7 Oct
2011
7 Oct
'11
8:38 a.m.
Carsten
Really appreciate your input.
I am completely new to this. Please bear with me with easy questions.
When you say I can use rtpproxy_manage() in the request and response and use
rtpproxy_destroy() in BYE, I want to know where I have to modify. Is it
kamailio.cfg file where I need to modify. If it is the case, do I need to
modify the route block as below..., kindly guide me here with steps...
# Main SIP request routing logic
# - processing of any incoming SIP request starts with this route
route {
# per request initial checks
route(REQINIT);
# NAT detection
route(NAT); .......................
Also, we are just telling rtpproxy_destroy in BYE case, will it not affect
other calls. At any given point of time, there might be more than 1 calls,
doing rtpproxy_destroy will stop rtp session for disconnecting call or can
it affect the existing active calls.
Thanks
Austin.
On Fri, Oct 7, 2011 at 5:39 PM, Carsten Bock <carsten(a)ng-voice.com> wrote:
Hi,
you can simply use Kamailio and RTPProxy here.
You simply can use rtpproxy_manage() in the request and in the reply;
rtpproxy_destroy() in the Bye-Message. You must also use
record_route() in order to have all subsequent requests going through
your proxy. That should do your job.
May you want also to look at SEMS (iptel.org/sems) as an SBC. SEMS can
do that as well, but additionally also teardown in case of
RTP-Timeout, Codec Enforcement (e.g. only allow G711) and of course
topology hiding.
Kind regards,
Carsten
2011/10/7 Austin Einter <austin.einter(a)gmail.com>
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--
Carsten Bock
CEO (Geschäftsführer)
ng-voice GmbH i. Gr.
Schomburgstr. 80
D-22767 Hamburg / Germany
http://www.ng-voice.com
mailto:carsten@ng-voice.com
Mobile +49 179 2021244
Office +49 40 34927219
Fax +49 40 34927220
Hier finden Sie unsere handelsrechtlichen Pflichtangaben:
http://www.ng-voice.com/imprint/
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Dear Carsten
Thanks for reply.
I did not install rtp proxy . I just installed kamailio after that I was
able to
run both kamailio and rtp proxy.
I am surprised, I thought after installing
kamailio 3.1.5, I will be able
to execute both kamailio and rtpproxy.
Not sure it it was installed in my lab pc by
somebody else previously.
First of all I would like to check if the scenario I am talking of is a
right case
to use Kamalio or not.
1. I will be running the sip softphones in private/public network.
2. I want to run kamailio proxy in public network (as an intermediate
proxy).
3. I will hav a main proxy in public detwork.
What I want is all my signalling and RTP packets should pass through
intermediate
proxy (that is Kamailio only).
For this purpose do you suggest Kamailio or someother proxy/b2bua.
I am able to force SIP packets through kamailio proxy (by use of route
stuff in
softphone).
My main concern is when INVITE message pass through Kamailio,
will Kamailio proxy
change the IP address of phone present in SDP to self
address so that media packets from called party side will come to kamailio
proxy.
Same concern otherway. Like when 200 OK to INVITE passes through
kamailio, will
kamailio change IPaddress in SDP to self IP, so that media
packets from calling side will reach kamailio.
Kindly help me to choose right proxy or should I look for b2bua.
Thanks
Austin
On Fri, Oct 7, 2011 at 3:05 PM, Carsten Bock <carsten(a)ng-voice.com>
wrote:
>
> Hi,
> did you also install the RTPProxy itself? It is an addon to Kamailio and
not a
part of Kamailio.
> You find the RTPProxy here: www.rtpproxy.org
> Carsten
>
> 2011/10/7 Austin Einter <austin.einter(a)gmail.com>
>>
>> Hi
>> I downloaded 3.1.5 Kamalio source, did make and install. Tried to run,
and
tested few basic scenarios, everything working fine.
>>
>> I have a main proxy, want to use kamailio as intermediate proxy and my
requirement is all RTP packets should pass through machine in which Kamalio
proxy is running. I beleive RTP Proxy I can run in same machine where
Kamailio proxy is running with proper configuration .
>>
>> For this I am referring,
http://nil.uniza.sk/sip/nat-fw/configuring-nat-traversal-using-kamailio-31-…
page.
>>
>> In one of the steps it says "So, if we are using the Rtpproxy server
with default configuration, we have to open /etc/default/rtpproxy file and
uncomment following line regarding of udp socket, that will be sued for
interconnection:".
>>
>> I tried to search rtpproxy configuration file, could not locate.
>> While making and installing Kamalio proxy, the commands I used "make
prefix=/usr/local/ all and make prefix=/usr/local/ install".
Can somebody point me why rtpproxy config file is
missing.
Also the link I am refering to is correct or wrong one.
Best Regards,
Austin
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SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
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--
Carsten Bock
CEO (Geschäftsführer)
ng-voice GmbH i. Gr.
Schomburgstr. 80
D-22767 Hamburg / Germany
http://www.ng-voice.com
mailto:carsten@ng-voice.com
Mobile +49 179 2021244
Office +49 40 34927219
Fax +49 40 34927220
Hier finden Sie unsere handelsrechtlichen Pflichtangaben:
http://www.ng-voice.com/imprint/
_______________________________________________
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