Hi Daniel.
I just build with latest commit.
It work as expected now.
Now I can see captured packets as they was send and received by
kamailio via network interfaces.
Thank you very match.
I make patch which backports your commits to kamailio 4.3.4
please find it in attach
--
Best regards,
Sergey
Hi, Thanks for the answer.
Do you have any options for sending this keys to opensips somehow, by
modifying the code in rtpengine and in opesips script file?
Also, I have another query. The SRTP keys that we are getting after the
DTLS handshake is common for both audio and video streams for both R
Not familiar with this per se, but as I understood, graphite is just a
rendering engine gathering metrics from a storage engine such as statsd.
So might be that rtpengine has the option to push metrics to statsd,
then graphite can build graphs out of them easily.
Cheers,
Daniel
On 13/01/16 03:53,
Hello,
the error with creating the SIP UA is most probable because of kamailio
listening on 5060 and freeswitch trying to do the same.
To troubleshoot the 408, use ngrep or other network sniffing tool, and
look on the network to see where the sip request is sent. Like:
ngrep -d any -qt -W byline
rtpengine README.md mentions "graphite statistics server". is that
server openly available somewhere?
-- juha
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Hello Daniel,
No I didn't configure freeswitch with loopback but for some reason it was going
to the loopback , it consider it as default network but I am able to point both
kamailio and freeswitch to 10.22.52.2 by disabling IP-v6 for both
external-ipv6.xml and internal-ipv6.xml. Freeswitch was
Hello,
can you get the SIP trace with all the packets of such dialog outside of
the NAT router? It will help to see the headers and based on that we may
be able to provide a solution.
Cheers,
Daniel
On 12/01/16 19:13, Nelson Migliaro wrote:
> Thank you for your answer.
>
> The problem I have is
Hello,
did you configure the freeswitch to listen on loopback? You would need
to do bridging of singnaling and eventually rtp between the network
interface and loopback if you want this kind of topology.
Cheers,
Daniel
On 12/01/16 19:00, malik sherif wrote:
>
> Hello Abdul Basit,
>
> I specified
Earlier today I pushed the patch to catch the incoming traffic before
topoh gets the chance to decode it. If you can try with latest master
and report the results, it will be appreciated.
Cheers,
Daniel
On 12/01/16 11:02, Sergey Basov wrote:
> I am sorry.
> I have missed commit 4fc969760d8eec6355
Any body has kamailio/freeswitch SBC working?
From: sr-users on behalf of malik
sherif
Sent: Tuesday, January 12, 2016 6:00 PM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC
Hello Abdul Basit,
I sp
Thank you for your answer.
The problem I have is with internet router doing to PAT to SIP port.
I am already advertising public IP but unfortunately I cant know the public
port I am using.
2015-12-28 18:17 GMT+01:00 Alexandru Covalschi <568...@gmail.com>:
> AFAIK bye is usually sent to the addre
Hello Abdul Basit,
I specified that kamailio and freeswitch to share same IP address but different
udp port but netstat -unlp show freeswitch use the loopback IP address (port
5090 and 5092) and kamailio show the loopback IP and 10.22.52.2 port 5060. Does
freeswitch has to go to loopback IP add
On 01/12/2016 04:09 AM, riko nir wrote:
Hi all,
I have a media server and it is able to handle SRTP, provided the crypto
key.
We are planning to give webrtc support to the media server. We are using
opensips+rtpengine for that.
For dtls, we are using rtpengine. The rtpengine just needs to do t
Hello,
optimizing for performances is a matter of configuration file. A special
attention must be done to database and dns interactions.
Some tips for performances are collected in the next presentation:
-
http://www.kamailio.org/events/2015-Astricon/dcm-kamailio-surfing-big-waves-of-sip-with-st
You should see the received packets if you use latest master, but they
are not yet captured before topoh decodes them.
Cheers,
Daniel
On 12/01/16 09:45, Sergey Basov wrote:
> Thank you for commits.
>
> I just try it and now I can see that sending packets captured properly..
>
> Now there is a pro
I am sorry.
I have missed commit 4fc969760d8eec6355ce661ccd3c5fd9ad2a36f0...
Now all works as I had expected.
Than you so match.
If you are interesting, I now try to backport changes to kamailio 4.3.4.
When it will be done I can email patch file to you.
--
Best regards,
Sergey Basov
Hi all,
I have a media server and it is able to handle SRTP, provided the crypto
key.
We are planning to give webrtc support to the media server. We are using
opensips+rtpengine for that.
For dtls, we are using rtpengine. The rtpengine just needs to do the dtls
handshake and it needs to fetch th
Hi all,
I have a media server and it is able to handle SRTP, provided the crypto
key.
We are planning to give webrtc support to the media server. We are using
opensips+rtpengine for that.
For dtls, we are using rtpengine. The rtpengine just needs to do the dtls
handshake and it needs to fetch th
Thank you for commits.
I just try it and now I can see that sending packets captured properly..
Now there is a problem with receiving packets, I does not see Initial
invite from client into the capture.
--
Best regards,
Sergey Basov e-mail: sergey.v.ba...@gmail.com
tel: (+
Hello Daniel,
I've enabled the debug only for a moment to get a better understanding of
Kamailio and what was happening under the hood.
I've done some testing on a newer version of Kamailio (4.3) and didn't see this
problem occur. So I'm not sure if it's also software version related.
What els
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