely to be ignored.
Thank you.
On 2013-11-14 16:05, Gavin Murphy wrote:
Hi all,
We're seeing
xx =>
+xx)
etc
This "bogging down" seems to happen on a regular basis, and the only way
we are able to resolve it at the moment is to restart OpenSIPS. Until it
is restarted all of the registrations are essentially failing. I believe
it is v1.8.2 that we are runn
We're running some tests with IPv6, and we're seeing OpenSIPS report the
following error:
Sep 5 12:40:18 user opensips[30809]: DBG:core:parse_via: next_via
Sep 5 12:40:18 user opensips[30809]: ERROR:core:parse_via: bad port
Sep 5 12:40:18 user opensips[30809]: ERROR:core:parse_via: ;tag=jge8
A couple of questions/comments regarding the 477 Send Failed condition:
1) When a 477 is sent, is there a way to detect it within a script so
that some custom handling can be applied (primarily logging)?
2) When the condition happens, and number of error logs are emitted. For
example:
Aug 27
Is it possible to load balance non-INVITE dialogs? In particular I am
encountering the following error when sending SUBSCRIBEs and PUBLISHes:
ERROR:load_balancer:do_load_balance: failed to create dialog
Where those requests can have an extended lifetime and may consume
resources on the server
We're trying to load up opensips with as many TCP connections as we
possibly can. So far we've got it to about 82K, but failures start
occurring at that point. We have 8GBs of RAM allocated to the server as
a whole (is that enough? we don't appear to be exhausting it). We've set
the following p
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