<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">Dear Alex Balashov,<br>thanks for helping<br>i try to disable force_rtp_proxy() in kamailio.cfg.but when i make call, no stream transfer. the call can make but can not hear anything .<br><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> sr-users@lists.sip-router.org<br><b><span style="font-weight: bold;">Sent:</span></b> Tue, August 24, 2010 4:58:27 PM<br><b><span
style="font-weight: bold;">Subject:</span></b> Re: [SR-Users] help to configure RTP stream with NAT.<br></font><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 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"
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