<div dir="ltr">Thanks Vasily<div><br></div><div>direction parameter i understood it was as ie options on rtpproxy, maybe im wrong.</div><div><br>About them , i was ngreping all trace and that was the option to get SDP correct mapped using RTPengine..</div><div><br></div><div>WIll check with your comments and let you know back. </div><div><br></div><div>BR</div></div><div class="gmail_extra"><br><div class="gmail_quote">2015-07-14 17:16 GMT+02:00 Vasiliy Ganchev <span dir="ltr"><<a href="mailto:vasiliy.ganchev@wildix.com" target="_blank">vasiliy.ganchev@wildix.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Alberto Sagredo-2 wrote<br>
<div><div class="h5">> Thanks Vasily i have changed a little today using a RTPPROXY route.<br>
><br>
> Thats what i have right now<br>
><br>
> But its not working as expected<br>
><br>
> What i try is to detect if i have SAVP from endpoint and translate to RTP<br>
> to ASterisk an later RTP from ASterisk translate to SRTP using rtpengine<br>
><br>
> I had extrange behaviour using rtpproxy that send SRTP to Asterisk and i<br>
> have SRTP calls, i though rtpproxy 2.0 could not manage SRTP calls. but it<br>
> pass it to Asterisk<br>
><br>
> Using RTPengine i have tested with rtpproxy_manage as you see and also<br>
> with<br>
> rtpengine.<br>
><br>
> If i load both start_recording() feature is lost.<br>
><br>
> On rtpengine (behind NAT) im using it as:<br>
><br>
> INTERFACES="192.168.0.178 internal/<a href="http://192.168.0.178" rel="noreferrer" target="_blank">192.168.0.178</a> external/<a href="http://192.168.0.179" rel="noreferrer" target="_blank">192.168.0.179</a><br>
> !EXTERN_IP<br>
><br>
><br>
> On NATMANAGE route i call directly<br>
><br>
> route(RTPPROXY);<br>
><br>
><br>
> Hope this helps<br>
><br>
><br>
> -----<br>
><br>
> route[RTPPROXY] {<br>
><br>
> if (is_method("INVITE")){<br>
><br>
> if(ds_is_from_list(1)){<br>
><br>
> if (is_ip_rfc1918("$si")) {<br>
><br>
> xlog("L_INFO", "LLamada desde los<br>
> Asterisk_$si -> RTPPROXY\n");<br>
><br>
> if (sdp_get_line_startswith("$avp(mline)", "m="))<br>
><br>
> {<br>
><br>
> #!ifdef WITH_RTPENGINE<br>
><br>
> if ($avp(mline) =~ "SAVP")<br>
><br>
> {<br>
><br>
> xlog("L_INFO", "Tenemos SRTP ");<br>
><br>
> xlog("L_INFO", "Llamada entre Extensiones<br>
> -> RTPENGINE INTERNAL");<br>
><br>
> rtpengine_manage("direction=internal<br>
> replace-origin replace-session-connection ICE=remove");<br>
><br>
> return;<br>
><br>
> }<br>
><br>
> #!endif<br>
><br>
><br>
> if ($avp(mline) =~ "AVP")<br>
><br>
> {<br>
><br>
> xlog("L_INFO", "Tenemos RTP ");<br>
><br>
> xlog("L_INFO", "Llamada entre Extensiones<br>
> -> RTPROXY ");<br>
><br>
><br>
> #!ifdef WITH_RTPPROXY<br>
><br>
> set_rtp_proxy_set("1");<br>
><br>
> rtpproxy_manage("fwei");<br>
><br>
> start_recording();<br>
><br>
> #!endif<br>
><br>
><br>
> #!ifdef WITH_RTPENGINE<br>
><br>
> set_rtp_proxy_set("2");<br>
><br>
> rtpproxy_manage("ie");<br>
><br>
> #!endif<br>
><br>
> }<br>
><br>
> }<br>
><br>
> }<br>
><br>
> }else if(!ds_is_from_list()){<br>
><br>
><br>
> if (sdp_get_line_startswith("$avp(mline)", "m="))<br>
><br>
> {<br>
><br>
> #!ifdef WITH_RTPENGINE<br>
><br>
> if ($avp(mline) =~ "SAVP")<br>
><br>
> {<br>
><br>
> xlog("L_INFO", "Tenemos SRTP ");<br>
><br>
> xlog("L_INFO", "Llamada entre Extensiones<br>
> -> RTPENGINE EXTERNAL ");<br>
><br>
> rtpengine_manage("direction=external<br>
> replace-origin replace-session-connection ICE=remove");<br>
><br>
> return;<br>
><br>
> }<br>
><br>
><br>
> #!endif<br>
><br>
> if ($avp(mline) =~ "AVP")<br>
><br>
> {<br>
><br>
> xlog("L_INFO", "Tenemos RTP ");<br>
><br>
> xlog("L_INFO", "Llamada entre Extensiones<br>
> -> RTPROXY ");<br>
><br>
><br>
> #!ifdef WITH_RTPPROXY<br>
><br>
> set_rtp_proxy_set("1");<br>
><br>
> rtpproxy_manage("fwie");<br>
><br>
