Hi,
I am setting up Raspberry Pi to run Kamailio with mediaproxy-ng. This
machine is running on local LAN behind a DSL router. Using dynamic DNS and
DMZ services of router, i can access this box from the Internet.
However, i do not know how to define advertised_address parameter to public
IP of router. The public IP is dynamic and changes roughly every 12 hours
or so. Is there any way kamailio can automatically learn this dynamic
public IP and use that as advertised_address?
Something similar to Asterisk's res_stun_monitor module,
https://reviewboard.asterisk.org/r/854/
Any help will be greatly appreciated.
Thank you.
Dear Daniel and Kamailio'ns
Greetings,
I am working on kamailio server (4.0.x),whicch is integreted with
media-proxy server (2.5.2) and it is all working fine in the point of SIP
functionality.
And i have a set-up like this below, with this when the calls (audio/video)
sessions started i am experiencing latency,jitter, echo's in audio calls
and a choppy ,pixelled video :
KAMAILIO
|
Android SIP client (IMSDroid) ---- Wi-Fi
router ---- Switch (level 2) ---- Wi-Fi router ---- Android SIP client
(IMSDroid)
(Total set-up runs on Intranet infrastructure).
When i call between two SIP clients, Calls are getting established
successfully and RTP flow is also relaying using Media-proxy server. But
the audio/video performnce is not good enough. I am experiencing all this
Latency , jitter , Choppy pixelled video.
How can i resolve all this issues? What could be the problem ?
Ofcourse SIP proxy is not responsible all this problems (as i know ), But
is there anything can be done on Media-proxy server ? Or anything Buffer
settings can be done on Kamailio server side (or anything else) ?
Even i changed Shared memory and Package memories to 512MB and 16MB
respectively.
Anybody can guess what may be the problems that causes these issues ?
PS: When i tried 'ping' Clients IP addresses in Kamailio server PC, there
is 20-30% packet loss also.
Please help me in resolving these issues.
Any help will greatly appreciate.
Regards,
Ravi.
John,
What db_mode are you using for dialog module? Are the entries in the db if
you check manually?
Are you able to fetch other data using avp_db_query? And do you see any
errors in the log?
Regards,
Charles
On 1 Feb 2014 16:24, "John Murray" <john.murray(a)skyracktelecom.com> wrote:
> Hello,
>
>
>
> I am managing calls with the dialog module in kamailio 4.0.4.
>
>
>
> However if is try to get the callid and from_tag of existing calls using:
>
>
>
> avp_db_query("select callid, from_tag from dialog", "$avp(s:s_callid),
> $avp(s:s_from_tag)");
>
>
>
> I get NULL, yet if I use:
>
>
>
> sercmd proxy dlg.list
>
>
>
> I get the calls listed correctly.
>
>
>
> What am I doing wrong?
>
>
>
> Thanks
>
>
>
> John
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users(a)lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
--
www.sipcentric.com
Follow us on twitter @sipcentric <http://twitter.com/sipcentric>
Sipcentric Ltd. Company registered in England & Wales no. 7365592. Registered
office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham
B7 4EJ.
Hello,
I am managing calls with the dialog module in kamailio 4.0.4.
However if is try to get the callid and from_tag of existing calls using:
avp_db_query("select callid, from_tag from dialog", "$avp(s:s_callid),
$avp(s:s_from_tag)");
I get NULL, yet if I use:
sercmd proxy dlg.list
I get the calls listed correctly.
What am I doing wrong?
Thanks
John
Hi Support,
I have recently kamailio 4.0.x version with tls on port 5061. Now our
development team needs to connect to it using unencrypted on port 5060 to
test an app for features and then work towards connecting through tls
connection.
So my question is how do I temporarily disable tls to test all the features
on the kamailio server and then once feature testing is done revert back to
tls connection?
Best Regards,
Neville D'Souza
Hi,
My 2 colleague and me (3 total) like to join the Kamailio dinner.
We went to Beermania 2 years ago with the Jitsi guys and DanB and I can
confirm the perfect fries!
Ivo
Hello there,
I think that i found a bug within failure routes on version 4.1.1, i have a
call flow that needs to use the failure routes 4 times if nobody answer the
call, but when the call goes to the 3º attempt to same failure route
kamailio doesn't handle the transaction,
With the same kamailio script but using the version 4.0.1 kamailio works
correctly.
Please check the attachment with kamailio debug level 3
--
Cumprimentos
José Seabra