<div>Carsten</div>
<div>Really appreciate your input.</div>
<div> </div>
<div>I am completely new to this. Please bear with me with easy questions.</div>
<div> </div>
<div>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...</div>
<div> </div>
<div><span lang="EN">
<p># Main SIP request routing logic</p>
<p># - processing of any incoming SIP request starts with this route</p>
<p>route {</p>
<p># per request initial checks</p>
<p>route(REQINIT);</p>
<p># NAT detection</p>
<p>route(NAT); .......................</p>
<p>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.</p>
<p> </p>
<p>Thanks</p>
<p>Austin.</p>
<p> </p></span><br><br></div>
<div class="gmail_quote">On Fri, Oct 7, 2011 at 5:39 PM, Carsten Bock <span dir="ltr"><<a href="mailto:carsten@ng-voice.com">carsten@ng-voice.com</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">Hi,<br>you can simply use Kamailio and RTPProxy here.<br>You simply can use rtpproxy_manage() in the request and in the reply;<br>
rtpproxy_destroy() in the Bye-Message. You must also use<br>record_route() in order to have all subsequent requests going through<br>your proxy. That should do your job.<br><br>May you want also to look at SEMS (<a href="http://iptel.org/sems" target="_blank">iptel.org/sems</a>) as an SBC. SEMS can<br>
do that as well, but additionally also teardown in case of<br>RTP-Timeout, Codec Enforcement (e.g. only allow G711) and of course<br>topology hiding.<br><br>Kind regards,<br>
<div>
<div></div>
<div class="h5">Carsten<br><br>2011/10/7 Austin Einter <<a href="mailto:austin.einter@gmail.com">austin.einter@gmail.com</a>><br>><br>> Dear Carsten<br>> Thanks for reply.<br>><br>> I did not install rtp proxy . I just installed kamailio after that I was able to run both kamailio and rtp proxy.<br>
> I am surprised, I thought after installing kamailio 3.1.5, I will be able to execute both kamailio and rtpproxy.<br>> Not sure it it was installed in my lab pc by somebody else previously.<br>><br>> First of all I would like to check if the scenario I am talking of is a right case to use Kamalio or not.<br>
><br>> 1. I will be running the sip softphones in private/public network.<br>><br>> 2. I want to run kamailio proxy in public network (as an intermediate proxy).<br>><br>> 3. I will hav a main proxy in public detwork.<br>
><br>> What I want is all my signalling and RTP packets should pass through intermediate proxy (that is Kamailio only).<br>><br>> For this purpose do you suggest Kamailio or someother proxy/b2bua.<br>> I am able to force SIP packets through kamailio proxy (by use of route stuff in softphone).<br>
><br>> 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.<br>
><br>> 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.<br>><br>> Kindly help me to choose right proxy or should I look for b2bua.<br>
><br>> Thanks<br>> Austin<br>><br>><br>><br>> On Fri, Oct 7, 2011 at 3:05 PM, Carsten Bock <<a href="mailto:carsten@ng-voice.com">carsten@ng-voice.com</a>> wrote:<br>>><br>>> Hi,<br>
>> did you also install the RTPProxy itself? It is an addon to Kamailio and not a part of Kamailio.<br>>> You find the RTPProxy here: <a href="http://www.rtpproxy.org/" target="_blank">www.rtpproxy.org</a><br>
>> Carsten<br>>><br>>> 2011/10/7 Austin Einter <<a href="mailto:austin.einter@gmail.com">austin.einter@gmail.com</a>><br>>>><br>>>> Hi<br>>>> I downloaded 3.1.5 Kamalio source, did make and install. Tried to run, and tested few basic scenarios, everything working fine.<br>
>>><br>>>> 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 .<br>
>>><br>>>> For this I am referring, <a href="http://nil.uniza.sk/sip/nat-fw/configuring-nat-traversal-using-kamailio-31-and-rtpproxy-server" target="_blank">http://nil.uniza.sk/sip/nat-fw/configuring-nat-traversal-using-kamailio-31-and-rtpproxy-server</a> page.<br>
>>><br>>>> 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:".<br>
>>><br>>>> I tried to search rtpproxy configuration file, could not locate.<br>>>> While making and installing Kamalio proxy, the commands I used "make prefix=/usr/local/ all and make prefix=/usr/local/ install".<br>
>>> Can somebody point me why rtpproxy config file is missing.<br>>>><br>>>> Also the link I am refering to is correct or wrong one.<br>>>><br>>>> Best Regards,<br>>>> Austin<br>
>>> _______________________________________________<br>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>>>> <a href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a><br>
>>> <a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" target="_blank">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><br>>>><br>>><br>>><br>>><br>
>> --<br>>> Carsten Bock<br>>> CEO (Geschäftsführer)<br>>> ng-voice GmbH i. Gr.<br>>> Schomburgstr. 80<br>>> D-22767 Hamburg / Germany<br>>> <a href="http://www.ng-voice.com/" target="_blank">http://www.ng-voice.com</a><br>
>> mailto:<a href="mailto:carsten@ng-voice.com">carsten@ng-voice.com</a><br>>> Mobile +49 179 2021244<br>>> Office +49 40 34927219<br>>> Fax +49 40 34927220<br>>> Hier finden Sie unsere handelsrechtlichen Pflichtangaben:<br>
>> <a href="http://www.ng-voice.com/imprint/" target="_blank">http://www.ng-voice.com/imprint/</a><br>>><br>>> _______________________________________________<br>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>
>> <a href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a><br>>> <a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" target="_blank">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><br>
>><br>><br>><br>> _______________________________________________<br>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>> <a href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a><br>
> <a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" target="_blank">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><br>><br><br><br><br>--<br>Carsten Bock<br>CEO (Geschäftsführer)<br>
ng-voice GmbH i. Gr.<br>Schomburgstr. 80<br>D-22767 Hamburg / Germany<br><a href="http://www.ng-voice.com/" target="_blank">http://www.ng-voice.com</a><br>mailto:<a href="mailto:carsten@ng-voice.com">carsten@ng-voice.com</a><br>
Mobile +49 179 2021244<br>Office +49 40 34927219<br>Fax +49 40 34927220<br>Hier finden Sie unsere handelsrechtlichen Pflichtangaben:<br><a href="http://www.ng-voice.com/imprint/" target="_blank">http://www.ng-voice.com/imprint/</a><br>
<br>_______________________________________________<br>SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br><a href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a><br><a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" target="_blank">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><br>
</div></div></blockquote></div><br>