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Hello,<br>
<br>
you have to use rtpproxy in bridge mode, to route packets between
the two local network interfaces. There are many examples out there,
one shows even how to bridge between ipv4 and ipv4 networks -- you
can use it as reference:<br>
<br>
- <a class="moz-txt-link-freetext" href="http://kb.asipto.com/kamailio:kamailio-mixed-ipv4-ipv6">http://kb.asipto.com/kamailio:kamailio-mixed-ipv4-ipv6</a><br>
<br>
Cheers,<br>
Daniel<br>
<br>
<div class="moz-cite-prefix">On 8/5/13 7:12 PM, Alexandr Usov wrote:<br>
</div>
<blockquote
cite="mid:CAEaKXwBS0FUgzd+vjLzdjcNkgLvJr64oSwxAX6f_A4-juZKGxw@mail.gmail.com"
type="cite">
<div dir="ltr">
<div><br>
<br>
I have Kamailio on OpenSUSE with static real Public IP (WAN),
for ex. 1.1.1.1. I have LAN IP 2.2.2.2. Asterisk as KVM
virtual machine with LAN IP 2.2.2.101 and default GW not the
SuSe (2.2.2.2), but 2.2.2.1 pfsense LAN with PUB IP 1.1.1.2) I
am configured Registration of UA on Kamailio DB, and on
Asterisk side create a static peers with Kamailio LAN ip
(host=2.2.2.2).<br>
<br>
RTP Proxy question.<br>
<br>
/usr/sbin/rtpproxy -u daemon -l 1.1.1.1 -s <a class="moz-txt-link-freetext" href="udp:127.0.0.1">udp:127.0.0.1</a> 12221<br>
<br>
</div>
.... kamailio.cfg ...<br>
<div><br>
# RTPProxy control<br>
route[NATMANAGE] {<br>
#!ifdef WITH_NAT<br>
if (is_request()) {<br>
if(has_totag()) {<br>
if(check_route_param("nat=yes")) {<br>
setbflag(FLB_NATB);<br>
}<br>
}<br>
}<br>
if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))<br>
return;<br>
<br>
rtpproxy_manage();<br>
<br>
if (is_request()) {<br>
if (!has_totag()) {<br>
add_rr_param(";nat=yes");<br>
}<br>
}<br>
if (is_reply()) {<br>
if(isbflagset(FLB_NATB)) {<br>
fix_nated_contact();<br>
}<br>
}<br>
#!endif<br>
return;<br>
}<br>
<br>
<br>
Testing call:<br>
<br>
Whe User 1-100 calling User 1-101, on Asterisk side I see:<br>
<br>
-- Called SIP/<a moz-do-not-send="true"
href="mailto:1-100@sip1.somedomain.com.ua">1-100@sip1.somedomain.com.ua</a><br>
-- SIP/sip1.somedomain.com.ua-000004cf is ringing<br>
-- SIP/sip1.somedomain.com.ua-000004cf answered
SIP/1-101-000004ce<br>
> 0x15bc370 -- Probation passed - setting RTP source
address to <a moz-do-not-send="true"
href="http://1.1.1.1:50868">1.1.1.1:50868</a><br>
> 0x7f2b6044bd10 -- Probation passed - setting RTP
source address to <a moz-do-not-send="true"
href="http://1.1.1.1:35082">1.1.1.1:35082</a><br>
<br>
Got RTP packet from <a moz-do-not-send="true"
href="http://1.1.1.1:50868">1.1.1.1:50868</a> (type 00, seq
027109, ts 000160, len 000160)<br>
Sent RTP packet to <a moz-do-not-send="true"
href="http://1.1.1.1:35082">1.1.1.1:35082</a> (type 00, seq
037469, ts 000160, len 000160)<br>
Got RTP packet from <a moz-do-not-send="true"
href="http://1.1.1.1:50868">1.1.1.1:50868</a> (type 00, seq
027110, ts 000320, len 000160)<br>
Sent RTP packet to <a moz-do-not-send="true"
href="http://1.1.1.1:35082">1.1.1.1:35082</a> (type 00, seq
037470, ts 000320, len 000160)<br>
Got RTP packet from <a moz-do-not-send="true"
href="http://1.1.1.1:50868">1.1.1.1:50868</a> (type 00, seq
027111, ts 000480, len 000160)<br>
Sent RTP packet to <a moz-do-not-send="true"
href="http://1.1.1.1:35082">1.1.1.1:35082</a> (type 00, seq
037471, ts 000480, len 000160)<br>
Got RTP packet from <a moz-do-not-send="true"
href="http://1.1.1.1:50868">1.1.1.1:50868</a> (type 00, seq
027112, ts 000640, len 000160)<br>
Sent RTP packet to <a moz-do-not-send="true"
