Hi all,
I have Opensips 3.0 + RTPengine setup on an AWS instance. I tried CGRates
recently, but couldn't get it to work (calls weren't redirected to the GW
if allowed, however, were declined if the call wasn't authorised).
Does anyone else have any experience with the above issue? Or is some othe
rm
(julien.royann...@orange.com)
--
Message: 1
Date: Fri, 29 Jul 2022 10:17:45 +
From:
To: "users@lists.opensips.org"
Subject: [OpenSIPS-Users] Opensips/Rtpengine Docker to Docker Swarm
Message-ID:
<1974_1659089865
Hi there,
I successfully use a containerized SBC with Opensips/Rtpengine on a single
virtual machine with a docker engine.
I need to do it on a Docker Swarm context. I can't manage to set up this
Opensips/RTP SBC in this SWARM context.
On a single VM with docker, the container has only one net
amp; $proto=="tcp|udp") {
route(manage_rtp);
} else if(is_method("BYE|CANCEL")) {
rtpengine_delete();
}
t_relay();
exit;
}
volga629
From: "Karsten Horsmann"
To: "volga629" , "OpenSIPS users mailling list"
Cc: "Alain Bieuzent&q
uzent ,
> OpenSIPS users mailling list
>
> *Objet : *Re: [OpenSIPS-Users] opensips + rtpengine
>
>
>
> Hello Alain,
>
>
>
> port-min = 5000
>
> port-max = 5
>
> delete-delay = 5
>
> timeout = 10
>
> silent-timeout = 900
>
>
>
>
Hello Everyone,
Looking to have a rock solid cluster. i have been working for opensips
for long time and it is very stable. But this time i plan to run it with
rtpengine.
Would you recommend having a cluster on two bare metal servers? Or have
two Virtual machines as cluster on a bigger serv
Objet : Re: [OpenSIPS-Users] opensips + rtpengine
Hello Alain,
port-min = 5000
port-max = 5
delete-delay = 5
timeout = 10
silent-timeout = 900
onreply_route
: Re: [OpenSIPS-Users] opensips + rtpengine
Hello Alain,
port-min = 5000
port-max = 5
delete-delay = 5
timeout = 10
silent-timeout = 900
onreply_route[handle_media_reply] {
xlog("incoming reply\n");
if(is_method("INVITE|UPDATE") &&
Hi,
Can you share value of delete-delay, port-min and port-max of your rtpengine configuration.
Have you also check if you handle rtpengine_delete on failed calls (in case sip cause code 4XX, 5XX and 6XX).
At @job, we handle max 6000 calls on a 6 cores servers without any issue.
Rega
g
list
Date : vendredi 13 mars 2020 à 18:39
À :
Objet : [OpenSIPS-Users] opensips + rtpengine
Hello
Everyone,
Might
be somebody can point me to right place.
Und
Can you share value of delete-delay, port-min and port-max of your rtpengine configuration.
Have you also check if you handle rtpengine_delete on failed calls (in case sip cause code 4XX, 5XX and 6XX).
At @job, we handle max 6000 calls on a 6 cores servers without any issue.
Regards
port-max of your rtpengine configuration.
Have you also check if you handle rtpengine_delete on failed calls (in case sip cause code 4XX, 5XX and 6XX).
At @job, we handle max 6000 calls on a 6 cores servers without any issue.
Regards
De : Users au nom de volga629 via Users
Répon
6XX).
>
>
>
> At @job, we handle max 6000 calls on a 6 cores servers without any issue.
>
>
>
> Regards
>
>
>
>
>
>
>
> De : Users au nom de volga629 via Users
>
> Répondre à : volga629 , OpenSIPS users mailling list
>
> Date :
.
Regards
De : Users au nom de volga629 via Users
Répondre à : volga629 , OpenSIPS users mailling list
Date : vendredi 13 mars 2020 à 18:39
À :
Objet : [OpenSIPS-Users] opensips + rtpengine
Hello Everyone,
Might be somebody can point me to right place.
Under load Rtpengine on
Hello Everyone,
Might be somebody can point me to right
place.
Under load Rtpengine on server with 12
core can't pass 400 channels/sessions.
Mar 13 18:14:53 CentOS-77-64-minimal
rtpengine[14588]: WARNING:
[1b17077c-654e-11ea-bd31-87b1c8fc-849]
And are you forcing RTPengine to act as an ice light client? It looks
like you are gettin a single ICE candidate in the answer back from
freeswitch which would indicate that you are.
