April 24, 2012 10:32 AM
> To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users
> Mailing List
> Subject: Re: [SR-Users] RTPproxy and kamailio
>
> So let me get this straight , so if i have a client behind a NAT , the client
> should be able to conne
SER) - Users
>> Mailing List
>> Subject: Re: [SR-Users] RTPproxy and kamailio
>>
>> Thank you it worked well enough , but do you have an idea if there any
>> configuration for Stun server ? , or i just install stun server and link
>> it up to kamailio ?
>>
-users-
> boun...@lists.sip-router.org] On Behalf Of Ryan Gholam
> Sent: Tuesday, April 24, 2012 9:02 AM
> To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users
> Mailing List
> Subject: Re: [SR-Users] RTPproxy and kamailio
>
> Thank you it worked well enough ,
Thank you it worked well enough , but do you have an idea if there any
configuration for Stun server ? , or i just install stun server and
link it up to kamailio ?
On 4/23/12, Jan Klepal wrote:
> Hello Ryan,
>
> kamailio default config is already prepared for rtpproxy. Do exactly what is
> noted
Hello Ryan,
kamailio default config is already prepared for rtpproxy. Do exactly what is
noted in kamailio.cfg :-)
- download and install rtpproxy from http://www.rtpproxy.org
- start rtpproxy: rtpproxy -l your.public.ip.address -s udp:127.0.0.1:7722
- start kamailio: kamailio -A WITH_NAT
Funct
I would appreciate if someone can guide me through the setup to
configure Rtpproxy with kamailio , and what changes should be made in
the kamailio.cfg file .
Thank you .
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-use
Hi List,
i tcpdump on my kamailio server.
when uac registered with source port 5060
the uac will remain Cflag:: 0 and did not have the Received: field.
i also tried some iphone sip apps.
some uac app will register with src port 5060 and some will not.
anyone have solution for this?
or how t
Hi,
Thanks for the suggestion. i tried running rtpproxy on private.
rtpproxy -l public_ip -s udp:192.168.2.3:7722 -u user
and in kamailio cfg.
modparam("rtpproxy", "rtpproxy_sock", "udp:192.168.2.3:7722")
but still the same issue..
thanks.
--
Regards,
MingHon
__
Try to run rtpproxy on private ip not on local 127.0.0.1
On Tue, Jun 28, 2011 at 11:23 AM, MingHon wrote:
> Hi,
>
> i fixed the audio issue for 102 to 103 vice versa.
>
> by fixing the canreinvite in asterisk.
>
> from uac the rtp packet will route to kamailio den forward to asterisk.
>
> can we
Hi,
i fixed the audio issue for 102 to 103 vice versa.
by fixing the canreinvite in asterisk.
from uac the rtp packet will route to kamailio den forward to asterisk.
can we bypass the rtp packet going to asterisk?
and here is the update for uac 101 issue.
when 101 call to voicemail or 102/103
Hi,
i registered 3 uac behind same nat successfully but when i try to call each
other i didnt get any audio.
but if i use uac 102 and 103 to call into the voicemail i heard the audio
but not for 101.
kamailio is listening 60.48.218.61 and 192.168.2.3
rtpproxy is running.
asterisk is at 192.168.2.2
Hi,
which firewall do you mean? the uac firewall or the kamailio firewall?
kamailio doesnt have firewall. kamailio and rtpproxy in centos 5.4 firewall
and selinux disabled.
eth0 is configured for ppp0e and eth1 is the private ip and
echo "1" > /proc/sys/net/ipv4/ip_forward
iptables -t nat -A P
Hi,
Yup i did define #!define WITH_NAT at the beginning of config file. it
doesnt work.
been struggle for a month hope you can help.
and also in the cfg im listening to both iface.
listen=public ip
listen=192.168.2.3 [kamailio ip]
Regards,
MingHon
__
Hi,
did you also set
#!define WITH_NAT
at the beginning of your config?
That should do the job.
Carsten
2011/6/24 MingHon :
> hey list,
>
> i currently test on my asterisk 1.6 box on on pc with private ip
> and tested kamailio 3.1.3/3.1.4 + rtpproxy 1.2.1/1.2.0/1.1 on another pc
> with public
hey list,
i currently test on my asterisk 1.6 box on on pc with private ip
and tested kamailio 3.1.3/3.1.4 + rtpproxy 1.2.1/1.2.0/1.1 on another pc
with public ip and private ip.
everything installed successfully. i follow this tutorial realtime
intergration with asterisk
http://kb.asipto.com/
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