<html><head><style type="text/css"><!-- DIV {margin:0px;} --></style></head><body><div style="font-family:times new roman,new york,times,serif;font-size:12pt">Hi Alex Balashov,<br>two clients is behind NAT, when i configure nathelper, the call make ok, but RTP proxy handle media stream, I want to make media stream go direct from sip client to another.<br>so is there any solve ?<br><div> </div>TRUONG NGOC THANH<br>Telecommunications Engineer<br>Tel: 0984 480 646<br>Y!M: ngoc217thanh<div><br></div><div style="font-family: times new roman,new york,times,serif; font-size: 12pt;"><br><div style="font-family: arial,helvetica,sans-serif; font-size: 13px;"><font face="Tahoma" size="2"><hr size="1"><b><span style="font-weight: bold;">From:</span></b> Alex Balashov <abalashov@evaristesys.com><br><b><span style="font-weight: bold;">To:</span></b> truong ngoc THANH <ngoc217thanh@yahoo.com><br><b><span style="font-weight: bold;">Cc:</span></b>
kamailio <sr-users@lists.sip-router.org><br><b><span style="font-weight: bold;">Sent:</span></b> Tue, August 24, 2010 5:41:34 PM<br><b><span style="font-weight: bold;">Subject:</span></b> Re: [SR-Users] help to configure RTP stream with NAT.<br></font><br>
In that case, there is a network or transport-layer reachability issue <br>between the two clients.<br><br>On 08/24/2010 06:24 AM, truong ngoc THANH wrote:<br><br>> Dear Alex Balashov,<br>> thanks for helping<br>> i try to disable force_rtp_proxy() in kamailio.cfg.but when i make<br>> call, no stream transfer. the call can make but can not hear anything .<br>><br>> TRUONG NGOC THANH<br>> Telecommunications Engineer<br>> Tel: 0984 480 646<br>> Y!M: ngoc217thanh<br>><br>><br>> ----------------------------------------------------------------------<br>> *From:* Alex Balashov <<a ymailto="mailto:abalashov@evaristesys.com" href="mailto:abalashov@evaristesys.com">abalashov@evaristesys.com</a>><br>> *To:* <a ymailto="mailto:sr-users@lists.sip-router.org" href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a><br>> *Sent:* Tue, August 24, 2010 4:58:27 PM<br>> *Subject:* Re: [SR-Users]
help to configure RTP stream with NAT.<br>><br>> On 08/24/2010 05:41 AM, truong ngoc THANH wrote:<br>><br>> > hi all,<br>> > i have using RTP proxy, and i see that RTP stream is handled by RTP<br>> > proxy. so how to configure in kamailio or which module make RTP stream<br>> > direct from sip client to another one ?<br>> > please suggest if anyone know.<br>><br>> On calls where you do not want rtpproxy to relay media, just don't use<br>> it (don't call force_rtp_proxy())?<br>><br>> -- Alex Balashov - Principal<br>> Evariste Systems LLC<br>> 1170 Peachtree Street<br>> 12th Floor, Suite 1200<br>> Atlanta, GA 30309<br>> Tel: +1-678-954-0670<br>> Fax: +1-404-961-1892<br><span>> Web: <a target="_blank" href="http://www.evaristesys.com/">http://www.evaristesys.com/</a></span><br>><br>> _______________________________________________<br>> SIP
Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>> <a ymailto="mailto:sr-users@lists.sip-router.org" href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a> <mailto:<a ymailto="mailto:sr-users@lists.sip-router.org" href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a>><br><span>> <a target="_blank" href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a></span><br>><br><br><br>-- <br>Alex Balashov - Principal<br>Evariste Systems LLC<br>1170 Peachtree Street<br>12th Floor, Suite 1200<br>Atlanta, GA 30309<br>Tel: +1-678-954-0670<br>Fax: +1-404-961-1892<br>Web: <a href="http://www.evaristesys.com/" target="_blank">http://www.evaristesys.com/</a><br></div></div>
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