> start_recording();<br>
><br>
> #!endif<br>
><br>
><br>
> #!ifdef WITH_RTPENGINE<br>
><br>
> set_rtp_proxy_set("2");<br>
><br>
> rtpproxy_manage("ei");<br>
><br>
> #!endif<br>
><br>
><br>
> }<br>
><br>
> }<br>
><br>
><br>
><br>
> }<br>
><br>
> }<br>
><br>
><br>
> }<br>
><br>
><br>
><br>
</div></div>> 2015-07-14 14:24 GMT+02:00 Vasiliy Ganchev <<br>
<br>
> vasiliy.ganchev@<br>
<br>
> >:<br>
<div><div class="h5">><br>
>> Alberto Sagredo-2 wrote<br>
>> > ...<br>
>> > I have been able to make SRTP To RTP to Asterisk<br>
>> ><br>
>> > But im not able to call between SRTP extensions, i understand also SRTP<br>
>> to<br>
>> > RTP would work as im doing with Asterisk (Only the speak SRTP as<br>
>> rtpengine<br>
>> > trasncode)<br>
>> ><br>
>> ><br>
>> > If you need any more info let me know.<br>
>> ><br>
>> > _______________________________________________<br>
>> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>
>><br>
>> > sr-users@.sip-router<br>
>><br>
>> > <a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" rel="noreferrer" target="_blank">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><br>
>><br>
>> Hi!<br>
>> If you make SRTP to RTP to Asterisk, you possibly will need vice versa<br>
>> conversion (when request coming from Asterisk to client with SRTP).<br>
>><br>
>> Can you describe the logic of test case: (UA-A (SRTP) --> Kamailio (make<br>
>> SRTP->RTP) .... etc.<br>
>><br>
>> Because your explanation is difficult to understand.<br>
>><br>
>> Cheers!<br>
>><br>
>><br>
>><br>
>> --<br>
>> View this message in context:<br>
>> <a href="http://sip-router.1086192.n5.nabble.com/Issue-handling-SRTP-and-RTP-with-rtpproxy-and-rtpengine-tp139488p139521.html" rel="noreferrer" target="_blank">http://sip-router.1086192.n5.nabble.com/Issue-handling-SRTP-and-RTP-with-rtpproxy-and-rtpengine-tp139488p139521.html</a><br>
>> Sent from the Users mailing list archive at Nabble.com.<br>
>><br>
>> _______________________________________________<br>
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>
>><br>
<br>
</div></div>> sr-users@.sip-router<br>
<br>
>> <a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" rel="noreferrer" target="_blank">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><br>
<span class="">>><br>
><br>
> _______________________________________________<br>
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>
<br>
> sr-users@.sip-router<br>
<br>
> <a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" rel="noreferrer" target="_blank">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><br>
<br>
</span>What about ICE, where it has to work? (client->Kamailio - yes,<br>
Kamailio->Asterisk - no) or somehow else.<br>
<br>
For your description, I think you need to add something like this:<br>
- Kamailio -> Asterisk<br>
rtpengine_manage("...............RTP/AVP"); ///// this will change<br>
profile to RTP/AVP<br>
<br>
- Asterisk -> Kamailio<br>
rtpengine_manage("...............RTP/SAVPF"); ///// this will make<br>
backward changes<br>
<br>
Also read thoroughly the meaning and usage of "direction" parameter, I think<br>
you have little misunderstanding of how it works (maybe I'm wrong and you<br>
use it as it has to be, but re-read it anyway)<br>
<br>
<br>
<br>
--<br>
View this message in context: <a href="http://sip-router.1086192.n5.nabble.com/Issue-handling-SRTP-and-RTP-with-rtpproxy-and-rtpengine-tp139488p139556.html" rel="noreferrer" target="_blank">http://sip-router.1086192.n5.nabble.com/Issue-handling-SRTP-and-RTP-with-rtpproxy-and-rtpengine-tp139488p139556.html</a><br>
<div class="HOEnZb"><div class="h5">Sent from the Users mailing list archive at Nabble.com.<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>
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</div></div></blockquote></div><br></div>