<meta content="text/html; charset=utf-8" http-equiv="Content-Type">
<font face="Courier New, Courier, monospace">Hello;<br>
you can use rtpproxy or rtpengine. you need to search how to use
on google. rtpproxy/rtpengine fixes sdp packets that has tons of
flags and there is a lot of option on sdpops module to change it.
for example; you can check with nat_uac_client(8) and fix sdp with
rptproxy_offer/rtpproxy_manage("cor") function.<br>
</font><br>
there is 2-3 sample about it , here is kamailio doc how to use
functions.<br>
<br>
<a class="moz-txt-link-freetext"
href="http://www.kamailio.org/docs/modules/4.3.x/modules/rtpproxy.html#idp2564344" target="_top" rel="nofollow" link="external">http://www.kamailio.org/docs/modules/4.3.x/modules/rtpproxy.html#idp2564344</a><br>
<a class="moz-txt-link-freetext"
href="http://www.kamailio.org/docs/modules/4.3.x/modules/nathelper.html#nathelper.f.nat_uac_test" target="_top" rel="nofollow" link="external">http://www.kamailio.org/docs/modules/4.3.x/modules/nathelper.html#nathelper.f.nat_uac_test</a><br>
<br>
-- samples ---<br>
<a class="moz-txt-link-freetext"
href="http://saevolgo.blogspot.com.tr/2013/08/rtpproxy-revisited-kamailio-40.html" target="_top" rel="nofollow" link="external">http://saevolgo.blogspot.com.tr/2013/08/rtpproxy-revisited-kamailio-40.html</a><br>
<a class="moz-txt-link-freetext"
href="http://www.fredposner.com/1457/kamailio-behind-nat/" target="_top" rel="nofollow" link="external">http://www.fredposner.com/1457/kamailio-behind-nat/</a><br>
<a class="moz-txt-link-freetext"
href="http://nil.uniza.sk/sip/nat-fw/configuring-nat-traversal-using-kamailio-31-and-rtpproxy-server" target="_top" rel="nofollow" link="external">http://nil.uniza.sk/sip/nat-fw/configuring-nat-traversal-using-kamailio-31-and-rtpproxy-server</a><br>
<br>
Cheers.<br>
<br>
<div class="moz-cite-prefix">2016-05-06 09:59 tarihinde 王欣鑫 [via SIP
Router] yazdı:<br>
</div>
<blockquote style='border-left:2px solid #CCCCCC;padding:0 1em' cite="mid:1462517980745-148183.post@n5.nabble.com"
type="cite"> Hello,ycaner! <br>
<br>
Now I use a real IMS user data to register this user(for example
18616278715) to China Unicom's IMS through kamailio. <br>
<br>
I can register successsfully, and can make call using this number,
the callee can pick up phone. However <br>
neither party can hear counter part. <br>
<br>
So, this should be rtp problem. And I captured a .pcap file on
port 5060 like below. In this, I see in SDP information <br>
it says 192.168.199.124, it's a private address. So maybe the
callee can't setup a circuit to this ip, and thus no media was
transfered? <br>
<br>
Should I uncomment RTPPROXY_ADDRESS, as in my pcscf.cfg there are
two lines. If I should do this, what ip and port should I use to
replace "udp:10.1.2.186:22222 udp:10.1.27.217:22222" ?
<br>
<br>
# IP-Adress(es) of the RTP-Proxy <br>
##!define RTPPROXY_ADDRESS "udp:10.1.2.186:22222
udp:10.1.27.217:22222" <br>
##!define RTPPROXY_ADDRESS "udp:10.1.2.186:22222" <br>
<br>
This is the .pcap file captured on port 5060. <br>
<a moz-do-not-send="true"
href="http://sip-router.1086192.n5.nabble.com/file/n148183/5060.pcap"
target="_top" rel="nofollow" link="external">5060.pcap</a><br>
<br>
<img moz-do-not-send="true"
src="http://sip-router.1086192.n5.nabble.com/file/n148183/5060.jpg"
border="0"><br>
<br>
<br>
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