href="http://1.1.1.1:35082">1.1.1.1:35082</a> (type 00, seq
037472, ts 000640, len 000160)<br>
<br>
<br>
Voice transfers OK.<br>
<br>
But why not Kamailio LAN ip I receiving on the Asterisk side
with the same LAN?<br>
<br>
And Kamailio log grep:<br>
<br>
skynet:~ # tail -f /var/log/messages | grep rtpproxy<br>
2013-08-05T19:18:17.508760+03:00 skynet kamailio[25462]:
3(25481) DEBUG: rtpproxy [rtpproxy_funcs.c:148]:
check_content_type(): type <application/sdp> found valid<br>
2013-08-05T19:18:17.508875+03:00 skynet kamailio[25462]:
3(25481) DEBUG: rtpproxy [rtpproxy.c:2624]: force_rtp_proxy():
proxy reply: 63566 1.1.1.1<br>
2013-08-05T19:18:17.530765+03:00 skynet kamailio[25462]:
6(25484) DEBUG: rtpproxy [rtpproxy_funcs.c:148]:
check_content_type(): type <application/sdp> found valid<br>
2013-08-05T19:18:17.530876+03:00 skynet kamailio[25462]:
6(25484) DEBUG: rtpproxy [rtpproxy.c:2624]: force_rtp_proxy():
proxy reply: 41958 1.1.1.1<br>
2013-08-05T19:18:18.625815+03:00 skynet kamailio[25462]:
4(25482) DEBUG: rtpproxy [rtpproxy_funcs.c:148]:
check_content_type(): type <application/sdp> found valid<br>
2013-08-05T19:18:18.627131+03:00 skynet kamailio[25462]:
4(25482) DEBUG: rtpproxy [rtpproxy.c:2624]: force_rtp_proxy():
proxy reply: 39876 1.1.1.1<br>
2013-08-05T19:18:18.632649+03:00 skynet kamailio[25462]:
a=nortpproxy:yes<br>
2013-08-05T19:18:18.648075+03:00 skynet kamailio[25462]:
6(25484) DEBUG: rtpproxy [rtpproxy_funcs.c:148]:
check_content_type(): type <application/sdp> found valid<br>
2013-08-05T19:18:18.649615+03:00 skynet kamailio[25462]:
6(25484) DEBUG: rtpproxy [rtpproxy.c:2624]: force_rtp_proxy():
proxy reply: 43500 1.1.1.1<br>
2013-08-05T19:18:18.653948+03:00 skynet kamailio[25462]:
a=nortpproxy:yes<br>
2013-08-05T19:18:18.688603+03:00 skynet kamailio[25462]:
4(25482) DEBUG: rtpproxy [rtpproxy_funcs.c:148]:
check_content_type(): type <application/sdp> found valid<br>
2013-08-05T19:18:18.689762+03:00 skynet kamailio[25462]:
4(25482) DEBUG: rtpproxy [rtpproxy.c:2624]: force_rtp_proxy():
proxy reply: 63566 1.1.1.1<br>
2013-08-05T19:18:18.701062+03:00 skynet kamailio[25462]:
6(25484) DEBUG: rtpproxy [rtpproxy_funcs.c:148]:
check_content_type(): type <application/sdp> found valid<br>
2013-08-05T19:18:18.701405+03:00 skynet kamailio[25462]:
6(25484) DEBUG: rtpproxy [rtpproxy.c:2624]: force_rtp_proxy():
proxy reply: 43500 1.1.1.1<br>
2013-08-05T19:18:18.705506+03:00 skynet kamailio[25462]:
a=nortpproxy:yes<br>
<br>
</div>
<div>My goal is using Asterisk boxes behind Kamailio with the
same LAN or even OpeVPN Lan2Lan, with Public IP on Kamailio
WAN for users registration and RTP routing. So is strange to
my, why RTPproxy not rewrite source of RTP traffic from PUBLIC
Kamailio IP to LAN Kamailio IP when user A calls B via
Asterisk?<br>
<br>
</div>
<div><br>
</div>
</div>
<br>
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<pre wrap="">_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
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</pre>
</blockquote>
<br>
<pre class="moz-signature" cols="72">--
Daniel-Constantin Mierla - <a class="moz-txt-link-freetext" href="http://www.asipto.com">http://www.asipto.com</a>
<a class="moz-txt-link-freetext" href="http://twitter.com/#!/miconda">http://twitter.com/#!/miconda</a> - <a class="moz-txt-link-freetext" href="http://www.linkedin.com/in/miconda">http://www.linkedin.com/in/miconda</a>
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