I'd check your chrome webrtc statistics to see if tis failed to do do
ice/stun negotiation on the 183. In gen
Yes fingerprints are different in Invite and session progress.
On Fri, Jun 24, 2016 at 3:00 AM, sevpal wrote:
> Take a look at the “fingerprint:” line.
>
> *From:* John Nash
> *Sent:* Thursday, June 23, 2016 3:42 PM
> *To:* OpenSIPS users mailling list
> *Subject:* R
128 and 256 to 256.
>
> You can print the request body ($rb) on the INVITE with “application/sdp”
> and visually compare the exchange, do this on offer and answer.
>
> *From:* John Nash
> *Sent:* Thursday, June 23, 2016 3:42 PM
> *To:* OpenSIPS users mailling list
> *
Take a look at the “fingerprint:” line.
From: John Nash
Sent: Thursday, June 23, 2016 3:42 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc
Actually the issue is i hear no audio on either side and just after session
progress (I guess when
exchange, do this on offer and answer.
From: John Nash
Sent: Thursday, June 23, 2016 3:42 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc
Actually the issue is i hear no audio on either side and just after session
progress (I guess when
Actually the issue is i hear no audio on either side and just after session
progress (I guess when media starts coming from remote media server) i see
error "SRTP output wanted, but no crypto suite was negotiated"
I had also checked media logs i could see RTP packets being sent from
freeswitch to
So - i dont see a problem here - Chrome is getting UDP/TLS/RTP/SAVPF and
Freeswitch is getting RTP/AVP. Freeswitch responded to the offer in the
invite with an answer in the 183, and in the 200. What is the failure
you are seeing, and where is it happening (in freeswitch? in the browser?)
Th
OK here is the log
https://gist.github.com/johnnash13/0d2cb5238f3551cd3a8c6b4e638dd744
Sorry took me a while to convert wireshark trace to text file.
My freeswitch is running on private IP (127.0.0.1) and opensips I run on
both public and private so that for outside world opensips is the only
pub
No - it's annoying to look at a trace that's had information removed and
try and piece together whats happening. Your paranoid side is wrong, sorry.
-Eric
On 06/23/2016 01:06 PM, Patrick Wakano wrote:
my paranoic side would recommend to hide/change private informations,
specially any authenti
my paranoic side would recommend to hide/change private informations,
specially any authentication line that might appear... this is certainly a
sort of social engineering threat we should worry...
better be safe than sorry
On Thu, Jun 23, 2016 at 3:31 PM, Eric Tamme wrote:
> I mean you can
I mean you can use a private gist, but you will be publishing the link
in a public email list. In general I personally dont believe revealing
ip addresses etc. is any problem - to put my money where my mouth is
here is a gist link to an unaltered SIP trace on my server :)
https://gist.github.
Ok i am ready with logs. About gist may I use private option as traces have
our IPs, user
On Thu, Jun 23, 2016 at 10:32 PM, Eric Tamme wrote:
> Hey John,
>
> Please paste a full UNALTERED sip trace into a gist (gist.github.com)
> from the proxy servers perspective and provide a link so that we c
Hey John,
Please paste a full UNALTERED sip trace into a gist (gist.github.com)
from the proxy servers perspective and provide a link so that we can see
what comes in, and what goes out from both sides.
EG: ngrep -qtd any -W byline port 5060
This will show us the traffic that is leaving the
I double checked my rtpengine offer answer calls and now using
https://github.com/onsip/sipjs-examples/tree/master/demo-phone but I face
same issue (no audio either side) and error "SRTP output wanted, but no
crypto suite was negotiated" Rtpengine also I updated to the latest now.
Am I using corr
Thank you Eric,
I will give it a try.
On Thu, Jun 23, 2016 at 7:58 PM, Eric Tamme wrote:
> 1. I would suggest using SIP.js - https://github.com/onsip/SIP.js it is a
> much more active project that sipml5.
>
> 2. Im guessing that you are not properly passing flags to RTPEngine. If
> you want to
1. I would suggest using SIP.js - https://github.com/onsip/SIP.js it is
a much more active project that sipml5.
2. Im guessing that you are not properly passing flags to RTPEngine. If
you want to have DTLS-SRTP between the browser, and plain RTP/AVP
between RTPEngine and freeswitch, you need
I am following http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2
and trying to test a call
sipml5 --->Opensips + rtpengine > SIP end point (Freeswitch)
But I do not have any audio on both sides. I see this error at rtpengine
log "SRTP output wanted, but no crypto suite
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